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<title>Connecting to Google Voice Via SIP in VOIP Tech Chat</title>
<link>http://www.dslreports.com/forum/r22105161</link>
<description></description>
<language>en</language>
<pubDate>Sun, 22 Nov 2009 23:18:51 EDT</pubDate>
<lastBuildDate>Sun, 22 Nov 2009 23:18:51 EDT</lastBuildDate>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22919224</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : At least we cleared up the mystery of whether GV has once again begun to alllow inbound SIP connections.<br><br>There is little standardization in the way call destinations are formatted. It takes a lot of trial and error, a lot of combinations and permutations, before you can conclude that GV is really not allowing these connections.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22919224</guid>
<pubDate>Mon, 24 Aug 2009 23:48:26 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22918078</link>
<description><![CDATA[<A HREF="/useremail/u/1663559"><b>dyrewolfe</b></A> : callwithus behaved exactly the same way (everything but the GV # was stripped away) even with "sip:" added. sipgate threw back an error "unknown domain" and setting up a dial string in sipsorcery resulted in a timeout.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22918078</guid>
<pubDate>Mon, 24 Aug 2009 19:53:31 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22917973</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : Using a dummy IP address still results in a connected call, so it must be discarding the @ipadress portion, and it also seems to not care that the string begins with "sip:". Very odd, especially how using "sip:" passes the call directly through the GV system (instead of terminating in the GV voicemail). ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22917973</guid>
<pubDate>Mon, 24 Aug 2009 19:33:21 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22917590</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : I suspect it is some sort of odd manipulation that both Sipgate and VoicePulse are doing with the string. You were also going to try calling your number at a dummy SIP address. That might show whether it strips out the number of an address that won't otherwise connect.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22917590</guid>
<pubDate>Mon, 24 Aug 2009 18:20:05 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22917473</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : Ok, this gets even stranger. I tried doing (gvnumber)@googleip and the call went through as before. Dialing via this method yields my GV voicemail if I dont answer the call. Upon checking the logs however, it shows that I connected to just (gvnumber) not a SIP address. I then decided to try the method by attaching the prefix "sip:" to (gvnumber)@googleIP and dialed away again. This time it also rang, however instead of being picked up by my GV voicemail the vm of my cellphone (part of the ring group) picked up the call. A check of the GV call logs shows a missed call from my Voicepulse account, but not a SIP call (I doubt if its even setup to display SIP calls). Finally, I tried it using Sipgate but also including the "sip:" prefix. My cellphone rang, but when I didnt answer it also skipped the GV voicemail and went directly to my cellphones vm box. A quick check of the Sipgate logs shows that even with the "sip:" prefix, the call is recorded simply as a regular number.  <br><small>--<br>Obama = Jimmy Carter part 2<br>"Secret operations are essential in war; upon them the army relies to make its every move"<br><b>-Sun Tzu-</b> <br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22917473</guid>
<pubDate>Mon, 24 Aug 2009 17:58:56 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22916915</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Another way to tell might be to look at your ATA logs. I'll be interested to see what you come up with.<br><br>It's purely academic at this point since, for me, the Sip Sorcery app does pretty much everything I was trying to do through a GV SIP connection.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22916915</guid>
<pubDate>Mon, 24 Aug 2009 16:11:45 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22916837</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>I will check this evening. I can also try testing this by entering in (myGVnumber)@bogusIPaddress and see if it still completes the call. If it does, then we know its just grabbing the digits and completing it via the PSTN. <br> </div>I just created some outbound forwarding rules (speed dials) on my VOIPo line, with destinations as follows:<br><br>myGVnumber@216.239.37.15:5061<br><br>and<br><br>sip:myGVnumber@216.239.37.15:5061<br><br>The first one worked, i.e., got to my GV voicemail, whereas the second one timed out. Call logs showed the outbound speed dial numbers, so there was no way to tell whether the first one was a SIP call. The only indication is that the speed dial entry itself shows only a number for the first one, but a SIP address for the second. Clearly, VOIPo is just grabbing the number on the first speed dial.<br> </div>I will test it out in about an hour and report back. <br><small>--<br>Obama = Jimmy Carter part 2<br>"Secret operations are essential in war; upon them the army relies to make its every move"<br><b>-Sun Tzu-</b> <br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22916837</guid>
<pubDate>Mon, 24 Aug 2009 15:56:04 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22916738</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I will check this evening. I can also try testing this by entering in (myGVnumber)@bogusIPaddress and see if it still completes the call. If it does, then we know its just grabbing the digits and completing it via the PSTN. <br> </div>I just created some outbound forwarding rules (speed dials) on my VOIPo line, with destinations as follows:<br><br>myGVnumber@216.239.37.15:5061<br><br>and<br><br>sip:myGVnumber@216.239.37.15:5061<br><br>The first one worked, i.e., got to my GV voicemail, whereas the second one timed out. Call logs showed the outbound speed dial numbers, so there was no way to tell whether the first one was a SIP call. The only indication is that the speed dial entry itself shows only a number for the first one, but a SIP address for the second. Clearly, VOIPo is just grabbing the number on the first speed dial.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22916738</guid>
<pubDate>Mon, 24 Aug 2009 15:38:23 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22916576</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>I do not think the provider is just using the SIP URI and passing it through, however I am not sure how to verify that. <br> </div>What do the Sipgate and Voicepulse call logs say? Does the destination number display as your GV SIP URI?<br> </div>I will check this evening. I can also try testing this by entering in (myGVnumber)@bogusIPaddress and see if it still completes the call. If it does, then we know its just grabbing the digits and completing it via the PSTN. <br><small>--<br>Obama = Jimmy Carter part 2<br>"Secret operations are essential in war; upon them the army relies to make its every move"<br><b>-Sun Tzu-</b> <br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22916576</guid>
<pubDate>Mon, 24 Aug 2009 15:07:11 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22916544</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I do not think the provider is just using the SIP URI and passing it through, however I am not sure how to verify that. <br> </div>What do the Sipgate and Voicepulse call logs say? Does the destination number display as your GV SIP URI?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22916544</guid>
<pubDate>Mon, 24 Aug 2009 14:59:14 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22915655</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>Perhaps it has something to do with the fact that both my numbers were originally GC. Are the people who are unable to do this using numbers that were assigned via GC, or are you using numbers that from day 1 were GV?<br> </div>No, mine was also originally GC. I can't say I've tried everything to make this work -- I haven't done any real troubleshooting. But when this was working before, it was pretty straightforward, and I didn't have a problem. Then it simply stopped working. I think I'm doing what I did before.<br><br>Strange. Have you tried originating from different phones and providers? Is it possible that, like dryewolfe, your provider is just taking the username of the SIP URI and sending it out as PSTN?<br> </div>I have been able to dial into my GV number(s) via SIP using a softphone with both my Sipgate and Voicepulse accounts. I do not think the provider is just using the SIP URI and passing it through, however I am not sure how to verify that. <br><small>--<br>Obama = Jimmy Carter part 2<br>"Secret operations are essential in war; upon them the army relies to make its every move"<br><b>-Sun Tzu-</b> <br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22915655</guid>
<pubDate>Mon, 24 Aug 2009 11:58:08 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22915546</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Perhaps it has something to do with the fact that both my numbers were originally GC. Are the people who are unable to do this using numbers that were assigned via GC, or are you using numbers that from day 1 were GV?<br> </div>No, mine was also originally GC. I can't say I've tried everything to make this work -- I haven't done any real troubleshooting. But when this was working before, it was pretty straightforward, and I didn't have a problem. Then it simply stopped working. I think I'm doing what I did before.<br><br>Strange. Have you tried originating from different phones and providers? Is it possible that, like dryewolfe, your provider is just taking the username of the SIP URI and sending it out as PSTN?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22915546</guid>
<pubDate>Mon, 24 Aug 2009 11:37:26 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22915030</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Hmm. So now we're back down to one person who says this works. I tried a few more times, and never could connect.<br> </div>Perhaps it has something to do with the fact that both my numbers were originally GC. Are the people who are unable to do this using numbers that were assigned via GC, or are you using numbers that from day 1 were GV?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22915030</guid>
<pubDate>Mon, 24 Aug 2009 09:55:14 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22913923</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Hmm. So now we're back down to one person who says this works. I tried a few more times, and never could connect.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22913923</guid>
<pubDate>Sun, 23 Aug 2009 23:20:34 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22913494</link>
<description><![CDATA[<A HREF="/useremail/u/1663559"><b>dyrewolfe</b></A> : nevermind. I think callwithus was dropping the '@216.239.37.15:5061' and dialing it as a regular number. It only appeared to be making a SIP call. Oh well.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22913494</guid>
<pubDate>Sun, 23 Aug 2009 21:28:08 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22912795</link>
<description><![CDATA[<A HREF="/useremail/u/1663559"><b>dyrewolfe</b></A> : I just tried with 10 digits and it wouldn't connect. 11 digits works though. Try that. Also, regarding CID, I can't connect if I set the CID to my GV # (testing all of this with a callwithus account).]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22912795</guid>
<pubDate>Sun, 23 Aug 2009 17:35:59 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22912752</link>
<description><![CDATA[<A HREF="/useremail/u/890631"><b>OmagicQ</b></A> : <div class="bquote"><small>said by  dyrewolfe <A HREF="/useremail/u/1663559"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  OmagicQ <A HREF="/useremail/u/890631"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>what's the IP address for GV for this to work?<br><br>I tried myGVnumber@216.239.37.15:5061 and got a request timeout.<br> </div>That's correct. Are you using 11 digits for your number?<br> </div>I dialed 10digits, Do I need to add the 1? Also does the outbound caller id make any difference because I'm using Voxalot and I can't set it.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22912752</guid>
<pubDate>Sun, 23 Aug 2009 17:19:26 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22907432</link>
<description><![CDATA[<A HREF="/useremail/u/1663559"><b>dyrewolfe</b></A> : <div class="bquote"><small>said by  OmagicQ <A HREF="/useremail/u/890631"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>what's the IP address for GV for this to work?<br><br>I tried myGVnumber@216.239.37.15:5061 and got a request timeout.<br> </div>That's correct. Are you using 11 digits for your number?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22907432</guid>
<pubDate>Sat, 22 Aug 2009 06:42:35 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22906586</link>
<description><![CDATA[<A HREF="/useremail/u/890631"><b>OmagicQ</b></A> : what's the IP address for GV for this to work?<br><br>I tried myGVnumber@216.239.37.15:5061 and got a request timeout.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22906586</guid>
<pubDate>Fri, 21 Aug 2009 22:35:59 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22905929</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  dyrewolfe <A HREF="/useremail/u/1663559"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>caseydoug, it is working for me on two GV accounts. Same method as wifi4milez except using port 5061 (I haven't tried 5060).<br> </div>Waddya know!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22905929</guid>
<pubDate>Fri, 21 Aug 2009 19:56:06 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22905168</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : Sorry I also used 5061, 5060 was a typo. ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22905168</guid>
<pubDate>Fri, 21 Aug 2009 17:01:31 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22905158</link>
<description><![CDATA[<A HREF="/useremail/u/1663559"><b>dyrewolfe</b></A> : caseydoug, it is working for me on two GV accounts. Same method as wifi4milez except using port 5061 (I haven't tried 5060).]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22905158</guid>
<pubDate>Fri, 21 Aug 2009 17:00:08 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22904476</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : I tried it briefly, and it didn't work. But then it didn't work for GC either, so something may not be configured right (my LAN, etc.). It's been a long time since I had it working, I've changed a lot of stuff, and I didn't put much effort into it.<br><br>I'd be interested if anyone else has had any success recently. I can't think of a reason this would work for one person and nobody else.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22904476</guid>
<pubDate>Fri, 21 Aug 2009 14:55:49 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22903944</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>Am I really the only one who can still reach GV via SIP?<br> </div>You are saying you can reach GV (not GC) by using a SIP address? I haven't tried it in a while, but everyone else has reported that it stopped working. Is there anything special you are doing?<br> </div>Thats exactly what I am saying. I am doing nothing other than dialing (myGVnumber)@googleIPaddress:5060. This works on both my GV accounts, long since converted from GC.<br><small>--<br>Obama = Jimmy Carter part 2<br>"Secret operations are essential in war; upon them the army relies to make its every move"<br><b>-Sun Tzu-</b> <br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22903944</guid>
<pubDate>Fri, 21 Aug 2009 13:14:35 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22903758</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Am I really the only one who can still reach GV via SIP?<br> </div>You are saying you can reach GV (not GC) by using a SIP address? I haven't tried it in a while, but everyone else has reported that it stopped working. Is there anything special you are doing?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22903758</guid>
<pubDate>Fri, 21 Aug 2009 12:43:13 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22903585</link>
<description><![CDATA[<A HREF="/useremail/u/1663559"><b>dyrewolfe</b></A> : <div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Am I really the only one who can still reach GV via SIP?<br></div>I can reach both of my GV #s via SIP. One of which I activated last week sometime.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22903585</guid>
<pubDate>Fri, 21 Aug 2009 12:14:34 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22903545</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I should point out that both of my Google Voice accounts work fine from a SIP perspective (I switched over from GC months ago). Am I really the only one who can still reach GV via SIP?</div>Perhaps, you are the  <b>ONE</b>, Neo! :D]]></description>
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<pubDate>Fri, 21 Aug 2009 12:07:18 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22903536</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I checked out nerdvittles, and it seems in order to get his solution to work you need a special home grown version of asterisk (that is available for download on his site).</div>Nope. Any asterisk-1.4.x should work with the solution provided by nerdvittles, except when using <A HREF="http://www.pmarks.net/posted_links/google-voice-dialout.agi">Paul Marks GV Dialer Python scripts for AGI</a> that requires asterisk-1.6.x that requires a new Bridge () function to make the outbound call transparent. If you want to use the <A HREF="http://www.pmarks.net/posted_links/google-voice-dialout.agi">Paul Marks GV Dialer Python scripts for AGI</a> to place an outbound call, hang up, and wait for the incoming call, then you can use asterisk-1.4.x too.<br><small>--<br>Mazilo always prays for FREEBIES!<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Fri, 21 Aug 2009 12:06:06 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22903507</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>GV will terminate GC by 9/12/2009  </div>Is this official?</div>Yup. I already received several e-mails more than a week ago. Sorry, it should be shutdown on 9/15/2009 as shown below.<br> <blockquote><small>quote:</small><hr>from&#9;voice-noreply@grandcentral.com<br>to&#9;xxx@xxx.yyy.com<br>date&#9;Wed, Aug 12, 2009 at 5:31 AM<br>subject&#9;GrandCentral is shutting down. Upgrade to Google Voice.<br>&#9;<br>&#9;<br>&#9;<br>Reply<br>&#9;<br>&#9;Follow up message<br>Dear GrandCentral User (xxx):<br><br>We are in the process of shutting down the GrandCentral service as we migrate all users to the new and improved Google Voice. Please upgrade your GrandCentral service to Google Voice now, by logging into your GrandCentral account and following the instructions at the top of your Inbox.<br><br>In one month, on September 15th, we will be turning off the telephony service of GrandCentral, and you will not be able to use any of the GrandCentral features. We will be leaving the GrandCentral website intact so that you may continue to access older voicemail messages for the time being.<br><br>If you would like to continue to use your GrandCentral number, it is important to upgrade to Google Voice now. It's free, has the GrandCentral features you are familiar with, plus several new features, like voicemail transcription, SMS, conference calling, and mobile applications for Blackberry and Android phones. To find out more, go to www.google.com/voice/about.<br><br>Thanks!<br><br>The GrandCentral/Google Voice Team<br><br><center>You've received this mandatory service announcement email to update you about important changes to your GrandCentral account.<br>Google Inc. - 1600 Amphitheatre Parkway, Mountain View, CA 94043</center><hr></blockquote><br><small>--<br>Mazilo always prays for FREEBIES!<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Fri, 21 Aug 2009 12:01:06 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22903300</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>So do we have an easy way to integrate GV (GC in this case) into Asterisk for outbound calls without dialing * or other keys to place calls?</div>If you take a look at the source code of <A HREF="http://thatsmith.com/2009/03/google-voice-add-on-for-firefox">GC Dialer add-on for Firefox</a> (no longer there), it is written in a JavaScript (JS) language that can be easily modified to use as an AGI script, let alone converting to other (scripting) languages, for those who are fluent with a JS language. However, knowing GV will terminate GC by 9/12/2009, this effort may be a waste.<br> </div>I should point out that both of my Google Voice accounts work fine from a SIP perspective (I switched over from GC months ago). Am I really the only one who can still reach GV via SIP?<br><small>--<br>Obama = Jimmy Carter part 2<br>"Secret operations are essential in war; upon them the army relies to make its every move"<br><b>-Sun Tzu-</b> <br></small>]]></description>
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<pubDate>Fri, 21 Aug 2009 11:21:57 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22903293</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Yes, look around for references to the blog by Nerdvittles. I don't use it, but that's been cited a lot as a workable Asterisk solution.<br><br>For those who want a hosted solution, check out the threads on Sip Sorcery, for example this one: &raquo;<A HREF="/forum/r22849519-GV-dial-with-sipsorcery">GV dial with sipsorcery</A>.<br><br>Edit: I'm not sure how either would work with GC.<br> </div>I checked out nerdvittles, and it seems in order to get his solution to work you need a special home grown version of asterisk (that is available for download on his site). I would do that in a flash, however that would mean reformatting my current server and losing all the configuration files. Also, I dont know if NV's version includes a GUI which is critical for me. <br><small>--<br>Obama = Jimmy Carter part 2<br>"Secret operations are essential in war; upon them the army relies to make its every move"<br><b>-Sun Tzu-</b> <br></small>]]></description>
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<pubDate>Fri, 21 Aug 2009 11:20:40 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22902880</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>GV will terminate GC by 9/12/2009  </div>Is this official?]]></description>
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<pubDate>Fri, 21 Aug 2009 10:00:50 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22902685</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>So do we have an easy way to integrate GV (GC in this case) into Asterisk for outbound calls without dialing * or other keys to place calls?</div>If you take a look at the source code of <A HREF="http://thatsmith.com/2009/03/google-voice-add-on-for-firefox">GC Dialer add-on for Firefox</a> (no longer there), it is written in a JavaScript (JS) language that can be easily modified to use as an AGI script, let alone converting to other (scripting) languages, for those who are fluent with a JS language. However, knowing GV will terminate GC by 9/12/2009, this effort may be a waste.<br><small>--<br>Mazilo always prays for FREEBIES!<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Fri, 21 Aug 2009 09:19:45 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22901897</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Yes, look around for references to the blog by Nerdvittles. I don't use it, but that's been cited a lot as a workable Asterisk solution.<br><br>For those who want a hosted solution, check out the threads on Sip Sorcery, for example this one: &raquo;<A HREF="/forum/r22849519-GV-dial-with-sipsorcery">GV dial with sipsorcery</A>.<br><br>Edit: I'm not sure how either would work with GC.]]></description>
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<pubDate>Fri, 21 Aug 2009 00:39:33 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22901204</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : So do we have an easy way to integrate GV (GC in this case) into Asterisk for outbound calls without dialing * or other keys to place calls?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22901204</guid>
<pubDate>Thu, 20 Aug 2009 21:52:16 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22901158</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Yes, it still works with GC, but not with GV.]]></description>
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<pubDate>Thu, 20 Aug 2009 21:45:35 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22900839</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : I am able to dial (myGVnumber)@216.239.37.15:5061 and it works fine. All my GV assigned phones ring, and if I dont answer it goes to VM. Perhaps this is because I have an old GC account. ]]></description>
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<pubDate>Thu, 20 Aug 2009 20:47:33 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22900819</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  wifi4milez <A HREF="/useremail/u/1054326"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Any updates on this? <br> </div>On what, exactly? GV still blocks inbound SIP connections, if that's what you're asking.]]></description>
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<pubDate>Thu, 20 Aug 2009 20:44:28 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22900668</link>
<description><![CDATA[<A HREF="/useremail/u/1054326"><b>wifi4milez</b></A> : Any updates on this? ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22900668</guid>
<pubDate>Thu, 20 Aug 2009 20:13:39 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22889945</link>
<description><![CDATA[<A HREF="/useremail/u/890631"><b>OmagicQ</b></A> : When I added my IPKall did to GV, I had to enable RFC2833 on X-lite for GV to recognize the 2 digit code. Afterwards I turned it off. I prefer In-band DTMF because from my experience GV doesn't touch it. ]]></description>
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<pubDate>Wed, 19 Aug 2009 01:06:38 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22889816</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  OmagicQ <A HREF="/useremail/u/890631"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>GV intercepts out of band DTMF for some of their in-call features, so that could be what your issue was and why in-band DTMF fixed it.<br> </div>Interesting. I tried several numbers with IVR menus, and all worked. I wonder, however, if I were now to try calling GV from this phone to pick up voicemail, for example, whether I might have a problem. I'll give it a try later.<br><br>Edit: I tried calling my GV number, and it rang many times with no opportunity to get into my voicemail. That's not surprising given that I'm essentially trying to bridge GV in with GV out through Sip Sorcery.]]></description>
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<pubDate>Wed, 19 Aug 2009 00:25:58 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22889616</link>
<description><![CDATA[<A HREF="/useremail/u/890631"><b>OmagicQ</b></A> : GV intercepts out of band DTMF for some of their in-call features, so that could be what your issue was and why in-band DTMF fixed it.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22889616</guid>
<pubDate>Tue, 18 Aug 2009 23:33:50 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22887038</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : That is pretty much what I do, with two exceptions. First, I do not register anything to G5 because of an incompatibility between G5 and my ATA having nothing to do with GV. Instead, I <i><b>forward</i></b> G5 to myusername@sipsorcery.com. That works fine.<br><br>Second, I adapted a dialplan from mxnerd (&raquo;<A HREF="http://mysipswitch.com/forum/viewtopic.php?p=8892" >mysipswitch.com/forum/viewtopic.php?p=8892</A>) which supports seven, ten, and eleven digit dialing.<br><br>The one weakness I have found in this system is that DTMF is a bit spotty. Sometimes it works, sometimes it doesn't. I suspect this is a GV issue, since I don't think that either G5 or Sip Sorcery touch the audio. Any thoughts on this?<br><br>Edit: I was able to fix the DTMF issue (I think) by switching to Inband on my ATA. The setting is unavailable on the ATA's browser interface, and I had forgotten that it can be accessed via the ATA's command line interface. A little research and I found my own post from years ago that explained how to change this setting. Good thing I write things down in this forum!]]></description>
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<pubDate>Tue, 18 Aug 2009 15:56:41 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22886603</link>
<description><![CDATA[<A HREF="/useremail/u/1461319"><b>emoci</b></A> : As it stands (provided you do not have your own PBX and are not happy with a little software on your PC) your best bet is:<br><br>Incoming SIP:<br>-Get a 1-747 G5 Number<br>-Point your GV to your G5 number<br>-Register your G5 Number on your ATA/SoftPhone/Agregator Service)<br><br>Outgoing SIP:<br>-Get a SipSorcery acct.<br>-Use either the G5 number from above (this is ideal if you want to register to SIPSorcery and be done with it) or another G5 number.<br>-Create secondary extension in SIP Sorcery (eg. user-gv)<br>-Register G5 Number to SipSorcery (eg. in this case the contact field would be user-gv@sipsorcery.com )<br>-Add the following dialplan as the 'Outgoing Dial Plan' for the extension:<br><br><textarea name="code" class="text" cols=50 rows=10>#Ruby&#012; &#012;sys.log("***Starting New Call Event***")&#012;sys.log("***Outgoing Call Starting***")&#012;sys.log("Call placed by  #{req.Header.From.FromName} at #{req.Header.From.FromURI.User}@#{req.Header.From.FromURI.Host}")&#012;sys.log("Call is for #{req.URI.User}")&#012;caller = req.Header.From.FromURI.User&#012;called = req.URI.User&#012;sys.log("*** Caller: #{caller} ***")&#012;sys.log("*** Number Called: #{called} ***")&#012; &#012;sys.log("Call Placed via Google Voice")&#012;sys.GoogleVoiceCall("user@gmail.com", "password", "1747xxxxxxx", "#{called}")&#012;</textarea><!--end code block--><br>Then simply register an ATA into SipSorcery<br>]]></description>
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<pubDate>Tue, 18 Aug 2009 14:37:17 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22886563</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  AllThumbs <A HREF="/useremail/u/1324679"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Just run the install script on any clunker PC you have lying around. </div>That's assuming you want to keep a PC running 24/7. If you wouldn't be running the PC otherwise, the electricity costs alone will quickly eat up any savings.<br><br>The Sip Sorcery solution, discussed elsewhere, does not require installing or running anything on your computer. Even that, however, requires that you have an ATA and a broadband connection. And I certainly wouldn't count on it being around very long. "Free" seems to mean different things to different people.]]></description>
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<pubDate>Tue, 18 Aug 2009 14:29:04 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22886508</link>
<description><![CDATA[<A HREF="/useremail/u/1324679"><b>AllThumbs</b></A> : Next best thing to free SIP calling: totally free PBX with totally free calls in and out (dialed like you normally do) using Google Voice. Just run the install script on any clunker PC you have lying around.<br><br>&raquo;<A HREF="http://nerdvittles.com/?p=637" >nerdvittles.com/?p=637</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22886508</guid>
<pubDate>Tue, 18 Aug 2009 14:19:09 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22884877</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  underworld <A HREF="/useremail/u/1666901"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Here's an updated step by step guide to enable SIP on Google Voice &raquo;<A HREF="http://truvoipbuzz.com/2009/08/how-to-enable-sip-dialing-with-google-voice-tutorial/" >truvoipbuzz.com/2009/08/how-to-e&middot;&middot;&middot;utorial/</A></div>Google Voice doesn't support a SIP outbound call. The link suggested to use GVOut.exe program as a proxy server to place a non-SIP GV outbound call.]]></description>
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<pubDate>Tue, 18 Aug 2009 08:59:01 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22884002</link>
<description><![CDATA[<A HREF="/useremail/u/1666901"><b>underworld</b></A> : Here's an updated step by step guide to enable SIP on Google Voice &raquo;<A HREF="http://truvoipbuzz.com/2009/08/how-to-enable-sip-dialing-with-google-voice-tutorial/" >truvoipbuzz.com/2009/08/how-to-e&middot;&middot;&middot;utorial/</A>]]></description>
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<pubDate>Mon, 17 Aug 2009 23:53:40 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22799976</link>
<description><![CDATA[<A HREF="/useremail/u/1518597"><b>dalrun</b></A> : This is getting a bit of track because the GV number is a "real US number" and I'm guessing it could be SIP forwarded to CallWithUs via Gizmo.<br><br>There's basically two options: GV > POTS (IPKall) > SIP and GV > SIP (Gizmo) > SiP]]></description>
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<pubDate>Sat, 01 Aug 2009 17:28:51 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22797692</link>
<description><![CDATA[<A HREF="/useremail/u/1572525"><b>PX Eliezer</b></A> : <div class="bquote"><small>said by mat :</small><br><br>Why do you need the IPKAll in between? Not sure about future nine, but with callWithUs you get a real US number for receiving 2000 minutes per month free.<br> </div>I think that you are confused.<br><br>CallWithUs no longer offers US "DID" numbers to US residents.<br><br>And in any event, NONE of their DID's are free. <br><br>CWU is an excellent outfit, but they don't give out free US numbers.]]></description>
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<pubDate>Sat, 01 Aug 2009 00:19:53 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22797621</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Why do you need the IPKAll in between? Not sure about future nine, but with callWithUs you get a real US number for receiving 2000 minutes per month free.]]></description>
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<pubDate>Sat, 01 Aug 2009 00:02:35 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22795230</link>
<description><![CDATA[<A HREF="/useremail/u/1663084"><b>jerryscuba</b></A> : I think I found a way of using Google Voice to call the US for free. <br>One can open an account with a VOIP provider that offers Pay As You Go. One provider I know of is Future-Nine.<br>Then one can get a free US number from IPKall and link it with the SIP account. Then one can use this number in GV to get calls forward to.<br>Well an than there is this little inconvenient call button on GV. GV will call your US number and then the party you wish to call within the US for free.<br><br>I did not try exactly this solution as I have an official US number already. However, it worked with my current VOIP provider with my own US number.<br><br>Good luck!<br>JS]]></description>
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<pubDate>Fri, 31 Jul 2009 14:40:34 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22595228</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : My attitude is ABG - Anything But Google.<br><br>Sure, I'll use them if there is no alternative: Google Voice is nice for free calling for example, but if somebody has something as good, and they usually do, then I will always choose it over whatever Google is pushing.<br><br>There are a dozen search engines that will do as good a job as Google's for anything I will need. I used ask.com for several years without a problem and am now giving Bing a try.<br><br>I hope Microsoft cleans Google's clock.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22595228</guid>
<pubDate>Tue, 23 Jun 2009 07:07:06 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22549779</link>
<description><![CDATA[<A HREF="/useremail/u/1639850"><b>josephf</b></A> : Anyone with any proper business acuteness should be worried when a competitor introduces a new product. I don't think that has detracted from Google introducing non-search products in the past. (In fact they have separate teams dedicated to different products. I doubt Craig Walker will now be working on search...)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22549779</guid>
<pubDate>Sun, 14 Jun 2009 15:07:28 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22549705</link>
<description><![CDATA[<A HREF="/useremail/u/1572525"><b>PX Eliezer</b></A> : This story notes that the Google leadership is really worried about Microsoft's Bing! search engine.<br><br>&raquo;<A HREF="http://news.cnet.com/8301-10805_3-10264417-75.html" >news.cnet.com/8301-10805_3-10264417-75.html</A><br><br>&raquo;<A HREF="http://www.nypost.com/seven/06142009/business/fear_grips_google_174235.htm" >www.nypost.com/seven/06142009/bu&middot;&middot;&middot;4235.htm</A><br><br>Considering the importance of Search to Google's whole business, I bet that they backburner GV and other non-core projects.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22549705</guid>
<pubDate>Sun, 14 Jun 2009 14:35:18 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22537052</link>
<description><![CDATA[<A HREF="/useremail/u/1021902"><b>jeffnyc</b></A> : Are outgoing calls using any free sip-uri to GV dead, or has a workaround been found?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22537052</guid>
<pubDate>Thu, 11 Jun 2009 20:37:39 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22400674</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Hi Lackluster,<br><br>I have been using the basic GV incoming and Gizmo5 call out credit for outgoing calls from my innomedia ATA. After reading through this thread I wanted to follow the nerdvittles advice (1) configuring your Google Voice number to go directly to voicemail, (2) configuring your Gizmo5 number to forward all calls and send them via SIP to your Google Voice phone number, <b>(3) configuring FreePBX to route all calls with a GV prefix to your Gizmo5 SIP URI, and (4) configuring Asterisk to jump through the autodialer hoops to place an outbound call to any U.S. number through the Google Voice telephone interface.</b><br><br>I created an account on pbxes.com but I need some help to set it up with proper information for the steps 3 and 4. Can you please let me know how you set it up? Appreciate your help.<br><br>Thanks<br>Ik.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22400674</guid>
<pubDate>Sun, 17 May 2009 00:29:02 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22367497</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  mbuugg <A HREF="/useremail/u/1542856"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>GV now does not use the offical Gizmo proxy (proxy01.sipphone.com) any more, instead, GV routes calls to proxy 198.65.166.147 (probably an internal proxy of Gizmo that was just set up to counteract this SIPping trick, remember the GV->Gizmo glitch on Thu or Fri?), and then the proxy 198.65.166.147 forwards calls to the official Gizmo proxy proxy01.sipphone.com.</div>Was this a coincidence that Navid (a Gizmo5 Support) posted <A HREF="http://www.dslreports.com/forum/r22360494-">here</a> (on Friday 5/8/2009) asking if anyone needs helps on G5 services?<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Sun, 10 May 2009 14:29:45 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22365384</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Aahh, Bassackwards.  :)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22365384</guid>
<pubDate>Sat, 09 May 2009 21:49:21 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22365383</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  mbuugg <A HREF="/useremail/u/1542856"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I took a look of the SIP trace of the incoming GV->Gizmo call, and it looks like the trick is now officially dead. GV now does not use the offical Gizmo proxy (proxy01.sipphone.com) any more, instead, GV routes calls to proxy 198.65.166.147 (probably an internal proxy of Gizmo that was just set up to counteract this SIPping trick, ... </div>Thanks for this information. This means other SIP -> GV shall still work.<br><br><div class="bquote">...remember the GV->Gizmo glitch on Thu or Fri?), and then the proxy 198.65.166.147 forwards calls to the official Gizmo proxy proxy01.sipphone.com. In another word, GV most probably has blocked traffic from proxy01.sipphone.com.</div>You could be right.<br><br><div class="bquote">Isn't GV or Grandcentral using Asterisk, or anything similar? Why don't they just block the incoming Invite from proxy01.sipphone.com but go the long way to set up another proxy?</div>Hush ..., don't give them ideas. They probably didn't know this. ;)<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22365383</guid>
<pubDate>Sat, 09 May 2009 21:49:10 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22365258</link>
<description><![CDATA[<A HREF="/useremail/u/1542856"><b>mbuugg</b></A> : Hi emoci, you can use 'p' for "waiting for answering" and 'w' for "waiting for half second" with pbxes.org. It is useful in the trunk outbound dialing prefix. You don't need a premium account to do this.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22365258</guid>
<pubDate>Sat, 09 May 2009 21:15:49 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22365232</link>
<description><![CDATA[<A HREF="/useremail/u/1542856"><b>mbuugg</b></A> : <div class="bquote"><small>said by LHM :</small><br><br>Gizmo SIP connect to GV not working as of last night (5/8/09)? Confirm?<br> </div>I tried it just now and the call was not connecting. I took a look of the SIP trace of the incoming GV->Gizmo call, and it looks like the trick is now officially dead. GV now does not use the offical Gizmo proxy (proxy01.sipphone.com) any more, instead, GV routes calls to proxy 198.65.166.147 (probably an internal proxy of Gizmo that was just set up to counteract this SIPping trick, remember the GV->Gizmo glitch on Thu or Fri?), and then the proxy 198.65.166.147 forwards calls to the official Gizmo proxy proxy01.sipphone.com. In another word, GV most probably has blocked traffic from proxy01.sipphone.com. <br><br>So, another party pooped. Enjoy it while it lasts!<br><br>Isn't GV or Grandcentral using Asterisk, or anything similar? Why don't they just block the incoming Invite from proxy01.sipphone.com but go the long way to set up another proxy? I am no expert but someone may know the reason.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22365232</guid>
<pubDate>Sat, 09 May 2009 21:09:26 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22364301</link>
<description><![CDATA[<A HREF="/useremail/u/1628469"><b>lacklusterbb</b></A> : caseydoug,<br><br>I simply followed the instructions on nerdvittles, using a hosted pbxes account not an asterisk server.  I was already usng pbxes.com so I had a ring group set up and didn't need a new one.  Following the instructions nerdvittles gives, you've got to set up an extension, trunk (again, you don't need to use 2 Gizmo accounts, I used an existing free digits account I had since they allow free SIP URI dialing in the account information for the trunk, and I plugged the dial plan in the instructions into the dial rules area of the trunk) and the outbound route.  You point your ATA to pbxes.org to register it, using your SIP userid, which you'll see in the dial box under "Device Options" in extensions, and password.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22364301</guid>
<pubDate>Sat, 09 May 2009 16:36:30 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22364278</link>
<description><![CDATA[<A HREF="/useremail/u/1572251"><b>cnr1109</b></A> : Yes, once again, unfortunately.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22364278</guid>
<pubDate>Sat, 09 May 2009 16:31:01 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22364201</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Last time they killed the SIP connection, they left one address (I believe it was .17) pointing to GrandCentral. Of course, you couldn't do anything with it.<br><br>I'd still like a more detailed explanation of what lacklusterbb did with pbxes. Which service was the trunk, which the extension, where did your ATA register, and how were outbound and inbound routes set up? Even if it doesn't work now, it might in the future.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22364201</guid>
<pubDate>Sat, 09 May 2009 16:08:36 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22364177</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : <div class="bquote"><small>said by LHM :</small><br><br>Gizmo SIP connect to GV not working as of last night (5/8/09)? Confirm?<br> </div>Same here.  I thought it was just me.  Interestly, IP ending in 18-20 all rang grand central and I was able to leave a voice mail there.  Any chance google voice was moved to a different range?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22364177</guid>
<pubDate>Sat, 09 May 2009 15:58:52 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22364167</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Perhaps they only changed the ip address. Can you determine if they have or not?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22364167</guid>
<pubDate>Sat, 09 May 2009 15:56:05 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22364153</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by LHM :</small><br><br>Gizmo SIP connect to GV not working as of last night (5/8/09)? Confirm?</div>At the moment, it looks like already dead. :(]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22364153</guid>
<pubDate>Sat, 09 May 2009 15:51:37 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22364060</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Gizmo SIP connect to GV not working as of last night (5/8/09)? Confirm?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22364060</guid>
<pubDate>Sat, 09 May 2009 15:18:18 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22363953</link>
<description><![CDATA[<A HREF="/useremail/u/1461319"><b>emoci</b></A> : <div class="bquote"><small>said by  lacklusterbb <A HREF="/useremail/u/1628469"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>caseydoug, emoci,<br><br>I use a free pbxes account.  I just tried an outgoing call and it still works.  This is different from the first set of instructions that nerd vittles gave out on general SIP dialing; it's based upon GV's integrated SIP dialing via Gizmo.  Google would have to lock down SIP dialing with Gizmo, which I don't see them doing at this point--then GV would truly just be a virtual number service and not much different from many others (except for the fact that there aren't many charges right now while it's beta).  <br> </div>My interest is in the specifics of the setup at PBXes...since you can't exactly create a custom extension...<br><br>I did find out last night that you can use 'p' as a pause character and got up to a point...but not functional yet...<br><br>Any insight on how you set up at PBXes...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22363953</guid>
<pubDate>Sat, 09 May 2009 14:46:53 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22363879</link>
<description><![CDATA[<A HREF="/useremail/u/1273917"><b>N9MD</b></A> : <div class="bquote"><small>said by  lacklusterbb <A HREF="/useremail/u/1628469"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>...I used a free digits account in the account information since they allow free SIP URI dialing... </div>Caveat: FreeDigits is currently functioning only under its TalkDigits incarnation --- althought the SIP-URL is still freedigits.net.  Nobody knows if the service will be viable in the near or distant future.  With that said, another SIP-URI option is a free account with CallCentric.  When forwarding to CC, the incoming CC SIP-URI should read 1777xxxxxxx@callcentric.com (using the CC assigned UserID) and the user selected CC password.  The outgoing SIP-URI from the redirected provider must show 1777xxxxxxx@<b>in.</b>callcentric.com. (Note the placement of "in." in front of the CC domain.)]]></description>
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<pubDate>Sat, 09 May 2009 14:22:50 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22363857</link>
<description><![CDATA[<A HREF="/useremail/u/1628469"><b>lacklusterbb</b></A> : No, it existed in Grand Central.  It was the only way to get a Grand Central/GV number to ring an ATA directly without routing to an IPKALL or other DID affiliated with  a VOIP service registered to the device.  Google has retained the relationship with Gizmo; nerd vittles just gives instructions on using Gizmo's SIP URI dialing with GV.  You don't need to use 2 Gizmo accounts.  In setting up the G5-GV trunk,  I used a free digits account in the account information since they allow free SIP URI dialing. You follow the nerd vittles instructions and plug the daling info into the trunk dialing rules.  You have to set up an extension and inbound routing following the instructions given by nerd vittles. ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22363857</guid>
<pubDate>Sat, 09 May 2009 14:12:55 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22363733</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  lacklusterbb <A HREF="/useremail/u/1628469"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>it's based upon GV's integrated SIP dialing via Gizmo.</div>Is this a new feature?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22363733</guid>
<pubDate>Sat, 09 May 2009 13:37:38 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22363721</link>
<description><![CDATA[<A HREF="/useremail/u/1628469"><b>lacklusterbb</b></A> : caseydoug, emoci,<br><br>I use a free pbxes account.  I just tried an outgoing call and it still works.  This is different from the first set of instructions that nerd vittles gave out on general SIP dialing; it's based upon GV's integrated SIP dialing via Gizmo.  Google would have to lock down SIP dialing with Gizmo, which I don't see them doing at this point--then GV would truly just be a virtual number service and not much different from many others (except for the fact that there aren't many charges right now while it's beta).  ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22363721</guid>
<pubDate>Sat, 09 May 2009 13:34:11 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22362076</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  lacklusterbb <A HREF="/useremail/u/1628469"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>You don't need an Asterisk server to connect to GV, you can use a hosted pbx service like pbxes.com.  I followed these instructions, &raquo;<A HREF="http://nerdvittles.com/index.php?p=593" >nerdvittles.com/index.php?p=593</A>, to set up an extension, trunk and outbound routing.   </div>I don't believe those instructions work any more. You can no longer make a direct call to the GV SIP address, although you <i>can</i> make a SIP call to G5 and have that forwarded via SIP to GV.<br><br>You should be able to do this with both pbxes and Asterisk. The advantage of Asterisk is that you can set up a dial plan to enter the "*, PIN, 2, #." You cannot do that with the free version of pbxes, although you may be able to do it with the paid version.<br><br>However, as lacklusterbb mentioned, GV has been pretty erratic of late, so it's probably not worth investing a lot of time until it settles down.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22362076</guid>
<pubDate>Fri, 08 May 2009 23:48:36 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22361791</link>
<description><![CDATA[<A HREF="/useremail/u/1461319"><b>emoci</b></A> : <div class="bquote"><small>said by  lacklusterbb <A HREF="/useremail/u/1628469"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>You don't need an Asterisk server to connect to GV, you can use a hosted pbx service like pbxes.com.  I followed these instructions, &raquo;<A HREF="http://nerdvittles.com/index.php?p=593" >nerdvittles.com/index.php?p=593</A>, to set up an extension, trunk and outbound routing.  However, I don't use GV much. I find the call quality has deteriorated to the point that more often than not there's so much static I can't hear the other party and they can't hear me.  I've also had calls dropped frequently after only 3-4 minutes.  GV is still very much beta.<br> </div>For the setup with PBXes...do you have a premium acct... or did you manage to do this on the regular free PBXes acct.?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22361791</guid>
<pubDate>Fri, 08 May 2009 22:39:25 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22361049</link>
<description><![CDATA[<A HREF="/useremail/u/1628469"><b>lacklusterbb</b></A> : You don't need an Asterisk server to connect to GV, you can use a hosted pbx service like pbxes.com.  I followed these instructions, &raquo;<A HREF="http://nerdvittles.com/index.php?p=593" >nerdvittles.com/index.php?p=593</A>, to set up an extension, trunk and outbound routing.  However, I don't use GV much. I find the call quality has deteriorated to the point that more often than not there's so much static I can't hear the other party and they can't hear me.  I've also had calls dropped frequently after only 3-4 minutes.  GV is still very much beta.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22361049</guid>
<pubDate>Fri, 08 May 2009 19:34:52 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22352977</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I've been following this thread since it started and had a question. So I've setup my Asterisk box with a custom trunk that dials out using Google Voice using the Gizmo5 SIP forwarding trick. My problem is when using the trunk, I get one way audio (I can hear the external call but the person I called can't hear me). When I dial my Gizmo5 "forwarding" number directly and manually input the *, PIN, 2, phone number, #, everything works as expected with two way audio. Can anybody shed some light into this for me?<br><br>Thanks,<br><br>Jared]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22352977</guid>
<pubDate>Thu, 07 May 2009 09:58:03 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22329747</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I was sort of hoping I could just enter the command "yum install asterisk" and everything would work.</div>I use OpenSuSE and it uses RPM that allows one to install an RPM package on a directory other than /. I am sure YUM does have an option for you to install a YUM package on to your USB memory stick. Speaking about a USB memory stick, perhaps you may want to consider to check out PBX in a Flash (PIAF) and see if it will run on your XO.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Sat, 02 May 2009 15:47:49 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22328891</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>An XO is an AMD base laptop and I don't think you will have any problem to run asterisk or freeswitch on it. If your XO is running a Windows OS, then asterisk ported to a Windows OS may be buggy. In this case, freeswitch will be a better choice since its source can be compiled (right out the box) on a Windows OS.<br> </div>What little I know about Linux, I learned from playing with this machine. It uses a Fedora distribution, but I believe it is somewhat stripped down. The XO has no hard drive and only about a gig of NAND, but it has an SD card slot and can also use a USB flash drive. I can run simple single-file programs from the SD card or the USB flash drive, but I have not figured out how to install more complex programs to those drives, so most programs on the machine run from the internal NAND. <br><br>I was sort of hoping I could just enter the command "yum install asterisk" and everything would work.  :) Are there some low-profile versions out there?]]></description>
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<pubDate>Sat, 02 May 2009 11:51:15 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22328291</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>I am aiming on the device on this <A HREF="http://www.dslreports.com/forum/r21975331-Equipment-Any-way-to-make-this-device-a-VoIP-client">discussion</a> to run either an asterisk or freeswitch to replace my under powered LaFonera FON2100 device.<br> </div>Interesting. I don't have a lot of Linux experience, and I'd like to try to do this without spending <i>any</i> money (I'm not cheap; I just enjoy the sport  :)).</div>Right now, I am trying to migrate from my slow FON2100 unit to a (long discontinued) Netgear WGT634U WiFi router.<br><br><div class="bquote">I do have a one-laptop-per-child XO, which is a low powered Linux machine. The specs are <A HREF="http://laptop.org/en/laptop/hardware/specs.shtml">here</a>. Do you think Asterisk would run on it?</div>An XO is an AMD base laptop and I don't think you will have any problem to run asterisk or freeswitch on it. If your XO is running a Windows OS, then asterisk ported to a Windows OS may be buggy. In this case, freeswitch will be a better choice since its source can be compiled (right out the box) on a Windows OS.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Sat, 02 May 2009 07:52:14 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22327992</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I am aiming on the device on this <A HREF="http://www.dslreports.com/forum/r21975331-Equipment-Any-way-to-make-this-device-a-VoIP-client">discussion</a> to run either an asterisk or freeswitch to replace my under powered LaFonera FON2100 device.<br> </div>Interesting. I don't have a lot of Linux experience, and I'd like to try to do this without spending <i>any</i> money (I'm not cheap; I just enjoy the sport  :)). I do have a one-laptop-per-child XO, which is a low powered Linux machine. The specs are <A HREF="http://laptop.org/en/laptop/hardware/specs.shtml">here</a>. Do you think Asterisk would run on it?<br><br>Oroman: Good idea. I hadn't thought of that, but one of my phone systems does support speed dial and may support pauses; I'll have to check. However, that is not the phone I'm currently using for GV, Gizmo, etc.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22327992</guid>
<pubDate>Sat, 02 May 2009 02:59:49 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22324847</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I have a basic ATA (HT386) that doesn't support pauses.   I put the pauses in my phone's contact list.  Can you do the same?]]></description>
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<pubDate>Fri, 01 May 2009 14:19:53 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22324069</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>OK, then I can confirm that bypassing the PIN has never worked for me (and still doesn't) when accessing GV through SIP.</div>Now, at least I know it wasn't my configuration that doesn't work to bypass the PINs for a SIP2GV forwarding.<br><br><div class="bquote">I also never got Oroman's earlier trick to work, i.e., setting up a Contact with the same phone number as the ring-to phone.</div>This option doesn't work for me neither. :(<br><br><div class="bquote">... or another computer to set up Asterisk.</div>Running an asterisk on a computer for a personal usage is overkill to just wasting your electrical bill, let alone maintenance is more cumbersome as compared to running an asterisk on a Linux embedded device. I am aiming on the device on this <A HREF="http://www.dslreports.com/forum/r21975331-Equipment-Any-way-to-make-this-device-a-VoIP-client">discussion</a> to run either an asterisk or freeswitch to replace my under powered LaFonera FON2100 device.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Fri, 01 May 2009 12:03:36 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22323473</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : OK, then I can confirm that bypassing the PIN has never worked for me (and still doesn't) when accessing GV through SIP. <br><br>I also never got Oroman's earlier trick to work, i.e., setting up a Contact with the same phone number as the ring-to phone. I would really like to do this, since I'm trying to simplify the outbound calling without spending any money on a new ATA or another computer to set up Asterisk.]]></description>
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<pubDate>Fri, 01 May 2009 10:12:01 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22322983</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>..., are you talking about calling your GV number from a Gizmo account that is set up as a ring-to number in GV?</div>No. I merely talked about G5 -> GV through SIP forwarding.<br><br><div class="bquote">the new option to bypass the PIN says it's available for cell phones and Gizmo phones as well as landlines.</div>At least, this is what I thought that calling a GV line from a G5 through a SIP forwarding was included.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Fri, 01 May 2009 08:23:15 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22322459</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by Oroman :</small><br><br>A PIN is not required for land lines.  It's a new option under Phones/Advanced Settings.</div>It doesn't seem to work with forwarding set to Gizmo. Can anyone confirm this?<br> </div>If I try it tonight I'll wake everyone up. However, are you talking about calling your GV number from a Gizmo account that is set up as a ring-to number in GV? In order to do that, I would require Gizmo credits, right? Because I've used Gizmo exclusively for incoming or SIP forwarding, so I don't have any credits.<br><br>Also, Oroman: the new option to bypass the PIN says it's available for cell phones and Gizmo phones as well as landlines. Whether it's really available is yet to be seen.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22322459</guid>
<pubDate>Fri, 01 May 2009 01:56:41 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22322265</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by Oroman :</small><br><br>A PIN is not required for land lines.  It's a new option under Phones/Advanced Settings.</div>It doesn't seem to work with forwarding set to Gizmo. Can anyone confirm this?]]></description>
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<pubDate>Fri, 01 May 2009 00:31:47 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22322213</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Mazilo suggested using a single Gizmo account with call hunting.<br>I tried call forwarding with a single Gizmo account and it worked!  It also connects faster than call hunting.<br><br>A PIN is not required for land lines.  It's a new option under Phones/Advanced Settings.]]></description>
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<pubDate>Fri, 01 May 2009 00:11:59 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22305616</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Actually, it was cnr1109 first indicated on his <A HREF="http://www.dslreports.com/forum/r22257337-">post</a> and I tested/played with it a bit.<br> </div>Either way it is a nasty hack. I guess if you're really cheap or desperate you'll want to use this. GV should just open up its SIP interface to the users.]]></description>
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<pubDate>Tue, 28 Apr 2009 00:04:59 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22300726</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : I have to apologize. When I said GV didn't require a PIN when accessing my account from my cell phone, I was wrong. I simply didn't wait long enough. I heard "You have no new messages," and assumed I had fully accessed my account. In fact, GV asked me for my PIN immediately AFTER telling me I had no new messages.<br><br>It looks like Google is changing this product every few days, probably trying out different configurations before offering it to the general public. I don't think we can rely on anything until this settles down.]]></description>
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<pubDate>Mon, 27 Apr 2009 00:46:36 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22299184</link>
<description><![CDATA[<A HREF="/useremail/u/1572251"><b>cnr1109</b></A> : PIN: Here today...gone tomorrow<br>&raquo;<A HREF="http://www.google.com/support/forum/p/voice/thread?tid=7c67073eb19f2b3e&hl=en" >www.google.com/support/forum/p/v&middot;&middot;&middot;3e&hl=en</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22299184</guid>
<pubDate>Sun, 26 Apr 2009 17:59:17 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22298456</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I'm also getting the security warning, but I can get to HISTORY.  My session hangs when I go to PHONES or GROUPS under SETTINGS.  <br><br>Responding "NO" to the security warning fixed my problem. ]]></description>
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<pubDate>Sun, 26 Apr 2009 14:21:51 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22298202</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I just tried this by enabling GV to ring to my cell phone, and then calling in (I'm away from home now). No * or PIN was required, and I got directly to my account.</div>I reckon this will only work if the incoming calls to GV is through its DID# and not SIP.<br><br><div class="bquote">OT: has anyone noticed that when you click the "History" link on the left side under "Balance," it can't load the screen and causes the page to hang? I am also now getting a security warning ("Do you want to view only the webpage content that was delivered securely") every time I go to my GV home page after logging out. Are others getting this?</div>It works just fine on my GV account.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Sun, 26 Apr 2009 13:16:14 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22297938</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  cnr1109 <A HREF="/useremail/u/1572251"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>PIN: You are referring to phone verification. That is new also. GV is now requiring a PIN at all times when you call in to your own number.<br> </div>I just tried this by enabling GV to ring to my cell phone, and then calling in (I'm away from home now). No * or PIN was required, and I got directly to my account. I assume you have enabled direct access to voicemail in GV settings for your phone? <br><br>OT: has anyone noticed that when you click the "History" link on the left side under "Balance," it can't load the screen and causes the page to hang? I am also now getting a security warning ("Do you want to view only the webpage content that was delivered securely") every time I go to my GV home page after logging out. Are others getting this?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22297938</guid>
<pubDate>Sun, 26 Apr 2009 12:03:18 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22297266</link>
<description><![CDATA[<A HREF="/useremail/u/1572251"><b>cnr1109</b></A> : PIN: You are referring to phone verification. That is new also. GV is now requiring a PIN at all times when you call in to your own number.<br>MJ>GV: Correct, direct SIP connection.<br>VM Greeting: I understand what you did. I was testing this scenario to ultimately forward GV1>GV2. Since the CID would always be changing, I couldn't use this as a solution. I also notice that CID is not passed in GV>G5>GV scenarios.<br>Direct SIP: As mentioned in other posts, the G5>G5>GV scheme does work using SIP only, but not "direct" SIP. ]]></description>
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<pubDate>Sun, 26 Apr 2009 07:32:57 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22296990</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  cnr1109 <A HREF="/useremail/u/1572251"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>PIN authentication was instituted by GV a couple of days ago because of spoofing and fraud; from any phone you are calling from.</div>I assume you are referring to GV's new practice of making you enter a code in order to add a new ring-to phone?<br><br><div class="bquote">Yes, I agree, the subject was SIP>GV directly. (snip)I was testing various call-forwarding scenarios.<br>1)IPK/G5>GV (via G5 website): works/inconsistent<br>2)IPK/G5>GV (via X-Lite): works/inconsistent<br>3)MJ>GV    (via X-Lite): works/consistent  </div>Again, so that I understand what you are doing, only the third scenario is a direct SIP connection, right? There you are entering the GV IP address into your X-Lite while registered to your MJ account?<br><br>In the first two scenarios, you are entering your IPK number, which is PSTN, IPK forwards to G5, and G5 forwards to GV. Several of us have succeeded in getting G5 to forward to GV since Google shut down SIP access, but this is not what I meant by "direct" SIP>GV.<br><br><div class="bquote">I also notice that I get a generic VM greeting when I access my GV number in this manner. Calls directly to my GV number continue to get my personalized VM greeting.<br> </div>Yes, I noticed this also. You can get GV to "recognize" the number by adding a contact with the number you are calling from. Thus, if I add a contact called "Test Contact" with the same 11-digit number as the first part of my G5 number, GV will see it as a call from Test Contact, but not a call from my G5 number. In fact, if I create a personalized greeting for Test Contact, it will play that greeting when I call in (assuming the G5 CID is passed).]]></description>
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<pubDate>Sun, 26 Apr 2009 02:16:31 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22295242</link>
<description><![CDATA[<A HREF="/useremail/u/1253008"><b>andre2</b></A> : Notice I said "public", not "out of beta".  I couldn't care less about whether they choose to call it beta (which is basically meaningless anyway), only about when it's possible for those of us without former GC accounts to actually sign up.<br><br>Edit: I checked the GV and Gmail pages and they have the "BETA" on the Gmail page, but not the GV one, for some reason.]]></description>
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<pubDate>Sat, 25 Apr 2009 17:23:09 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22294898</link>
<description><![CDATA[<A HREF="/useremail/u/1324679"><b>AllThumbs</b></A> : <div class="bquote"><small>said by  andre2 <A HREF="/useremail/u/1253008"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Google is probably using you guys for testing.  When you can't find any more back doors, that's when GV goes public....  <br> </div>Not sure that's quite accurate. After all, Gmail is still beta, and it's been years. A more plausible explanation is that beta = no Google responsibility when things go horribly wrong (as they sometimes do). ]]></description>
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<pubDate>Sat, 25 Apr 2009 16:06:24 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22294209</link>
<description><![CDATA[<A HREF="/useremail/u/1572251"><b>cnr1109</b></A> : PIN authentication was instituted by GV a couple of days ago because of spoofing and fraud; from any phone you are calling from.<br>Yes, I agree, the subject was SIP>GV directly. I don't have access to my ATA at my current location. This is what I have tried, using my magicJack credentials and IPKall DID for testing. I understand calling a MJ or IPK DID defeats the purpose of free outbound calling, but I was testing various call-forwarding scenarios.<br>1)IPK/G5>GV (via G5 website): works/inconsistent<br>2)IPK/G5>GV (via X-Lite): works/inconsistent<br>3)MJ>GV    (via X-Lite): works/consistent<br>I also notice that I get a generic VM greeting when I access my GV number in this manner. Calls directly to my GV number continue to get my personalized VM greeting.]]></description>
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<pubDate>Sat, 25 Apr 2009 12:36:31 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22292959</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  cnr1109 <A HREF="/useremail/u/1572251"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I just tried (216.239.37.12:5061) and it works. It was not working a few days ago. Could this be the result of PIN authentication; which was introduced yesterday?<br> </div>I'm not sure what you mean by PIN authentication (I'm out of town so can't test), but how are you accessing this address? Forwarding from Gizmo, or direct access from any SIP source? If it's forwarding from Gizmo, then several addresses have worked this way, albeit inconsistently. But if you could access GV via SIP without using Gizmo, that's news.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22292959</guid>
<pubDate>Sat, 25 Apr 2009 00:20:41 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22290804</link>
<description><![CDATA[<A HREF="/useremail/u/1572251"><b>cnr1109</b></A> : I've been trying over the last hour...I get through about 75% of the time....obviously not usable. I'm assuming they are still playing with it, or perhaps a G5 issue?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22290804</guid>
<pubDate>Fri, 24 Apr 2009 15:49:33 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22290774</link>
<description><![CDATA[<A HREF="/useremail/u/1003137"><b>garys_2k</b></A> : <div class="bquote"><small>said by  celtic <A HREF="/useremail/u/308189"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  cnr1109 <A HREF="/useremail/u/1572251"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>I just tried (216.239.37.12:5061) and it works. ...<br> </div>I just tried 2 calls using your settings and neither worked.  Anyone else able to make SIP calls directly to Google Voice?<br> </div>Wow, that didn't take long...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22290774</guid>
<pubDate>Fri, 24 Apr 2009 15:44:58 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22290721</link>
<description><![CDATA[<A HREF="/useremail/u/308189"><b>celtic</b></A> : <div class="bquote"><small>said by  cnr1109 <A HREF="/useremail/u/1572251"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I just tried (216.239.37.12:5061) and it works. ...<br> </div>I just tried 2 calls using your settings and neither worked.  Anyone else able to make SIP calls directly to Google Voice?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22290721</guid>
<pubDate>Fri, 24 Apr 2009 15:35:44 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22290275</link>
<description><![CDATA[<A HREF="/useremail/u/1572251"><b>cnr1109</b></A> : I just tried (216.239.37.12:5061) and it works. It was not working a few days ago. Could this be the result of PIN authentication; which was introduced yesterday?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22290275</guid>
<pubDate>Fri, 24 Apr 2009 14:09:41 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22289764</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  andre2 <A HREF="/useremail/u/1253008"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Google is probably using you guys for testing.  When you can't find any more back doors, that's when GV goes public.  (Not that I disapprove of what you're doing - the sooner the bugs are found, the sooner they'll get fixed.)</div>Could be, after all GV is still out on a beta period.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22289764</guid>
<pubDate>Fri, 24 Apr 2009 12:35:01 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22289758</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Actually cnr1109 first pointed out that the Gizmo SIP connection might be working (he referenced the post at &raquo;<A HREF="http://googlevoices.blogspot.com/" >googlevoices.blogspot.com/</A>).</div>That's correct. Nevertheless, I already had this working before I posted my article.<br><br><div class="bquote">What I haven't been able to figure out is whether Google closed that connection and then reopened it, or whether it was never closed. When we lost the SIP connection a couple of weeks ago, I tested it using a number of different proxies, but never tried forwarding from Gizmo until mazilo indicated it was working.</div>Actually, it was cnr1109 first indicated on his <A HREF="http://www.dslreports.com/forum/r22257337-">post</a> and I tested/played with it a bit.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22289758</guid>
<pubDate>Fri, 24 Apr 2009 12:33:32 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22289608</link>
<description><![CDATA[<A HREF="/useremail/u/1253008"><b>andre2</b></A> : Google is probably using you guys for testing.  When you can't find any more back doors, that's when GV goes public.  (Not that I disapprove of what you're doing - the sooner the bugs are found, the sooner they'll get fixed.)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22289608</guid>
<pubDate>Fri, 24 Apr 2009 12:02:34 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22289600</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : That's why I asked my question about whether this was EVER shut down for Gizmo. Look, it's always been a mystery to me why GC/GV allowed Gizmo as their only SIP connection outbound. Perhaps they've always had an inbound SIP connection as well, but nobody discovered it until recently. If that's the case, I could see them closing off all the other connections but leaving this one open for the few geeks that would play with it. I doubt that they are losing a lot of revenue from people who are doing this.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22289600</guid>
<pubDate>Fri, 24 Apr 2009 11:59:50 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22289559</link>
<description><![CDATA[<A HREF="/useremail/u/229804"><b>B</b></A> : Okay, I'll be the jerk one more time, and repeat -- how long before Google shuts down this latest iteration of this sorta-backdoor functionality?  It's lots of fun to hack this stuff, but relying on it (you know, for stuff like critical <i>phone service</i>) is another story entirely.<br><br>I'm guessing that the relatively high profiles of both NerdVittles and DSLR will do nothing but hasten things along, again. :(<br><br>-- B<br><small>--<br>In a realm outside causality and function</small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22289559</guid>
<pubDate>Fri, 24 Apr 2009 11:51:46 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22289517</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Actually cnr1109 first pointed out that the Gizmo SIP connection might be working (he referenced the post at &raquo;<A HREF="http://googlevoices.blogspot.com/" >googlevoices.blogspot.com/</A>). What I haven't been able to figure out is whether Google closed that connection and then reopened it, or whether it was never closed. When we lost the SIP connection a couple of weeks ago, I tested it using a number of different proxies, but never tried forwarding from Gizmo until mazilo indicated it was working. Can anybody confirm that Google had shut this connection down along with the others?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22289517</guid>
<pubDate>Fri, 24 Apr 2009 11:44:49 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22289171</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  AllThumbs <A HREF="/useremail/u/1324679"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Nerd Vittles strikes again with a new recipe for making free U.S. calls with Asterisk. This time they're using Asterisk, Google Voice, and Gizmo. All of the accounts and calls are free, free, free.</div>Yup and I have done this way before I posted <A HREF="http://www.dslreports.com/forum/r22267360-">this</a>, except I didn't bother to post the whole asterisk dialplan context and let NerdVittles do this for us. ;)<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22289171</guid>
<pubDate>Fri, 24 Apr 2009 10:31:43 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22288848</link>
<description><![CDATA[<A HREF="/useremail/u/1324679"><b>AllThumbs</b></A> : Nerd Vittles strikes again with a new recipe for making free U.S. calls with Asterisk. This time they're using Asterisk, Google Voice, and Gizmo. All of the accounts and calls are free, free, free.<br><br>&raquo;<A HREF="http://nerdvittles.com/?p=605" >nerdvittles.com/?p=605</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22288848</guid>
<pubDate>Fri, 24 Apr 2009 09:12:30 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22284801</link>
<description><![CDATA[<A HREF="/useremail/u/1454991"><b>DaveTap</b></A> : You might want to check out pennytel.com for cheaper Australia numbers and calls. I set up a GoldCoast number for  for around $12/yr. They work great with PBXes.com and also can place calls to US for about 1 cent/min so I use as a backup trunk for US calls. If you don't want the annual charge you can now get a free number for a specific phone# to call but you have to "activate" the number as if you own it by answering a call and input a 9 digit authentication code.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22284801</guid>
<pubDate>Thu, 23 Apr 2009 13:29:30 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22267988</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by nick digger :</small><br><br>Didn't work for me since i never had any call-out credit at gizmo.<br> </div>That's the point of forwarding to the GV SIP address: you don't need call-out credits.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22267988</guid>
<pubDate>Mon, 20 Apr 2009 17:38:24 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22267946</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Actually, you can do this with a single (not duo) Gizmo5 account + a GV account. Configure your Gizmo5 account with <i>Call Hunting</i> (If no answer on Gizmo5, ring elsewhere) feature and just call your own Gizmo5 number from your ATA ... </div>Didn't work for me since i never had any call-out credit at gizmo.  Since my phone is next to my pc, and my pc is always on, it's not a huge hassle to just call through the GV website.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22267946</guid>
<pubDate>Mon, 20 Apr 2009 17:31:28 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22267523</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Thanks, mazilo, I'll take a look at it.<br><br>Thanks also for the tip about Call Hunting.<br><br>Earlier in this thread, Oroman posted about being able to avoid the *2PIN requirement by setting up a dummy contact and routing calls from that number to the (disabled) GV number. Have you been able to get that to work with Gizmo? It works fine from my PSTN numbers, but not from Gizmo, possibly because the Gizmo connection is forwarded rather than direct. If you try it, please let me know one way or the other.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22267523</guid>
<pubDate>Mon, 20 Apr 2009 16:25:54 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22267360</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>This feature has another benefit: the ability to make and receive free calls without requiring either a PSTN or a paid SIP account. I set up two free Gizmo5 accounts, one of which is forwarded to GV via SIP for outbound calling, and the other is set up as one of my GV primary numbers for inbound calling. Using a speed dial entry (or Mysipswitch or various other methods), I can dial the forwarding Gizmo number and make a call out using GV.</div>I understood this. Actually, you can do this with a single (not duo) Gizmo5 account + a GV account. Configure your Gizmo5 account with <i>Call Hunting</i> (If no answer on Gizmo5, ring elsewhere) feature and just call your own Gizmo5 number from your ATA configured to your Gizmo5 account. It will probably ring six times before the call gets forwarded to your GV line through SIP. Once on GV, press * and 2 to place outbound PSTN call.<br><br><div class="bquote">One thing I haven't figured out, however, is how to call out without dialing extra digits. I'm using Mysipswitch, which has powerful dial plan scripting, but I don't see how to make it pass the *, 2, PIN, and phone number after the call has been forwarded to GV.</div>May be, <A HREF="http://nerdvittles.com/?p=593">this</a> will help if you are using asterisk PBX system.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22267360</guid>
<pubDate>Mon, 20 Apr 2009 15:59:10 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22266451</link>
<description><![CDATA[<A HREF="/useremail/u/160346"><b>cbrain</b></A> : Google Voice Add-on for Firefox<br><br>&raquo;<A HREF="/forum/r22132527-General-Google-Voice-Addon-for-Firefox#22266184">[General] Google Voice Add-on for Firefox</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22266451</guid>
<pubDate>Mon, 20 Apr 2009 13:24:20 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22264401</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Anyway, I tested this Gizmo5 (SIP) to my Google Voice and it worked. This only will benefit those who have PSTN numbers to forward incoming SIP calls to. <br> </div>Mazilo, I missed your comment yesterday. This feature has another benefit: the ability to make and receive free calls without requiring either a PSTN or a paid SIP account. I set up two free Gizmo5 accounts, one of which is forwarded to GV via SIP for outbound calling, and the other is set up as one of my GV primary numbers for inbound calling. Using a speed dial entry (or Mysipswitch or various other methods), I can dial the forwarding Gizmo number and make a call out using GV. Gizmo won't let me call a PSTN number directly without buying minutes, but it will forward via SIP for free.<br><br>One thing I haven't figured out, however, is how to call out without dialing extra digits. I'm using Mysipswitch, which has powerful dial plan scripting, but I don't see how to make it pass the *, 2, PIN, and phone number after the call has been forwarded to GV. I will post a question over in the MSS forum once they let me in, but I don't have a lot of hope.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22264401</guid>
<pubDate>Mon, 20 Apr 2009 01:36:39 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22263189</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : I've been playing with this a bit more, and it looks like the ONLY way to connect to GV via SIP is by forwarding from Gizmo. I've tried to reach the GV URI directly while registered with several other providers, including Gizmo/Sipphone, and I can't connect. Forwarding from Gizmo seems to work most of the time, but not always.<br><br>I'm not familiar with the various dial plan scripts, but wouldn't this be pretty limiting? That is, if you need to dial Gizmo first (which then forwards to GV), is it possible to create a dial plan that sends "*-PIN-2-DestinationNumber" AFTER dialing Gizmo? If anybody knows how to do this using the Ruby script in MySipSwitch, or any other way, please post it.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22263189</guid>
<pubDate>Sun, 19 Apr 2009 19:59:29 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22263036</link>
<description><![CDATA[<A HREF="/useremail/u/853361"><b>Dude111</b></A> : Google Voice is awesome!!!!!!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22263036</guid>
<pubDate>Sun, 19 Apr 2009 19:25:47 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22258088</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>Anyway, I tested this Gizmo5 (SIP) to my Google Voice and it worked.<br> </div>Really?! I was using a non-Gizmo method to test it, but it didn't work. Which ip did you use?<br><br>Edit: What do you know? It does work, using Gizmo5 and IP .11. I wonder whether they opened this up again, or whether it's always worked using Gizmo. It did NOT work for me using other accounts, so perhaps it recognizes my Gizmo number.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22258088</guid>
<pubDate>Sat, 18 Apr 2009 13:53:23 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22257543</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  cnr1109 <A HREF="/useremail/u/1572251"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Am I missing something? I thought Google killed this. The following was posted yesterday, even with disclaimer.<br><br>&raquo;<A HREF="http://googlevoices.blogspot.com/" >googlevoices.blogspot.com/</A></div>I believe GV has already patched the hole to turn off this feature. Otherwise, can you imagine how much GV has to pay all SIP2PSTN calls.<br><br>Anyway, I tested this Gizmo5 (SIP) to my Google Voice and it worked. This only will benefit those who have PSTN numbers to forward incoming SIP calls to. For instance, you can configure all your SIP accounts to forward unanswered incoming calls to your celphone through GC/GV using this approach. So, you can be somewhere else to receive your incoming SIP calls through your cellphone and/or other PSTN phone.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22257543</guid>
<pubDate>Sat, 18 Apr 2009 11:33:01 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22257519</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : I tried it, and it still doesn't work. The guy is just behind the times. He still thinks Coleman won in Minnesota.  :)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22257519</guid>
<pubDate>Sat, 18 Apr 2009 11:26:20 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22257337</link>
<description><![CDATA[<A HREF="/useremail/u/1572251"><b>cnr1109</b></A> : Am I missing something? I thought Google killed this. The following was posted yesterday, even with disclaimer.<br><br>&raquo;<A HREF="http://googlevoices.blogspot.com/" >googlevoices.blogspot.com/</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22257337</guid>
<pubDate>Sat, 18 Apr 2009 10:35:25 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22180553</link>
<description><![CDATA[<A HREF="/useremail/u/535779"><b>dkmcknight</b></A> : I use both Skype AND Google Voice. <br><br>Right now, I'm on a Skype Unlimited US & Canada subscription ($2.95/month, which is dirt cheap). Their Unlimited World plan is $9.95/month, which connects you to 36 countries (with some limitations on cell phone access in most countries). <br><br>I'm working with a partner in Australia, so bought a Melbourne DID through Skype ($30/year) -- so when he calls me on the Melbourne number it's forwarded to my GV number. No additional Skype cost for that call, since the inbound number terminates at a US number, which is included in my US/Canada subscription. He's going to do the same thing going the other direction, so I'll be able to call a US number, and it will ring to him in Melbourne.<br><br>I also have my GV <i>and </i> Skype numbers in my cell phone service calling plan so I can call to, or receive calls from, either number, and it doesn't count against my cell minutes. I'm pretty pleased with what I have.<br><br>Right now, though, neither service offers <i>everything</i> I need. Both, combined, get pretty close (for very little money).<br><br>I have an office number that's with another VoIP provider, that I would move in a New York minute to <i>either</i> Skype or GV -- but unfortunately neither allow number porting, yet.<br><br>The first provider that offers GV features, SIP, number portability and Skype pricing gets 100% of my business. I'll consolidate. In the meantime, I don't mind using them both.<br><small>--<br>...And don't forget to vote. <i>With your wallet.</i></small>]]></description>
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<pubDate>Fri, 03 Apr 2009 22:13:13 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22176517</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I use skype connecting my comp trhu an adapter to my PBX at home, which gives me phone service, domestic and international.<br><br>Unfortunately you have to pay for domestic calls also, so I use it mainly for international calls.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22176517</guid>
<pubDate>Fri, 03 Apr 2009 11:23:51 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22158131</link>
<description><![CDATA[<A HREF="/useremail/u/229804"><b>B</b></A> : Not sure what Ward's smoking over there, but this sudden lockdown was easily foreseen (and was ;) ).  The only mystery to me was why the smart folks at NerdVittles would take so much time to research and publish an exploit that was doomed from the start.<br><br>Yes, it would be lovely if Google Voice became a more open SIP-compliant platform as the article hopes, but I don't think Google, or GrandCentral before it, made any mention of its internal architecture or opening that up -- they likely use SIP compliant equipment out of de facto convenience.  I mean, if GrandCentral servers were SIP compliant all along and they were open to the idea of an open architecture, why did they even bother getting Gizmo involved?<br><br>Still, I hope he's right this time.<br><br>-- B<br><small>--<br>In a realm outside causality and function</small>]]></description>
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<pubDate>Tue, 31 Mar 2009 12:11:33 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22153577</link>
<description><![CDATA[<A HREF="/useremail/u/1324679"><b>AllThumbs</b></A> : I kinda doubt it: <A HREF="http://nerdvittles.com/?p=597">Google Voice: Is the SIP and Asterisk Honeymoon Over?</a>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22153577</guid>
<pubDate>Mon, 30 Mar 2009 16:12:03 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22152384</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  mazilo <A HREF="/useremail/u/637921"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>This is true; however, GC has disabled outbound PSTN calls using this approach.<br></div>Was this option available in GC before GV was introduced? Does it still work for accounts that have not yet been converted?<br><br><div class="bquote">I believe GC will not allow its calls forwarded to another GC number. I don't know if Google has lifted this option off of a GV service. Anyone?<br></div>When I tried to in GC to forward to my own GC/GV number, I saw a warning that GC cannot be forwarded to other GC numbers, but I assumed it was refusing to forward to MY OWN GC number. Are you saying that GC could not be forwarded to ANY GC number? I guess this means that it would not let you define another GC number as a "ring to" number? And if that is the case, I suspect that GV would be the same.<br><br>I wonder whether all of these restrictions are in preparation for Google's introduction of a real VoIP service, or what else they have in mind. With no LNP and no SIP access (i.e., no real ability to give up my existing PSTN and VoIP services), I'm unlikely to make a lot of use of GV.]]></description>
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<pubDate>Mon, 30 Mar 2009 13:01:13 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22151488</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>By the way, I have noticed that you can press * almost as soon as the phone starts ringing to get the PIN prompt.</div>Make sure you check if your phone is already on your GC contact list. On my GC/GV account, if the phone I used to call my GC/GV account isn't listed on its contact list, then pressing * will not switch to GC/GV IVR until the VM takes over. :(<br><br><div class="bquote">As celtec points out, if you have converted to GV, you can still reach your GC number via SIP using the IP address he lists.</div>This is true; however, GC has disabled outbound PSTN calls using this approach.<br><br><div class="bquote">If I had two GC accounts and converted only one of them, I believe I would have been able to reach GC via SIP, and then forward that call for free to my GV number.</div>I believe GC will not allow its calls forwarded to another GC number. I don't know if Google has lifted this option off of a GV service. Anyone?<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Mon, 30 Mar 2009 10:40:29 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22150258</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Yes, I understand. I was just hoping you had figured a way to get around this block and reach your GV number via SIP. By the way, I have noticed that you can press * almost as soon as the phone starts ringing to get the PIN prompt.<br><br>As celtec points out, if you have converted to GV, you can still reach your GC number via SIP using the IP address he lists. In fact, while I can no longer reach my GV number via SIP, SIP is the ONLY way I can reach GC, for example to leave a voice mail message. But there is not much I else I can do with GC: the settings tab is unavailable, so I can't add or change "ring to" numbers. When I call GC via SIP, the call forwards to the numbers selected in the last setting I had before converting to GV. The only other setting that sort of works is the "Quick Rule," which allows me to forward to voice mail, but I cannot forward to any other number (I think this is the case; I have been unable to test it fully). <br><br>If I had two GC accounts and converted only one of them, I believe I would have been able to reach GC via SIP, and then forward that call for free to my GV number. If someone still has a working GC account and a GV account, I'd be interested to know whether this can be done. (I still have an unused GC invite, but it won't allow me to create a new GC account.)]]></description>
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<pubDate>Mon, 30 Mar 2009 01:12:31 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22149775</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>mazilo, are you reaching your GV number over PSTN?</div>Yes. First, you need to make sure the phone (PSTN / VoIP / Cell) you will be using to call your own GV number must be listed on your GV contact list. Otherwise, you will need to let GV voicemail to intercept the call before you can press the * key for it to prompt you for your PIN numbers. Once you have correctly entered your PIN numbers, you can press 2 to place an outbound call.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
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<pubDate>Sun, 29 Mar 2009 22:33:53 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22148230</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : mazilo, are you reaching your GV number over PSTN?<br><br>I had never done much with my GC number, so I didn't realize there was previously a call out option. GC does still let me change temporary ring to numbers by calling in. I haven't tested this because I don't have a temporary number to use. I was unable to call out using the GC site no matter what number I tried, however. It said I needed funds, but when I added funds, it still didn't work.]]></description>
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<pubDate>Sun, 29 Mar 2009 16:14:19 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22148192</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>This address provides SIP access to my GC number, but that number does not let me call out. Because I have converted to GV, I can't even change the "ring to" settings on my GC account.</div>When I called my GC account, option #2 seemed to have been taken offline by GC. So, SIP hacks doesn't work on my GC account. OTOH, I can still place an outbound call when I call my GV number with option #2.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22148192</guid>
<pubDate>Sun, 29 Mar 2009 16:07:08 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22147946</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  celtic <A HREF="/useremail/u/308189"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>I can still send SIP calls to 216.239.37.17:5061 to ring my GrandCentral numbers.<br> </div>Shhh!  ;)<br><br>Edit: I just got it. This address provides SIP access to my GC number, but that number does not let me call out. Because I have converted to GV, I can't even change the "ring to" settings on my GC account.]]></description>
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<pubDate>Sun, 29 Mar 2009 14:53:27 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22147928</link>
<description><![CDATA[<A HREF="/useremail/u/308189"><b>celtic</b></A> : I can still send SIP calls to 216.239.37.17:5061 to ring my GrandCentral numbers.]]></description>
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<pubDate>Sun, 29 Mar 2009 14:47:55 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22147501</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  Liz <A HREF="/useremail/u/167641"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>The Grandcentral IP's still work but not much you can do with it after upgrade.<br> </div>Still work for what? Browsing or calling?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22147501</guid>
<pubDate>Sun, 29 Mar 2009 13:03:15 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22146903</link>
<description><![CDATA[<A HREF="/useremail/u/167641"><b>Liz</b></A> : <div class="bquote"><small>said by  PX Eliezer <A HREF="/useremail/u/1572525"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Go back one page in this thread.<br>You will read that it is dead.  :(<br> </div>But that was more than 8 hours ago.  With the current pace at Google Voice ... old news.<br><br>The Grandcentral IP's still work but not much you can do with it after upgrade.]]></description>
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<pubDate>Sun, 29 Mar 2009 10:14:23 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22146730</link>
<description><![CDATA[<A HREF="/useremail/u/1572525"><b>PX Eliezer</b></A> : Go back one page in this thread.<br>You will read that it is dead.  :(]]></description>
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<pubDate>Sun, 29 Mar 2009 09:03:01 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22146170</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  celtic <A HREF="/useremail/u/308189"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Are any of the SIP options currently working?<br> </div>I've been away all week, so I've had just a short time to work on this, but I'm not finding any SIP access any more. From my logs, I see they have changed some of the addresses, but none of the new addresses seem to work either. Too bad. Google giveth and Google taketh away.]]></description>
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<pubDate>Sun, 29 Mar 2009 01:25:29 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22143314</link>
<description><![CDATA[<A HREF="/useremail/u/308189"><b>celtic</b></A> : Are any of the SIP options currently working?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22143314</guid>
<pubDate>Sat, 28 Mar 2009 13:06:48 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22142066</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  mbuugg <A HREF="/useremail/u/1542856"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>To summerize, GV does not accept SIP address in phone settings. What Oroman did was, he entered a fake number 1234567890 in his GV phone list, checked it and set it to not ask for PIN. And in his ATA, he used a fake CID 1234567890 with the outbound proxy set to be the GV proxy. The whole thing has nothing to do with GV accepting SIP address in phone list. :D </div>You are right, and I apologize to Oroman for not reading his earlier post carefully enough. My ATA supports a second line, but requires that it use the same SIP proxy address and outbound proxy address for both lines. It is therefore difficult to create a "dummy" user id without losing use of the adapter for other providers. In fact I was trying to change the name on my pbxes user account so that I would have a numeric user id which GV would "recognize" (otherwise, it comes through as "unknown," which I believe I discussed earlier in this thread). I never got around to doing that.<br><br>But Oroman's trick is pretty interesting in another respect. He said he was able to dial his GV number by adding that GV IP address as an <b><i>outbound proxy server address</b></i>, and then just autodialing the GV number. In other words, he used a Gizmo SIP server, a Google IP address as an Outbound Proxy server, and then dialed his Google Voice number (was it the PSTN version, without the @, or was it just a 10 digit number?). That is very strange indeed, inasmuch as one shouldn't be able to use an outbound proxy server that way. For example, to make a SIP call to my Lingo phone, one would enter number@SipProxyServer, not number@OutboundProxyServer. Outbound proxy servers (border session controlers) help navigate the network; to my knowledge, they don't set up the call.<br><br>I'm not saying it can't be done; I just don't understand it. Oroman, since you say you autodialed your GV number, did it include the IP address as part of the number dialed, or was it just a 10 digit number? If the latter, is there any possibility that the call went out over PSTN via your Gizmo account?<br><br>Unfortunately, we may never get to test this because GV has shut down SIP access. It would be interesting to know just what that IP address was.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22142066</guid>
<pubDate>Sat, 28 Mar 2009 03:21:12 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22141032</link>
<description><![CDATA[<A HREF="/useremail/u/803435"><b>Test99</b></A> : <div class="bquote"><small>said by  PX Eliezer <A HREF="/useremail/u/1572525"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>Remember, Google was founded by Larry PAGE and thus their whole world revolves around web pages.</div>ROFL!  :)<br><br>Poxilla?  (OT, but hey - it's Friday!)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22141032</guid>
<pubDate>Fri, 27 Mar 2009 21:48:58 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22140720</link>
<description><![CDATA[<A HREF="/useremail/u/1572525"><b>PX Eliezer</b></A> : <div class="bquote"><small>said by  Test99 <A HREF="/useremail/u/803435"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Let's hope that's not the end of VOIP access to Google Voice.  Without full VOIP access, Google Voice seems irrelevant.<br> </div>Agreed.  But for the mindset of the Googlefins (Google staff boffins), nothing counts unless the display of a web page is somehow a part of the process.<br><br>Remember, Google was founded by Larry PAGE and thus their whole world revolves around web pages.<br><br>--------------------------------------<br><br>It seems ironic for Voxilla to be concerned about security, considering how their forum is prone to spam links about, shall we say, proper up-bringing.  Think of the TV commercial with the two bathtubs on the beach.  How do they get those bathtubs on the beach, anyway?<br><br>To add insult to injury, on one Voxilla forum, every time the guy posted another spam link about pills (the pills that make it easier to be harder), Voxilla was sending me an e-mail notification of a new posting.  Thus, I was receiving SPAM announcing SPAM.  True Monty Python.<br><br>A pox on Voxilla.]]></description>
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<pubDate>Fri, 27 Mar 2009 20:44:31 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22140620</link>
<description><![CDATA[<A HREF="/useremail/u/401196"><b>pandora</b></A> : I like GC and will like Google Voice once I move to it.<br><br>Eventually Google will provide phone access. It will probably be by purchase of a VOIP provider.<br><small>--<br>"People demand freedom of speech as a compensation for the freedom of thought which they seldom use."</small>]]></description>
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<pubDate>Fri, 27 Mar 2009 20:24:04 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22140612</link>
<description><![CDATA[<A HREF="/useremail/u/803435"><b>Test99</b></A> : Let's hope that's not the end of VOIP access to Google Voice.  Without full VOIP access, Google Voice seems irrelevant.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22140612</guid>
<pubDate>Fri, 27 Mar 2009 20:22:38 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22140591</link>
<description><![CDATA[<A HREF="/useremail/u/1572525"><b>PX Eliezer</b></A> : GoogleVoice deliberately killed off SIP access.<br><br>In fact, the folks at Voxilla, among others, apparently suggested that they do so.<br><br>See:<br><br>&raquo;<A HREF="http://voxilla.com/2009/03/24/despite-fixes-google-voice-security-flaws-remain-1627" >voxilla.com/2009/03/24/despite-f&middot;&middot;&middot;ain-1627</A><br><br>Furthermore:<br><br>"...Voxilla has contacted Google officials to make them aware of the remaining security vulnerability and to ask for comment. We will post updates when we hear back from them...."]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22140591</guid>
<pubDate>Fri, 27 Mar 2009 20:15:57 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22140494</link>
<description><![CDATA[<A HREF="/useremail/u/160346"><b>cbrain</b></A> : <div class="bquote"><small>said by  ArgMeMatey <A HREF="/useremail/u/448156"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Settings > Phones > Edit > Show Advanced Settings > Voicemail Settings > Yes (Require pressing star and entering PIN to check voicemails from this phone)<br> </div>Thanks.<br><br><div class="bquote"><small>said by  Pinan <A HREF="/useremail/u/195361"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>1. By Default, that is how it's set.<br><br>2. They would have to dial from the (your) phone in question.<br><br>Edit: I wonder if someone could indeed access your account by manipulating their own CID....? This would only work though if you changed the default setting to "No" (from Yes) for using a PIN.<br> </div> For cell phones, the default is no PIN.  I'm taking a good look at how to make best use of this service now.  I use Asterisk.  I set my extension to my cell number and made a test call.  It was answered without PIN request to give me access to messages and dial out.  Took less than a minute to make the change.  There is a legitimate use for this feature and most won't abuse it ... but it is way too easy.]]></description>
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<pubDate>Fri, 27 Mar 2009 19:52:41 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22140290</link>
<description><![CDATA[<A HREF="/useremail/u/195361"><b>Pinan</b></A> : <div class="bquote"><small>said by  cbrain <A HREF="/useremail/u/160346"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>1. Is there a way to set Google Voice to always require a PIN to access the account functions?<br><br>  2. The way it currently works, anyone who knows your numbers and can set outbound CID can access your account and messages.<br> </div>1. By Default, that is how it's set.<br><br>2. They would have to dial from the (your) phone in question.<br><br>Edit: I wonder if someone could indeed access your account by manipulating their own CID....? This would only work though if you changed the default setting to "No" (from Yes) for using a PIN.]]></description>
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<pubDate>Fri, 27 Mar 2009 19:12:23 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22140231</link>
<description><![CDATA[<A HREF="/useremail/u/1542856"><b>mbuugg</b></A> : To summerize, GV does not accept SIP address in phone settings. What Oroman did was, he entered a fake number 1234567890 in his GV phone list, checked it and set it to not ask for PIN. And in his ATA, he used a fake CID 1234567890 with the outbound proxy set to be the GV proxy. The whole thing has nothing to do with GV accepting SIP address in phone list. :D<br><br>GV closed the backdoor for the outsider to directly access its SIP proxy. That may be a consideration of Ads revenue or being abused, it is also a big security hole.]]></description>
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<pubDate>Fri, 27 Mar 2009 19:01:13 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22140064</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by Oroman :</small><br><br>Caseydoug, did you define your SIP phone on the GV phone list?  <br> </div>I am not able to define a SIP phone (other than Gizmo) in the GV phone list (Settings/Phones). I have been out of town all week, so I can't really test this properly now, but when I did, it wouldn't let me save my SIP address, which begins with letters. I WAS able to save a SIP address beginning with numbers (e.g., Callcentric or Gizmo), but as I recall, the Callcentric number ended up ringing on a PSTN number I didn't recognize in the 206 area code (I think I discussed this earlier in this thread). Others apparently got different results (cbrain, for example), so maybe I entered the number improperly. I'll try it again when I get home.<br><br>I am still trying to figure out what GV does with various types of phone entry in both Contacts and Phones. It looked to me like entries in the form "number@domain.com" were being truncated so that only the number was dialed. On the other hand, when I tried to dial a SIP number in the form "name3-201@domain.com," GV told me it was charging 2 cents per minute, and then it commenced to ring (no one answered). Again, this will take more work, assuming GV will permit any SIP dialing at all.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22140064</guid>
<pubDate>Fri, 27 Mar 2009 18:27:03 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22139757</link>
<description><![CDATA[<A HREF="/useremail/u/448156"><b>ArgMeMatey</b></A> : <div class="bquote"><small>said by  cbrain <A HREF="/useremail/u/160346"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Is there a way to set Google Voice to always require a PIN to access the account functions?  <br> </div>Settings > Phones > Edit > Show Advanced Settings > Voicemail Settings > Yes (Require pressing star and entering PIN to check voicemails from this phone)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22139757</guid>
<pubDate>Fri, 27 Mar 2009 17:36:56 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22139499</link>
<description><![CDATA[<A HREF="/useremail/u/160346"><b>cbrain</b></A> : Is there a way to set Google Voice to always require a PIN to access the account functions?  The way it currently works, anyone who knows your numbers and can set outbound CID can access your account and messages.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22139499</guid>
<pubDate>Fri, 27 Mar 2009 16:47:38 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22139335</link>
<description><![CDATA[<A HREF="/useremail/u/1140228"><b>soitgoes2</b></A> : <div class="bquote"><small>said by  AllThumbs <A HREF="/useremail/u/1324679"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Click-to-Dial is cute to show your nerdy friends, but no one is going to use it very long for real phone calls.  </div>Some will, but most won't (and that's good enough).]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22139335</guid>
<pubDate>Fri, 27 Mar 2009 16:17:59 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22139196</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : caseydoug , <br>I said &#145;Then apply my fix by adding a dummy contact with the same number as your PSTN phone and only route calls from that dummy contact to the disabled phone.&#146;<br>I should have said &#145;Then apply my fix by adding a dummy contact with the same number as your PSTN phone and route calls from that dummy contact to the disabled phone and no other phones in your list.&#146;  You can route other contact phone calls to the disabled phone.<br><br>I think GV has a problem with a contact having the same number as one in the phone list.  After I added the dummy contact and I was editing several contacts and GV would not allow me to put a check next to some contacts.  Logging off and back on fixes this problem.  This never happen before adding the dummy contact. ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22139196</guid>
<pubDate>Fri, 27 Mar 2009 15:58:03 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22138934</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Guys, I told you I was a novice at this stuff, but I&#146;m not a complete idiot.  I thought I was clear on what I did.  I even gave you the config of the second port on my ATA, but apparently I wasn&#146;t clear enough.  I believe the second port on my ATA is straight SIP.  I did log on to GV with it and got the voice prompt without a PIN.<br> <br>Mazilo, the second port on my ATA is not really a Gizmo account.  I am not providing any Gizmo account information.  I am using the Gizmo&#146;s proxy server, but according to caseydoug I could have used any other free server.<br>SIP is the name I assigned to the phone. It was defined as HOME.<br>I was not routing any GV calls to the SIP phone. I defined it in the GV phone list to avoid having to enter a PIN. I disabled (unchecked) it to prevent GV from trying to route calls to it. A disabled phone equals PIN problem (nothing to do with SIP or PSTN) which requires the PIN FIX.  <br><br>Caseydoug, did you define your SIP phone on the GV phone list? If it was defined did you disable it?  Did you try to duplicate the PIN problem on a PSTN phone by disabling it?  If not please try. Then apply my fix by adding a dummy contact with the same number as your PSTN phone and only route calls from that dummy contact to the disabled phone. ]]></description>
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<pubDate>Fri, 27 Mar 2009 15:13:09 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22138840</link>
<description><![CDATA[<A HREF="/useremail/u/1532944"><b>nitzan</b></A> : <div class="bquote"><small>said by  AllThumbs <A HREF="/useremail/u/1324679"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Click-to-Dial is cute to show your nerdy friends, but no one is going to use it very long for real phone calls.</div>And this is exactly why they either killed or will kill SIP access. They don't want to become your next phone company for free. ;)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22138840</guid>
<pubDate>Fri, 27 Mar 2009 14:54:13 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22138319</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>mazilo, I understand that his SIP account is Gizmo.</div>That's my first impression. However, unless the poster is confused Gizmo with SIP, s/he indicated his ATA is configured with Gizmo (on Line 1) and SIP (on Line 2) accounts.<br><br><div class="bquote">It is no trick to avoid entering the PIN when calling in from your Gizmo account provided you call the GV PSTN number.</div>Yes and you can use asterisk to do the trick.<br><br><div class="bquote">If he was able to avoid the PIN when calling the GV SIP address, however, that would be significant.</div>That's why I responded to his post to find out.<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22138319</guid>
<pubDate>Fri, 27 Mar 2009 13:17:44 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22137858</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : mazilo, I understand that his SIP account is Gizmo. It is no trick to avoid entering the PIN when calling in from your Gizmo account provided you call the GV PSTN number. If he was able to avoid the PIN when calling the GV SIP address, however, that would be significant.<br><br>Of course, all of this is academic if GV has shut down SIP access.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22137858</guid>
<pubDate>Fri, 27 Mar 2009 11:59:39 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22137665</link>
<description><![CDATA[<A HREF="/useremail/u/1324679"><b>AllThumbs</b></A> : Sad to see Google has closed off SIP access. :huh: At a time when Skype is opening up their service to SIP, you'd think Google of all companies could see the telephony writing on the wall. Otherwise, they'll end up in the same boat as AOL with their Internet telephony venture. Click-to-Dial is cute to show your nerdy friends, but no one is going to use it very long for real phone calls. And, their calling rates are nothing to write home about except for the U.S. calling teaser which no doubt will vanish in the next few months. Too bad!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22137665</guid>
<pubDate>Fri, 27 Mar 2009 11:33:45 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22136731</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by Oroman :</small><br><br>I have an ATA with 2 ports, one defined for Gizmo the other for SIP. The SIP port is defined as follows:<br>SIP Server = proxy01.sipphone.com<br>Outbound proxy=216.239.37.15:5061<br>SIP Userid=1234567890<br>Offhook Autodial=(My GV#)<br>When I took the SIP phone offhook, it would connect to GV without PIN at no cost and I was able to dial out.</div>Your explanation above showed you have an ATA configured with Gizmo and SIP accounts. The SIP account you showed above is also a Gizmo5 account. As such, your ATA is configured with two Gizmo5 accounts. Can you confirm this?<br><br>Also, GV only allows forwarding to Cell, Home, Work, Gizmo, and no SIP numbers, AFAIK. If so, can you show us how did you configure your GV account to forward incoming calls to a SIP account?<br><br><div class="bquote">This stopped working this evening.</div>This is a known issue.<br><br>BTW, your configuration shown above indicated your ATA is behind a symmetrical NAT/Firewall router, isn't it? Also, what is the name/model of your ATA?<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22136731</guid>
<pubDate>Fri, 27 Mar 2009 07:48:01 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22136357</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Caseydoug, I made a mistake in my previous post. I have 3 phones defined to GV (PSTN, Gizmo and SIP). I have an ATA with 2 ports, one defined for Gizmo the other for SIP. The SIP port is defined as follows:<br>SIP Server = proxy01.sipphone.com<br>Outbound proxy=216.239.37.15:5061<br>SIP Userid=1234567890<br>Offhook Autodial=(My GV#)<br>When I took the SIP phone offhook, it would connect to GV without PIN at no cost and I was able to dial out.<br>This stopped working this evening.]]></description>
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<pubDate>Fri, 27 Mar 2009 01:45:03 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22136210</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : cbrain, can you save a sip number with alpha characters, not necessarily dial out, but just save it?<br><br>Edit: I was wrong. I AM able to enter alpha characters. I'm not at home, so I can't be sure where the call is going. GV says the cost of the call is 2 cents per minute, which suggests that it is converting the alpha characters to numbers, and trying to make an international call.<br><br>oroman, if your sip number is your Gizmo number, then I think I understand what is happening. The real test is whether you can call in with a free sip account, with no cost for the call and no monthly charge. Is that what you are doing?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22136210</guid>
<pubDate>Fri, 27 Mar 2009 00:30:09 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22136165</link>
<description><![CDATA[<A HREF="/useremail/u/160346"><b>cbrain</b></A> : For the moment, I can save a sip number for a contact and can click that number to successfully dial out.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22136165</guid>
<pubDate>Fri, 27 Mar 2009 00:16:07 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22136067</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Mazilo asked:<br>'What exactly did you mean to have the SIP phone enabled on GV phone list? I don't see this option. Can you please elaborate?'<br><br>I'm in test mode on GV and only have two phones (PSTN and Gizmo) defined.  I'm using GV to receive inbound calls on my Gizmo phone and make long distance calls on my PSTN phone. It is my understanding if a call comes in that is not defined as a contact it will ring all the phones that are enabled on the phone list. Enabled,I mean a check mark next to the phone# in the phone list. I did not want to receive calls from GV on my PSTN phone, but when I unchecked it GV required a PIN.  I found the above solution on the GV forum.<br>Maybe you didn't define the SIP phone on GV.  I'm using an ATA and I used 1234567890 as the SIP UserID.  When the call came in to GV it had phone number 123-456-7890.  I then defined 123-456-7890 in my GV phone list.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22136067</guid>
<pubDate>Fri, 27 Mar 2009 00:04:23 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22136109</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  cbrain <A HREF="/useremail/u/160346"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I entered "Test Contact" into my Goog contact list with a SIP number to my Asterisk box. It saved the number and click to call has worked for the past hour.<br> </div>Now I'm confused. You are saying you can make GV calls <b><i>out</b></i> via SIP? I was not able to do that. SIP numbers with alpha characters before the "@" are not accepted as contact numbers. SIP numbers with all numeric characters before the "@" are accepted, but are essentially truncated, so they do not work. I can make inbound SIP calls to GV, but not outbound SIP calls from GV.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22136109</guid>
<pubDate>Fri, 27 Mar 2009 00:01:36 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22136053</link>
<description><![CDATA[<A HREF="/useremail/u/160346"><b>cbrain</b></A> : <div class="bquote"><small>said by  nitzan <A HREF="/useremail/u/1532944"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by ifed  :</small><br><br>It seems that they disabled SIP access to GV.</div>Wow, that was quick! so basically all the workarounds are now null and void?  </div> Don't know, but, I think I'll wait and see if they made a decision to not allow SIP or this is simply an effort to get control of their network during this rapid shift. I entered "Test Contact" into my Goog contact list with a SIP number to my Asterisk box. It saved the number and click to call has worked for the past hour.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22136053</guid>
<pubDate>Thu, 26 Mar 2009 23:48:51 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22135834</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by Oroman :</small><br><br>I&#146;m assuming you didn&#146;t have the SIP phone enabled on your GV phone list.</div>What exactly did you mean to have the <i>SIP phone enabled on GV phone list</i>? I don't see this option. Can you please elaborate?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22135834</guid>
<pubDate>Thu, 26 Mar 2009 22:59:09 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22135669</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : <div class="bquote"><small>said by  nitzan <A HREF="/useremail/u/1532944"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by ifed  :</small><br><br>It seems that they disabled SIP access to GV.</div>Wow, that was quick! so basically all the workarounds are now null and void?<br> </div>To a degree this might be good. To stop even what at this point to them is a slight bleed of money may be a sign they are interested in really keeping GV. Plus I bet the loophole with a callcenter type thing???? <br>Shows they are watching. ]]></description>
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<pubDate>Thu, 26 Mar 2009 22:24:04 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22135644</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Caseydoug asked:<br>&#147;Are you saying there is a way to avoid the PIN when calling in via SIP? I know you can do it using a PSTN number, but I haven't found a way to call GV-SIP from some other SIP number without entering your PIN.&#148;<br><br>Yes, it worked on SIP without a PIN.  I&#146;m assuming you didn&#146;t have the SIP phone enabled on your GV phone list.  The same thing will happen on a PSTN number.  Try it.   Uncheck one of your PSTN phones and call GV with it.  A PIN will be required.  Adding a contact will the same phone number with calls routed to the unchecked PSTN phone will correct the problem.<br>This is all irrelevant now that they cut us off unless you know another way to do it via SIP.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22135644</guid>
<pubDate>Thu, 26 Mar 2009 22:20:14 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22134967</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : I suspect they read these boards (and others).]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22134967</guid>
<pubDate>Thu, 26 Mar 2009 20:06:50 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22134956</link>
<description><![CDATA[<A HREF="/useremail/u/1532944"><b>nitzan</b></A> : <div class="bquote"><small>said by ifed :</small><br><br>It seems that they disabled SIP access to GV.</div>Wow, that was quick! so basically all the workarounds are now null and void?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22134956</guid>
<pubDate>Thu, 26 Mar 2009 20:04:08 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22134806</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : It seems that they disabled SIP access to GV. Calling to .13, .14, .15 addresses produces ICMP reply. While .16, .17 address es are still alive but route calls to GC not GV account.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22134806</guid>
<pubDate>Thu, 26 Mar 2009 19:32:46 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22134163</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by Oroman :</small><br><br>If you don&#146;t have a phone enable (checked) GV will require a PIN.  To get around this bug add a contact with the same phone number and have it ring itself.  I can&#146;t take credit for this solution, I found it on the GV forum.<br> </div>Are you saying there is a way to avoid the PIN when calling in via SIP? I know you can do it using a PSTN number, but I haven't found a way to call GV-SIP from some other SIP number without entering your PIN.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22134163</guid>
<pubDate>Thu, 26 Mar 2009 17:38:41 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22133758</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Alternatively, after they get a large user base, they could begin charging a few dollars a month (or buy Vonage).]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22133758</guid>
<pubDate>Thu, 26 Mar 2009 16:32:04 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22132565</link>
<description><![CDATA[<A HREF="/useremail/u/1140228"><b>soitgoes2</b></A> : <div class="bquote"><small>said by  B <A HREF="/useremail/u/229804"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>if it gets widespread and used by home PBXes I've got to think they'll curb it somehow?<br> </div>yes. Not only would they not have you on their website to view ads, but they would see high expenses for call termination.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22132565</guid>
<pubDate>Thu, 26 Mar 2009 13:10:47 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22132396</link>
<description><![CDATA[<A HREF="/useremail/u/1591150"><b>otty</b></A> : Oh right I forgot there is a good reason they don't officially support SIP: you would then not visit the website where they can advertise to you. ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22132396</guid>
<pubDate>Thu, 26 Mar 2009 12:43:45 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22132303</link>
<description><![CDATA[<A HREF="/useremail/u/229804"><b>B</b></A> : I assume it would have something to do with paying for unlimited nationwide calling to the PSTN. ;)<br><br>I'm quite serious really.  If there's something I'm missing please let me know.  For now the ability to do PSTN outbound calling is kinda sorta hidden; if it gets widespread and used by home PBXes I've got to think they'll curb it somehow?<br><br>-- B<br><small>--<br>In a realm outside causality and function</small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22132303</guid>
<pubDate>Thu, 26 Mar 2009 12:27:20 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22132280</link>
<description><![CDATA[<A HREF="/useremail/u/1591150"><b>otty</b></A> : Another question is WHY would they break them? <br><br>I can see no reason why they don't just allow SIP connections as an upfront policy.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22132280</guid>
<pubDate>Thu, 26 Mar 2009 12:24:04 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22132272</link>
<description><![CDATA[<A HREF="/useremail/u/229804"><b>B</b></A> : Now that NerdVittles is making it (relatively) easy, the only questions I have are:<br><br>1.  How long will it take for Google to break these techniques, and<br><br>2.  How will they break them?<br><br>-- B<br><small>--<br>In a realm outside causality and function</small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22132272</guid>
<pubDate>Thu, 26 Mar 2009 12:22:38 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22132199</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Thank you caseydoug.  I&#146;m a novice at this stuff and it took me several tries, but it&#146;s working perfectly now.<br><br>If you don&#146;t have a phone enable (checked) GV will require a PIN.  To get around this bug add a contact with the same phone number and have it ring itself.  I can&#146;t take credit for this solution, I found it on the GV forum.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22132199</guid>
<pubDate>Thu, 26 Mar 2009 12:10:37 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22120233</link>
<description><![CDATA[<A HREF="/useremail/u/1442320"><b>joebobcooter</b></A> : Confirmed that this works with les.net.<br><br>Very strong.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22120233</guid>
<pubDate>Tue, 24 Mar 2009 12:47:09 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22118846</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : You wouldn't call the ipkall number. You call the sip url of the gv number. That forwards it to the ipkall number. You have set up the ipkall number so that it forwards to your overseas sip url.<br><br>So when you call the sip url of the gv number, it will ring you back via the forwarding to the ipkall number, which will forward to the sip url of your overseas number.<br><br>I have only done this from Canada to the US so can't swear it works the same way overseas, but don't see why it would not.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22118846</guid>
<pubDate>Tue, 24 Mar 2009 07:52:40 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22118778</link>
<description><![CDATA[<A HREF="/useremail/u/1324679"><b>AllThumbs</b></A> : Just tried mine. Still works fine. Nerd Vittles has updated the article and added a way to configure Google Voice as an Asterisk trunk. You might try that. Now you can just dial GV-202-456-1111 from any phone to call the White House. :D]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22118778</guid>
<pubDate>Tue, 24 Mar 2009 07:19:41 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22118697</link>
<description><![CDATA[<A HREF="/useremail/u/671633"><b>zenjabba</b></A> : it seems the nerdvettles method isn't working, all it does now is ring my asterisk box straight back.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22118697</guid>
<pubDate>Tue, 24 Mar 2009 06:20:22 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22117161</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by Bezel :</small><br><br>I didn't realize it earlier, but a gc/gv number is a godsend to people overseas who want free calling within the US. They can dial the sip url (or do the callback via ipkall => sip url), and then call anywhere in the US for free.  </div>I may be missing something, but wouldn't it be an international call to call the IPKall number from overseas? The IPKall number is within the PSTN. I agree that this would work if someone made a SIP call to his or her GV number and used GV as the gateway to the PSTN.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22117161</guid>
<pubDate>Mon, 23 Mar 2009 20:42:50 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22115826</link>
<description><![CDATA[<A HREF="/useremail/u/1003137"><b>garys_2k</b></A> : <div class="bquote"><small>said by  cbrain <A HREF="/useremail/u/160346"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br><div class="bquote"><small>said by  cbrain <A HREF="/useremail/u/160346"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</small><br><br>...<br>Enter "sip:destination#@host.domain.com" into the number space and select "Gizmo" from the drop down.  It will not accept your Google Voice number as a destination.  It rings Asterisk instantly.<br> </div>SIP uri is not working this morning.  Google removes everything but the number after you close the page.  <br> </div>That's what I get, too. The "sip:" and the "@" (and everything past that) are removed. Must be Gizmo only, again.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22115826</guid>
<pubDate>Mon, 23 Mar 2009 16:47:58 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22113842</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I didn't realize it earlier, but a gc/gv number is a godsend to people overseas who want free calling within the US. They can dial the sip url (or do the callback via ipkall => sip url), and then call anywhere in the US for free. <br><br>I suppose google and/or the overseas telecoms will plug that loophole eventually.<br><br>I can see gc/gv numbers on ebay already. $25? $50?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22113842</guid>
<pubDate>Mon, 23 Mar 2009 10:43:11 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22113829</link>
<description><![CDATA[<A HREF="/useremail/u/1324679"><b>AllThumbs</b></A> : Nerd Vittles has a new tutorial on free U.S. SIP calling and free message transcription for Asterisk using Google Voice...<br><br><b>Googlified Messaging: Asterisk's New Best Friend</b><br><br>&raquo;<A HREF="http://nerdvittles.com/?p=593" >nerdvittles.com/?p=593</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22113829</guid>
<pubDate>Mon, 23 Mar 2009 10:40:31 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22113683</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by Bezel :</small><br><br>Using gc by calling a sip url is pretty handy, but it is finicky. My gv number terminates at a phone here (the number is local to relatives, not me), and I couldn't get the * / pin / 2 combo to work unless I let it ring at least once. </div>Yes, I noticed that the timing of the key press makes a difference. If I wait too long, the * takes me right to voice mail rather than to the "Enter your PIN" prompt.<br><br>I never tested to see whether this occurs only when I dial a SIP address from an adapter or soft phone, or whether this is also true when dialing from an ordinary phone.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22113683</guid>
<pubDate>Mon, 23 Mar 2009 10:08:48 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22113575</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : This may be obvious to most here, but I am sure there are a few people as unknowledgeable as I am.<br><br>Using gc by calling a sip url is pretty handy, but it is finicky. My gv number terminates at a phone here (the number is local to relatives, not me), and I couldn't get the * / pin / 2 combo to work unless I let it ring at least once.<br><br>I finally solved the problem by adding to my dial plan:<br><br><textarea name="code" class="text" cols=50 rows=10>&lt;9999:1xxxxxxxxxx@216.239.37.15:5061&gt;S0|&#012;</textarea><!--end code block--><br>This lets me call even from the phone where the gv number is terminated.<br>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22113575</guid>
<pubDate>Mon, 23 Mar 2009 09:42:16 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22113351</link>
<description><![CDATA[<A HREF="/useremail/u/160346"><b>cbrain</b></A> : <div class="bquote"><small>said by  cbrain <A HREF="/useremail/u/160346"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>...<br>Enter "sip:destination#@host.domain.com" into the number space and select "Gizmo" from the drop down.  It will not accept your Google Voice number as a destination.  It rings Asterisk instantly.<br> </div>SIP uri is not working this morning.  Google removes everything but the number after you close the page.  ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22113351</guid>
<pubDate>Mon, 23 Mar 2009 08:50:39 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22112394</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : My point is that the GV Settings/Phones tab would not save several SIP addresses (ones with letters before the "@"), but it did accept my CallCentric address. When I tried to call that extension through GV, however, it did not reach the phone I was trying to call. I was sent to some GC member's voicemail, but the ring was sent to a woman in the 206 area code.<br><br>Apropos of "The Starlost," this is like the guy who shows up at the airline ticket counter and says, "I'm going to Denver. I'd like this bag to go to Detroit, and that bag to go to New Orleans." <br><br>"We can't do that," the agent says.<br><br>"Why not? You did it the last time I flew."]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22112394</guid>
<pubDate>Sun, 22 Mar 2009 22:57:10 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22112189</link>
<description><![CDATA[<A HREF="/useremail/u/1572525"><b>PX Eliezer</b></A> : "777" is what CallCentric uses, as you said.  It is not a real POTS area code.  Nevada tried to get 777 (think about it) but was denied.<br><br>-------------------------------------<br><br>However, an upcoming problem is "747" which is the Gizmo5/SIPphone code.  Later this year 747 will also become a real POTS area code in California.<br><br>Mark my words, this will cause problems.  But Gizmo5 is like the spaceship in the awful 1973 SF TV series "The Starlost"---the bridge crew has vanished and the ship is on autopilot.<br><br>BTW, "The Starlost" was a fine work by Harlan Ellison.  The original screenplay was given a Writers Guild Award.  But the actual TV series, produced in 1973-74, was so altered by the producers (and was SO bad) that Ellison had his name removed from the credits and replaced with the pseudonym "Cordwainer Bird". ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22112189</guid>
<pubDate>Sun, 22 Mar 2009 22:02:16 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22112127</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : OK, I got it to take the number. It apparently needs a 10 or 11 digit number before the @. <br><br>I then tried to enter my Callcentric number, which is 11 digits followed by @callcentric.com. The strangest thing happened when I then called my GV number with that callcentric number as the only active extension. It didn't ring the phone connected to callcentric. It did, however, lead me to a GRAND CENTRAL voicemail announcement (not GV). Finally, a few moments after I hung up I received a call from a woman who said I had been trying to call her. I didn't think to ask her what her number was, but Callcentral uses a 777 "area code," which I think does not exist.<br><br>I was a little embarrassed, and I don't want to experiment with using that number again. :)<br><br>Edit: I just checked my GC voicemail, and the message I left was not on it, so it went to someone else's voicemail. I also checked the CID of the woman who called me, and the number was nothing like the one my GV number was forwarded to. Go figure.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22112127</guid>
<pubDate>Sun, 22 Mar 2009 21:47:01 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22111969</link>
<description><![CDATA[<A HREF="/useremail/u/160346"><b>cbrain</b></A> : What are you entering?  Google took both numbers or name from me.  Anything but my Goog number.<br><br>Try sip:2029876543@call.dslreports.com - but substitute your number and domain.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22111969</guid>
<pubDate>Sun, 22 Mar 2009 21:10:58 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22111935</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Maybe I'm not understanding you, cbrain. When I try to add a SIP address in the "Phones" tab of Settings, it won't take it. It says, "International numbers are not supported as forwarding phones." Interestingly, when I tried to add, literally, "sip:destination#@host.domain.com," it gave me a different error message ("Invalid phone number"). <br><br>I wonder whether the format of your specific SIP address was somehow more acceptable. I have noticed that, for example, if I try to call a Gizmo or Callcentric number (both of which are numeric prior to the @, it will treat it as an international number, dropping everything after the @.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22111935</guid>
<pubDate>Sun, 22 Mar 2009 21:04:59 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22111737</link>
<description><![CDATA[<A HREF="/useremail/u/160346"><b>cbrain</b></A> : I entered my SIP URI as 1 of my forwarded numbers to my Google Voice settings page.  It accepted and saved the number, but did not work.  I then tried every name option and found that if I selected "Gizmo" from the drop down, it worked.<br><br>Enter "sip:destination#@host.domain.com" into the number space and select "Gizmo" from the drop down.  It will not accept your Google Voice number as a destination.  It rings Asterisk instantly.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22111737</guid>
<pubDate>Sun, 22 Mar 2009 20:18:06 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22111669</link>
<description><![CDATA[<A HREF="/useremail/u/1294553"><b>jwill370</b></A> : My ATA (old Sunrocket Gizmo) registers with mysipswitch.com. I added an extension to mysipswitch dialplan:<br><br>exten = 100,1,Switch(myGVnumber@216.239.37.13:5061);<br><br>This gets me to my GV number but as an unknown call. I can enter * and then my pin to access vm and place calls.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22111669</guid>
<pubDate>Sun, 22 Mar 2009 19:59:46 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22111456</link>
<description><![CDATA[<A HREF="/useremail/u/1075020"><b>burgerwars</b></A> : GoogleVoiceNumber@216.239.37.15:5061 also works with IPKALL and VOIP.MS.  I Just tried.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22111456</guid>
<pubDate>Sun, 22 Mar 2009 18:57:24 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22111235</link>
<description><![CDATA[<A HREF="/useremail/u/160346"><b>cbrain</b></A> : Great find!  I just set up Asterisk and it works with the 3 Google Voice numbers I tested.  If I send the same outbound CID as the called number, it rings busy(could be an internal * thing).  Any other CID works and rings all lines as set up on your Google Voice page.  I can complete the call over a provider line using the same number and CID.  I only tested 216.239.37.15 but if I figure a useful reason to use this I will look at the other options.<br><br>Thanks.  Let us know what else you discover. ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22111235</guid>
<pubDate>Sun, 22 Mar 2009 18:01:37 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108690</link>
<description><![CDATA[<A HREF="/useremail/u/531841"><b>TheMole</b></A> : Works for me.  I have a google voice number now.<br><br>I'm running asterisk, so i'm browsing my logs and am seeing some funny results here. Seems the IP address might control which of your multiple GV forward numbers ring.<br><br>IP ending in .15 rings forward numbers that are "checked" in GV. Local voice mail picks up and is connected.<br><br>IP ending in .17 rings a number that is NOT checked in GV (i know becuase the number it is ringing is a DID on my asterisk box and see it in the logs).<br><br>IP ending in .18 rings the same non checked number that .17 did above.<br><br>IP ending in .19 rings the same non checked number that .17 did above.<br><br>IP ending in .20 does the same as .17 and .19 for me.<br><br>IP ending in .16 rings someplace (not sure if it one of my non checked numbers), and when nobody picks up my GV voice mail picks up.<br><br>IP ending in .11, .12, .13, and .14 rings the same as .15 above.<br><br>I cannot determine how to ring a certain non checked number.  Only one of my non checked numbers rings (ip ending in .17 - .20)<br><br>IP ending in .10 is dead. <br><br>The CID number coming over to my asterisk box is 4065309999<br><br>edit: clarity, it is late.  sorry.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108690</guid>
<pubDate>Sat, 21 Mar 2009 23:54:13 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108395</link>
<description><![CDATA[<A HREF="/useremail/u/1461319"><b>emoci</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>   :</small><br><br><div class="bquote"><small>said by  emoci <A HREF="/useremail/u/1461319"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>    :</small><br><br>No I mean GC(not GV number)@IPAdress rings all my phones... </div>I think I understand. But aren't your GC and GV numbers the same?<br> </div>I have one that I upgraded to GV and one that is still GC...<br><br>Nonetheless it seems that having the right IP range makes a difference...(switching the xx from 15 to 19 fixes the VM presentment issue for me)<br><br>Good find indeed...<br><br>Anyone still wondering how this is beneficial: You can now have any third party DID from F9, CallCentric, or even international forward directly to GV via SIP URI (hence no cost)...from there GV will ring all your phones as usual]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108395</guid>
<pubDate>Sat, 21 Mar 2009 22:24:05 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108384</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : <div class="bquote"><small>said by  emoci <A HREF="/useremail/u/1461319"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>No I mean GC(not GV number)@IPAdress rings all my phones... </div>I think I understand. But aren't your GC and GV numbers the same?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108384</guid>
<pubDate>Sat, 21 Mar 2009 22:20:16 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108366</link>
<description><![CDATA[<A HREF="/useremail/u/1461319"><b>emoci</b></A> : No I mean GC(not GV number)@IPAdress rings all my phones...but when it comes time for VM to take the call, GC VM doesn't answer...cell's VM (where GC is forwarding to) answers instead...<br><br>GV doesn't seem to have the same issue...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108366</guid>
<pubDate>Sat, 21 Mar 2009 22:16:41 EDT</pubDate>
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<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108361</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : You mean if your cell phone is one of the GV "ring to" numbers, and you call CellNumber@216.238.37.xx, it reaches your cell phone? Funny, I tried that in the very beginning and it didn't work.<br><br>As for reaching voicemail quickly, it may be reading as busy.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108361</guid>
<pubDate>Sat, 21 Mar 2009 22:14:46 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108315</link>
<description><![CDATA[<A HREF="/useremail/u/1461319"><b>emoci</b></A> : One odd thing...this actually works with GC numbers as well...but when I call this way a GC (non-upgraded number) I end up in the VM of the phone to which that number is forwarding within a ring or two... (no press 1...but I don't know how it is triggering VM so early on the receiving phone either...)<br><br>Update:<br><br>SIP Calls to GV numbers: Ring all forward-to phones, receive GV voicemail if no answer<br><br>SIP Calls to GC numbers: Ring all forward-to phones, do not receive GC voicemail...the VM from one of the forward-to phones takes the call...<br><br>The SIP URI does not seem to get updated instantly if your Forward-To number changes...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108315</guid>
<pubDate>Sat, 21 Mar 2009 22:02:01 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108296</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Thanks, emoci. That's not quite what I had in mind, however. I am using an old Azacall adapter to receive GV forwarded calls and also to call out using a GV SIP connection. I previously had been using this adapter as a pbxes extension. I can do both simultaneously -- making a SIP call to GV does not use any of my pbxes trunks -- but GV does not recognize the number it sees when I use this adapter to call in if the adapter is configured for pbxes. Switching the adapter from pbxes to Gizmo or Callcentric solves this problem because both services use numeric user names. But if I do that, I'm not using pbxes.<br><br>It doesn't really matter too much, however, since there is not much benefit to GV recognizing the number. I still have to press "* PIN" to get to my voicemail and other services.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108296</guid>
<pubDate>Sat, 21 Mar 2009 21:56:01 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108264</link>
<description><![CDATA[<A HREF="/useremail/u/1461319"><b>emoci</b></A> : So far 216.239.37.xx:5061 where xx is 15, 17, 19, 20 all work...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108264</guid>
<pubDate>Sat, 21 Mar 2009 21:48:06 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108206</link>
<description><![CDATA[<A HREF="/useremail/u/1461319"><b>emoci</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>Are either of you pbxes users? I would like to be able to let this adapter register to pbxes again, while at the same time using it to call GV. But as a pbxes "extension," the adapter needs to have a name based on my pbxes user name. All pbxes extensions have a dash ("-") in them. Use of alpha characters in the phone number seems to prevent GV from recognizing the phone as a contact. If you happen to have an all-numeric pbxes user name, I'd be interested to know whether the dash alone causes this problem in GV. In other words, if you try to SIP-call your GV number from a pbses extension, how does GV treat the call?<br><br>Alternatively, I could terminate and recreate my pbxes account. However, I've heard horror stories about accounts getting blocked when someone tries to do that.<br> </div>If you're assigning a user-number@pbxes.org to the above SIP URI and trying to call that form outside...it is one of those known issues with PBXes...not just in this scenario..<br>(solutiong is to create an inbound route with the user-number as the Trunk name...)<br><br>If you have trouble calling this SIP URI from PBXes that's a different story...I'll have to test<br><br>So far I simply added the SIP URI as a VoXalot Speed Dial...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108206</guid>
<pubDate>Sat, 21 Mar 2009 21:34:10 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108180</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Are either of you pbxes users? I would like to be able to let this adapter register to pbxes again, while at the same time using it to call GV. But as a pbxes "extension," the adapter needs to have a name based on my pbxes user name. All pbxes extensions have a dash ("-") in them. Use of alpha characters in the phone number seems to prevent GV from recognizing the phone as a contact. If you happen to have an all-numeric pbxes user name, I'd be interested to know whether the dash alone causes this problem in GV. In other words, if you try to SIP-call your GV number from a pbses extension, how does GV treat the call?<br><br>Alternatively, I could terminate and recreate my pbxes account. However, I've heard horror stories about accounts getting blocked when someone tries to do that.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108180</guid>
<pubDate>Sat, 21 Mar 2009 21:25:35 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108111</link>
<description><![CDATA[<A HREF="/useremail/u/1461319"><b>emoci</b></A> : I can also confirm it works...<br><br>What bothers me is that I've seen that IP when receiving calls from GC to MagicJack since last February but always assumed it was MJs servers... :uhh:]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108111</guid>
<pubDate>Sat, 21 Mar 2009 21:04:06 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22108018</link>
<description><![CDATA[<A HREF="/useremail/u/803435"><b>Test99</b></A> : That IP address works for me.  Just left a message in my Google Voice mailbox and received it by email.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22108018</guid>
<pubDate>Sat, 21 Mar 2009 20:41:35 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22107641</link>
<description><![CDATA[<A HREF="/useremail/u/1512792"><b>Livadia</b></A> : So that you would not have to look: The above address is<br>registered to Google at Mountainview]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22107641</guid>
<pubDate>Sat, 21 Mar 2009 19:02:16 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22107591</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : Ok, here's a little more information. I found the IP address by forwarding GV to my Gizmo number. When I called my GV number from my cell phone, the logs showed the call as MyCellNumber@IPAddress. I noticed that a couple of the via statements also used MyGizmoNumber@IPAddress (this is from memory, so details may be slightly off). Putting 2 and 2 together, I tried "dialing" MyGVNumber@IPAddress using a speed dial setting on my adapter, and it worked to get me in to my GV number.<br><br>For those who don't want to look this up themselves, IPAddress was 216.239.37.15:5061. Yours may be use a different server, however.<br><br>To this point, I have not been able to get in using a direct URI connection from my adapter or a soft phone. The logs show a connection, but then an immediate "BYE" from GV. So I'm guessing GV will accept connections only from SIP proxy servers. However, I have made this work using several free SIP proxy servers, including pbxes, Callcentric, and Gizmo. Pbxes requires that telephone extensions be named with a format that includes alpha characters, so GV does not recognize the telephone number and treats it as "unknown." Both Gizmo and Callcentric use numeric usernames, however, and these are recognized as 10-digit numeric telephone numbers which you can name and put in your contact list. When a call comes in from one of these numbers, and if it is in your contact list, the caller is announced just like any other caller. Interestingly, the Gizmo number looks identical to the number of the Gizmo "ring-to" phone, but is treated as a different phone. I don't believe you can use the "Call" button to call one of these SIP numbers.<br><br>It was quite easy to forward my Gizmo number to my adapter, and to set my adapter to register with Gizmo. I set GV to ring the phone connected to that adapter, and turned off call presentation and screening for the "contact" connected to the SIP number for that adapter. Using speed dial, I can call into my GV number, press "*2 PIN" while it is ringing, and then dial out to US numbers for free. The speed dial on my adapter is not flexible enough to include pauses and other characters, but I'm certain that this could be done using a more powerful adapter. And of course a dial plan using asterix could make the dialing process identical to using a normal telephone for outbound calls. <br><br>If Google allows these SIP connections to continue, I would seriously think about giving up other telephone services. Between its features and its free US calling, GV appears to be a pretty complete substitute.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22107591</guid>
<pubDate>Sat, 21 Mar 2009 18:49:26 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22106827</link>
<description><![CDATA[<A HREF="/useremail/u/1518597"><b>dalrun</b></A> : "What is the IP? Are you sure it is a Google IP"<br><br>I wonder if he's calling the Google IP address that sets up the call (I tried it w/ CC and got 'invalid #'). I haven't paid close attention, but it seems that all my incoming calls involve a single port 5060 packet from the provider and that the source IP is a constant.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22106827</guid>
<pubDate>Sat, 21 Mar 2009 15:23:53 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22105931</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : mazilo, aren't you the Sultan of Cheap?  :) I thought you would appreciate this. I'm right now working on a dial plan from my old Ling adapter.<br><br>celtic, I'd prefer not to publish the IP right now. It may not be the same for everybody, and in any event, I'd like to keep the usage low until I've checked out a couple of things. Suffice it to say it wasn't hard to find. I looked it up on whois, and it's definitely a Google IP address. It's not going through Gizmo because my adapter does not connect to Gizmo (although Gizmo forwards to my adapter).<br><br>Right now I'm working on spoofing my phone number so that I don't have to go through call presentation. If this is possible, there will constitute a potential big security hole.<br><br>Edit: I may be wrong about this being a direct connection. This adapter normally registers to pbxes in order to get dial tone. When I disabled the pbxes connection, I was not able to connect to the GV number. At the moment, I haven't determined whether reaching the GV number requires going through a SIP proxy. Even if it does, however, this is definitely a SIP to SIP (i.e., free) call.<br><br>In addition, I did set up the adapter to register to Gizmo. Interestingly, GV recognized the caller ID as my Gizmo number, but did not recognize it as the Gizmo phone that was already set up in GV. In other words, it is the same number, but it is treated as a GV contact, not as "me." When I turned off call presentation and screening for this "contact," the call went right through, and pressing "*" led me to my voicemail. Obviously, more tinkering is required. ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22105931</guid>
<pubDate>Sat, 21 Mar 2009 11:27:56 EDT</pubDate>
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<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22105529</link>
<description><![CDATA[<A HREF="/useremail/u/308189"><b>celtic</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>...I just checked the logs of my VoIP adapter and got the IP address, then used the speed dial feature of my adapter to dial the URI. This is a direct connection, without going through Gizmo or any other SIP provider. From there I was able to get to my voicemail, press 2 to dial out, etc. ...<br> </div>What is the IP?  Are you sure it is a Google IP or could it be Gizmo?  I didn't think either Google Voice or GrandCentral directly communicate with your computer or adapter.  Could you be going through Gizmo?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22105529</guid>
<pubDate>Sat, 21 Mar 2009 09:27:17 EDT</pubDate>
</item>

<item>
<title>Re: Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22105298</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><small>said by  caseydoug <A HREF="/useremail/u/451342"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</small><br><br>I'm not sure whether this has already been discussed, but it <b><i>is</i></b> possible to reach to your Google Voice number via a SIP connection. I just checked the logs of my VoIP adapter and got the IP address, then used the speed dial feature of my adapter to dial the URI.</div>Interesting found. I believe you can configure asterisk with a dialplan context to dial out. :D<br><small>--<br>Mazilo always prays for FREEBIES!<br>US Phone: +1-678-601-0907<br>UK Phone: +44-703-194-2574<br></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22105298</guid>
<pubDate>Sat, 21 Mar 2009 07:37:11 EDT</pubDate>
</item>

<item>
<title>Connecting to Google Voice Via SIP</title>
<link>http://www.dslreports.com/forum/remark,22105161</link>
<description><![CDATA[<A HREF="/useremail/u/451342"><b>caseydoug</b></A> : I'm not sure whether this has already been discussed, but it <b><i>is</i></b> possible to reach to your Google Voice number via a SIP connection. I just checked the logs of my VoIP adapter and got the IP address, then used the speed dial feature of my adapter to dial the URI. This is a direct connection, without going through Gizmo or any other SIP provider. From there I was able to get to my voicemail, press 2 to dial out, etc. There were a few hiccups because GV didn't recognize the "phone number" (so, for example, I had to enter my PIN to get to voicemail), but I suspect even this can be corrected. <br><br>What this means, of course, is that you can basically get totally free phone service using a telephone rather than a computer. I don't know whether the IP addresses will change or Google will otherwise block this method of making outbound calls. And for inbound calls, I pointed Gizmo directly to my adapter. I haven't found any way to forward inbound calls to my adapter without going through Gizmo.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,22105161</guid>
<pubDate>Sat, 21 Mar 2009 05:13:53 EDT</pubDate>
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