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<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 17:46:42 EDT</pubDate>
<lastBuildDate>Sun, 08 Nov 2009 17:46:42 EDT</lastBuildDate>

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<item>
<title>Linksys SPA-942 SIP Registration Problem - Voip.ms</title>
<link>http://www.dslreports.com/forum/remark,23296656</link>
<description><![CDATA[Hi. I have a sip registration problem on 2 sub accounts that i have setuped at voip.ms.

I'm using 2 devices here :

1 - Linksys PAP2T, this one registers ok and stays registered.
2 - Linksys SPA-942, this one has registration issues since 2 days.

My setup is fairly simple, it's a SOHO with a Linksys WRT54G running tomato 1.25, and the devices connected to it.

Here is my issue :

When the phone is powered up, the registration works on the voip.ms server(s) (i've tried them all, tried ip's instead of using server name, wont change anything). 

If i place a call, during the call, i see the phone leds go to amber, which means there's a registration problem. I can stay on the call and it wont cut, but the phone isnt registered anymore. 

Checking the phone's web interface confirms it, it says 'Failed - Not Reachable' for the registration status of both accounts im using on it. If i hang up, it wont register anymore. If i power cycle the unit, it registers. 

I've contacted voip.ms on the issue, and was redirected to Steve which is a level 2 tech. He did some tracing on the problem, and here is the info he provided :

..   1.&#012;      154.330987 74.58.211.105 -&gt; 24.102.60.67 SIP Request: BYE sip:5148168690@24.102.60.67&#012;   2.&#012;      154.331533 24.102.60.67 -&gt; 74.58.211.105 SIP Status: 200 OK&#012;   3.&#012;      154.336407 74.58.211.105 -&gt; 24.102.60.67 ICMP Destination unreachable (Port unreachable)&#012;   4.&#012;      154.358235 74.58.211.105 -&gt; 24.102.60.67 ICMP Destination unreachable (Port unreachable)&#012;   5.&#012;      155.876784 74.58.211.105 -&gt; 24.102.60.67 SIP Request: NOTIFY sip:sip.ca2.voip.ms&#012;   6.&#012;      155.880697 74.58.211.105 -&gt; 24.102.60.67 SIP Request: NOTIFY sip:sip.ca2.voip.ms&#012;   7.&#012;      159.876347 74.58.211.105 -&gt; 24.102.60.67 SIP Request: NOTIFY sip:sip.ca2.voip.ms&#012;   8.&#012;      159.880444 74.58.211.105 -&gt; 24.102.60.67 SIP Request: NOTIFY sip:sip.ca2.voip.ms&#012;   9.&#012;      163.876170 74.58.211.105 -&gt; 24.102.60.67 SIP Request: NOTIFY sip:sip.ca2.voip.ms&#012;  10.&#012;      163.880152 74.58.211.105 -&gt; 24.102.60.67 SIP Request: NOTIFY sip:sip.ca2.voip.ms&#012;  11.&#012;      167.398064 74.58.211.105 -&gt; 24.102.60.67 SIP Request: REGISTER sip:sip.ca2.voip.ms&#012;  12.&#012;      167.398706 24.102.60.67 -&gt; 74.58.211.105 SIP Status: 100 Trying    (1 bindings)&#012;  13.&#012;      167.398897 24.102.60.67 -&gt; 74.58.211.105 SIP Status: 401 Unauthorized    (0 bindings)&#012;  14.&#012;      167.403280 74.58.211.105 -&gt; 24.102.60.67 SIP Request: REGISTER sip:sip.ca2.voip.ms&#012;  15.&#012;      167.403833 24.102.60.67 -&gt; 74.58.211.105 SIP Status: 100 Trying    (1 bindings)&#012;  16.&#012;      167.404012 24.102.60.67 -&gt; 74.58.211.105 SIP Status: 401 Unauthorized    (0 bindings)&#012;  17.&#012;      167.448122 74.58.211.105 -&gt; 24.102.60.67 SIP Request: REGISTER sip:sip.ca2.voip.ms&#012;  18.&#012;      167.448709 24.102.60.67 -&gt; 74.58.211.105 SIP Status: 100 Trying    (1 bindings)&#012;  19.&#012;      167.452355 74.58.211.105 -&gt; 24.102.60.67 SIP Request: REGISTER sip:sip.ca2.voip.ms&#012;  20.&#012;      167.467521 24.102.60.67 -&gt; 74.58.211.105 SIP Status: 200 OK    (1 bindings)&#012;  21.&#012;      167.468108 24.102.60.67 -&gt; 74.58.211.105 SIP Status: 100 Trying    (1 bindings)&#012;  22.&#012;      167.469203 24.102.60.67 -&gt; 74.58.211.105 SIP Status: 200 OK    (1 bindings)&#012;
He wondered if the ICMP packets sent by the phone was causing the problem. I've enabled logging in the tomato firmware, looked at the log, i can't see anything about this in there.

When the reset of the phone is done, and the phone registers OK at the voip.ms server, i see the registration status of the phone stays OK there, and the registration time matches the power cycle time. But the phone itself isnt reporting itself as registered. !?!

The setup was working for 1 month, and i haven't done any configuration changes in the last days (maybe the spa-942 updated it's firmware itself, that i can't tell, it seems to be automatic up to some point).

If anyone has a clue that could possibly fix my issue, go ahead, i'd be glad to get my buisness phone working again :S

Thanks, Vincent
]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23296656</guid>
<pubDate>2009-11-05 13:37:56</pubDate>
</item>

<item>
<title>[Other] Hoax about cell numbers</title>
<link>http://www.dslreports.com/forum/remark,23309234</link>
<description><![CDATA[An email this morning from a friend.  I set him straight.

http://www.snopes.com/politics/business/cell411.asp

Looks like another (private) forum saw this back in January!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23309234</guid>
<pubDate>2009-11-08 07:53:17</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>[Skype] Windows 7 and SKYPE</title>
<link>http://www.dslreports.com/forum/remark,23309876</link>
<description><![CDATA[I just upgraded to W7 from Vista and Skype does not function.

Am I the only one ???????

Thanks]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23309876</guid>
<pubDate>2009-11-08 12:02:06</pubDate>
</item>

<item>
<title>VOXOX with ATA</title>
<link>http://www.dslreports.com/forum/remark,23275914</link>
<description><![CDATA[I am surprised that the ATA worked with them. X-Lite had no sound either way.
A different post on DSL Reports showed their Codec to be G723, and X-lite did not have that Codec. 
With the parameters of User ID: 1xxxxxxxxxx (Voxox number including the 1), and the password that turns on the application, and of course setting the preferred codec to G723, it worked.
To keep the ATA working, I sign into the application as invisible. I only sign in to change the outgoing CID for fun.
This is the only application (that I know of) without a monthly fee, and only One cent per minute* that lets you SPOOF your caller ID at will, even to a number that you don't have.
*If you pay $2.45 per month, minutes are virtually unlimited.
I have paid for lines (Magic Jack and Future Nine, and of course my DSL) but I like to occasionally play around with "free" services.
Their website indicates that you can have as many free numbers as yow want in your account (probably California numbers) but I haven't been able to find the application to add additional incoming numbers. If someone else knows how to get additional numbers within the same account, please post it. 
I will test the voice quality with my friends. If it is decent, I will pay $2.45 monthly for unlimited minutes. 
Sure beats paying $100.00 up front for TK6000 or $230.00 up front for Ooma]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23275914</guid>
<pubDate>2009-11-01 20:05:45</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>[Broadvox Direct] Castaway hunting around. ....</title>
<link>http://www.dslreports.com/forum/remark,23216244</link>
<description><![CDATA[So I am wondering what many of you Broadvox castaways are going to do TOO?  Stay with PP or look for other provider since it seems like it a good time to revisit VOIP.  

Below are my observations if you care to read them or just post your response to the question above is welcome too. 

I been hunting around the different providers looking at  pricing  then looking at "true pricing" and at what point the provider decides to show that true pricing.  (Hey I am a Saturn owner give me a break).   Prior to this month I really had no clue that Broadvox would want to move us out. I knew that they would not take anyone new in since more than a year ago but never thought we be moved out.  So, not to dwell on how much better facilities based companies are usually I going to stay to point in this post.
And again prior to this month my focus I would say was 75% what the true price of the service is. Now I come to realize hunting around I was in a unique situation with Broadvox not charging for 911 or fees or so it seemed externally to me because the onllne statements would just say $12.95 was what I was paying and that its. 
There are a few providers I was looking at  AND I took a closer look at Phone Power.

The only provider that I liked how the pricing is laid out is Quantum Voice. 

Among many other providers these were a few that stood out little bit to me.... 
Voipo
AxVoice
inPhonex
VoiceEclipse---I thought maybe these guys would be close to being a product of a Facilities based company. It's ironic the Parent company Paetec is the same company my formal place of employment used for internet/phone.  And they were very good for service and support. I think it was like 2 outages in like 6 years.  Anyway I called the number for VoiceEclipse sales and It went to a receptionist She got my number, email and name and said sales would get back to me to answer my questions I not heard back today.  I noticed there web site is  dated to 2007 and it talks about Sunrocket too. I have a feeling that I not getting a call back. I even emailed the sales rep I knew from the Paetec side and he said he would try to find some things out for me too.   
I also looked at FutureNine and Callcentric 
And of course in greater detail Phone Power.  The 2nd year free deal no contract is to hard to pass up. 

My needs are mostly for incoming calls, light usage for local out calling, and calls to Australia and Greece.  That's it. 
Phone Power is not such a bad deal for the two year no contract.  I going to see if I can get into that tomorrow. 
Enough about price now service and ratings I looked at that to but I also take it with a grain of salt too reading some of of those too.   I would expect around 10 to 20% being bad reviews the rest being good reviews and some of the providers almost all good reviews (which is almost to good to be true).   To me I value price as 51%,Services tech support provided for rest.

So if your staying with Phone Power I want to hear your reason as well as not staying.  
Since I landed with Power Phone I came to my senses looking at the details I going to see if I can get the 2yr no contract deal. 
I think that Phone Power does use the same system the Broadvox used  what was it called IP unity? Maybe even they use the carrier services too does anyone know?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23216244</guid>
<pubDate>2009-10-21 00:48:59</pubDate>
</item>

<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>SIPGATE - Incoming calls terminate after 15 minutes</title>
<link>http://www.dslreports.com/forum/remark,23309421</link>
<description><![CDATA[I have been using SIPGATE service for about a week and noticed that after 15 minutes of an incoming call, the call gets terminated.

Anyone else have this issue?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23309421</guid>
<pubDate>2009-11-08 09:40:22</pubDate>
</item>

<item>
<title>[Equipment] FXO recommendation</title>
<link>http://www.dslreports.com/forum/remark,23307508</link>
<description><![CDATA[Anyone have any recommendations on FXO's? I have a good idea of the common FXO's that exist, but maybe someone could recommend something we haven't heard of.

If you really recommend one, or dislike one, please let me know.
It can be pci or ata based.

Thanks]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,23307508</guid>
<pubDate>2009-11-07 18:48:50</pubDate>
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