<?xml version="1.0" encoding="UTF-8"?>

<rss version="2.0" xmlns:blogChannel="http://backend.userland.com/blogChannelModule">

<channel>
<title>VOIP Tech Chat forum - dslreports.com community</title>
<link>http://www.dslreports.com/forum/voip</link>
<description>VOIP Tech Chat forum current topics</description>
<language>en</language>
<copyright>Copyright 2007, dslreports.com</copyright>
<pubDate>Sun, 08 Nov 2009 04:20:10 EDT</pubDate>
<lastBuildDate>Sun, 08 Nov 2009 04:20:10 EDT</lastBuildDate>

<image>
<title>dslreports.com</title>
<url>http://i.dslr.net/bbrdisc1.gif</url>
<link>http://www.dslreports.com</link>
<width>19</width>
<height>18</height>
<description>bbr disc</description>
</image>

<item>
<title>Question for the Asterisk gurus of this forum</title>
<link>http://www.dslreports.com/forum/remark,23308792</link>
<description><![CDATA[I have a question that has frustrated me for a few days now. Currently I have a PAP2T-NA that I use for Callcentric and an overseas VoIP provider without any problem. I can make and receive calls on both. Now, I thought it was time to experiment with Asterisk. I installed it on my Ubuntu computer, set the appropriate settings on the sip.conf and extensions.conf and I thought I would be ready to go. Not so easy, however. Let me backtrack, for a second. 

My home network consists of two chain-linked routers, a modem-router (network 192.168.1.xxx, I will call it subnet O) and a regular router (subnet 192.168.0.xxx, subnet I). Both the PAP2T and the asterisk are located on subnet I, not both at the same time. As I said, the ATA registers with both providers, without any problem, one line uses port 5060 and the other port 5061. No SIP-related ports have been opened on any of the routers. When I start the asterisk, it finds both providers (peers) unreachable, and I see on my (I) router a message such as
 "Unrecognized attempt blocked from 204.11.192.22:5080 to 192.168.1.105 UDP:63011"
204.11..... is the callcentric service, 192.168.1.105 is my (I) router 'WAN' address.

I am sure I will have more questions the deeper I delve into this service, but my question now is the following: How does the PA2T manage to bypass the (whatever) firewall of my router, while asterisk has a hard time with it? (I should note that if I set asterisk to listen to the subnet O IP address, it seems to register without any problem, but I have not made extensive tests on that. The computer where asterisk is installed has two ethernet cards.)

Any help will be greatly appreciated.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23308792</guid>
<pubDate>2009-11-08 01:05:15</pubDate>
</item>

<item>
<title>Free Google Voice calls with Windows Mobile 6</title>
<link>http://www.dslreports.com/forum/remark,23305085</link>
<description><![CDATA[Edit: link fixed

I picked up an HTC Tilt 2 (aka Touch Pro 2) from AT&T a couple of weeks ago. One of my first priorities was figuring out how to make free VoIP calls using Google Voice and Gizmo.

I finally got it working to my satisfaction today using the following free programs:

iDialer (Google Voice version)
iContact

Fring (configured with my Gizmo settings)

I use iDialer to originate the call through Google Voice, using my Gizmo5 account as the callback number. I use Fring to answer the call.

This works on both wifi and 3G. I've tested it only on my device using Windows Mobile 6.5, but I assume it should work on earlier versions and other devices as well.

I posted complete instructions on my blog for anyone who's interested:
http://www.fromthefencepost.com/2009/11/06/how-to-set-up-free-google-voice-calling-on-windows-mobile-65-htc-tilt-2-att/--
Detailed instructions and tips for Google Voice, Gizmo5, Linksys SPA2102, and Nokia N810: http://www.fromthefencepost.com/category/how-to/google-voice-gizmo5/

My home page: http://www.FromTheFencepost.com/]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23305085</guid>
<pubDate>2009-11-07 01:37:56</pubDate>
</item>

<item>
<title>[General] Chicago Going to 11 Digit Dialing</title>
<link>http://www.dslreports.com/forum/remark,23293672</link>
<description><![CDATA[Will this have any impact on VoIP calls to PSTN numbers? I doubt it, because a VoIP-originated call is delivered right to the CO of the called party, I think.

Anyway, I wonder if outbound CID could have the "1" prefixed, or would need to.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23293672</guid>
<pubDate>2009-11-04 22:06:56</pubDate>
</item>

<item>
<title>SPA-2102 slows my internet connection</title>
<link>http://www.dslreports.com/forum/remark,23307120</link>
<description><![CDATA[Howdy crew. 

Just noticed that my Sipura SPA-2102 really slows down my internet connection.

I did a speed test with computer plugged directly into my modem and another with SPA-2102 in between.

When the Sipura is being used, I get about half the throughput. 

Any thoughts? ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23307120</guid>
<pubDate>2009-11-07 16:49:15</pubDate>
</item>

<item>
<title>[Equipment] SPA2102 with PPPOE</title>
<link>http://www.dslreports.com/forum/remark,23289223</link>
<description><![CDATA[Hi,

This is a networking question, not really a SIP or VOIP question other than the device I'm using.  If there's a better place to post this please let me know.

I've got an SPA2102, and DSL and I'm using it in this way:

Modem (in bridge mode) --> SPA2102 (with PPPOE) --> Wireless Router in SPA DMZ

Everything works (SPA2102 connects, gets IP from ISP).  However, over time, I notice that latency between the SPA and the ISP gateway goes up.  Over the course of a couple of days ping reply times go up from 20ms to 2000ms (yes 2 seconds).  Cycling the power on the SPA (power down, wait a few seconds, power on) resets the latency back down to 20ms again.

Has anyone seen this?  Any ideas?  Is there any way to set MTU and connection timer settings on the SPA?

Thanks,
Lloyd]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23289223</guid>
<pubDate>2009-11-04 09:39:39</pubDate>
</item>

<item>
<title>Gizmo and GV</title>
<link>http://www.dslreports.com/forum/remark,23307509</link>
<description><![CDATA[Currently I use GV as my primary phone number. When people call my GV number, I have it forward to Gizmo then to F9 via SIP broker. I have to forward the call via SIP broker, since F9 will not accept calls directly from Gizmo.
I also have GV ring a DID from Callcentric, as a backup. This setup has worked well for the last several weeks.

For the last several days, when I get an incoming call (and answer it on my F9 line), it will drop after about 30 seconds into the conversation. Does anyone else have a similar setup, or have any suggestions as to what I can do?

I would just forward GV to my F9 DID, but this causes too much delay on the line and I cannot have a normal conversation.

At this point, I am just thinking of not using GV anymore. However, it seems like not many people (if any) have been able to port out.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23307509</guid>
<pubDate>2009-11-07 18:49:00</pubDate>
</item>

<item>
<title>Asterisk + Google Voice + Gizmo</title>
<link>http://www.dslreports.com/forum/remark,23304815</link>
<description><![CDATA[Hi Guys
I could not get my Google Voice to call into my pbx or extension and I have used this guide: http://geeklad.com/make-free-phone-calls-anywhere-in-the-usa-with-google-voice-gizmo-and-asterisk

what else can I use to get it to work.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23304815</guid>
<pubDate>2009-11-06 23:36:58</pubDate>
</item>

<item>
<title>A question about where to setup QoS</title>
<link>http://www.dslreports.com/forum/remark,23281855</link>
<description><![CDATA[On Sunday, I received an incoming call while I was watching a HD movie online. I had no problems to hear the other end. But the other end could only heard couple word here, there, not the entire conversation. I had no any problems if I didn't watch the movie.

I believe I should do the QoS. Here is my setup:
DSL modem -> Linksys WRT54GS -> Linksys PAP2-NA, as you can see the phone adapter is behind my router. Both the router and phone adapter support QoS. My questions is that; where I should setup the QoS?
1) on the router
2) on the phone adapter
3) on both

Thanks.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23281855</guid>
<pubDate>2009-11-02 21:54:10</pubDate>
</item>

<item>
<title>Sipgate - Lose phone # from inactivity?</title>
<link>http://www.dslreports.com/forum/remark,23304454</link>
<description><![CDATA[I had a free Sipgate One phone # I was using as a fax line. I didn't receive a fax for some time (perhaps 2 months), and upon logging in I discovered my phone # was gone. I could set up another phone #, but they already ran out of phone #s in my area code.

Does anyone know what activity keeps the phone # from being lost? (e.g. call/fax activity, logging into account, etc.)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23304454</guid>
<pubDate>2009-11-06 21:41:27</pubDate>
</item>

</channel>
</rss>
