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<body class='hmmessage'><div dir='ltr'>Hi Phil,<div><br></div><div>You can look at the debug output using this command from the CLI</div><div><br></div><div><span style="color: rgb(64, 64, 64); font-family: 'courier new', courier; font-size: 10pt; line-height: 18px; background-color: rgb(255, 255, 255);">sip set debug</span></div><div><br></div><div><font face="Arial, sans-serif">73,</font></div><div><font face="Arial, sans-serif"><br></font></div><div><font face="Arial, sans-serif">Lito KK6OOS</font></div><div><br><div><hr id="stopSpelling">Date: Thu, 14 Apr 2016 16:41:43 -0400<br>To: arm-allstar@hamvoip.org<br>Subject: Re: [arm-allstar] Polycom SIP phone hangs up 32 seconds after nodestatus<br>From: arm-allstar@hamvoip.org<br>CC: phil@philv.net<br><br><div dir="ltr">Stanley,<div><br></div><div>I have not, but i just installed wireshark and will run a packet capture when i get home tonight if i have time (gotta finish my taxes d'oh!)...... otherwise i'll sit down to do some tinkering for sure tomorrow after work. I guess i didn't even think about doing a packet capture because I didn't think it was a network related issue. I thought i'd be able to get any debug info i needed from the asterisk console. </div><div><br></div><div>Thanks for the nudge .... i'll report back when i get some more info</div><div><br><div class="ecxgmail_extra"><br><div class="ecxgmail_quote">On Thu, Apr 14, 2016 at 1:00 PM, <span dir="ltr"><<a href="mailto:arm-allstar-request@hamvoip.org" target="_blank">arm-allstar-request@hamvoip.org</a>></span> wrote:<br><blockquote class="ecxgmail_quote" style="border-left:1px #ccc solid;padding-left:1ex;">Send arm-allstar mailing list submissions to<br>
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Today's Topics:<br>
<br>
1. Polycom SIP phone hangs up 32 seconds after node status<br>
(Phil Visalli)<br>
2. Re: Polycom SIP phone hangs up 32 seconds after node status<br>
(Stanley Stanukinos)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Thu, 14 Apr 2016 11:26:09 -0400<br>
From: Phil Visalli <<a href="mailto:phil@philv.net" target="_blank">phil@philv.net</a>><br>
To: <a href="mailto:arm-allstar@hamvoip.org" target="_blank">arm-allstar@hamvoip.org</a><br>
Subject: [arm-allstar] Polycom SIP phone hangs up 32 seconds after<br>
node status<br>
Message-ID:<br>
<<a href="mailto:CAAMk9h_hXSQ2URNXkFh5eL331BGdeJztLTdSiOHkGvLv_NrTFQ@mail.gmail.com" target="_blank">CAAMk9h_hXSQ2URNXkFh5eL331BGdeJztLTdSiOHkGvLv_NrTFQ@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
I know this might be a little bit outside the scope of this mailing list<br>
but i figured I'd try and see if anyone can point me in the right direction.<br>
<br>
For starters, Im still running the RPi2 V1.0 version, but i dont think that<br>
has anything to do with the problem im having.<br>
<br>
I've got 2 nodes, one radio node, and one hub node. One Polycom 550 SIP<br>
phone and 2 sip softphone clients running on android devices. I've also<br>
got autopatch set up using a gvoice number for a sip trunk. The three sip<br>
extensions can call each other, call either allstar node, or use the gvoice<br>
number to call out. Also, from the radio node I can use autopatch to dial<br>
out on the gvoice line or "call" the polycom sip phone (*611 rings the<br>
phone ext). Audio and all the call routing in all these scenarios works<br>
great.<br>
<br>
Now to the problem. When i call from the Polycom to either of my two<br>
nodes, the sip phone seems to be hanging up at what appeared to be random<br>
intervals. After further testing I've figured out that its hanging up<br>
about 30-32 seconds after the node does something like playing an ID or *70<br>
to play node status. Watching the asterisk console I see this:<br>
<br>
-- <DAHDI/pseudo-672364238> Playing 'rpt/node' (language 'en') --<br>
<DAHDI/pseudo-672364238> Playing 'digits/4' (language 'en') --<br>
<DAHDI/pseudo-672364238> Playing 'digits/2' (language 'en') --<br>
<DAHDI/pseudo-672364238> Playing 'digits/1' (language 'en') --<br>
<DAHDI/pseudo-672364238> Playing 'digits/9' (language 'en') --<br>
<DAHDI/pseudo-672364238> Playing 'digits/0' (language 'en') --<br>
<DAHDI/pseudo-672364238> Playing 'rpt/repeat_only' (language 'en') --<br>
Hungup 'DAHDI/pseudo-672364238'<br>
<br>
Which leads me to believe that the phone is seeing that "hungup" at the end<br>
of the server status message and thinks that the call is over and therefore<br>
it hangs up. Its worth mentioning that this ONLY happens on the polycom<br>
phone and not on the softphone clients (zoiper). I've gone over all the<br>
settings on the phone and nothing seems to change this behavior. I can be<br>
passing audio back and forth for several min with no problem, but as soon<br>
as I do something like *70, 30ish seconds later the polycom hangs up.<br>
<br>
While i'd love it if someone has an answer, id be happy to find out if<br>
anyone has run into this before.... or if someone has a similar setup that<br>
could discuss with me (either on the mailing list, offline or on the<br>
radio). Im going to try to get my hands on another brand sip phone to see<br>
if it happens on others too. Again, sorry if this is the wrong place for<br>
this but it seems to be the most active allstar specific community so<br>
hopefully no one minds me dropping my problem here :p<br>
<br>
Thanks<br>
<br>
Phil V<br>
K2ELV<br>
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Message: 2<br>
Date: Thu, 14 Apr 2016 10:52:42 -0500<br>
From: Stanley Stanukinos <<a href="mailto:ka5iid@swbell.net" target="_blank">ka5iid@swbell.net</a>><br>
To: ARM Allstar <<a href="mailto:arm-allstar@hamvoip.org" target="_blank">arm-allstar@hamvoip.org</a>><br>
Subject: Re: [arm-allstar] Polycom SIP phone hangs up 32 seconds after<br>
node status<br>
Message-ID: <<a href="mailto:841D1E0C-0F5E-4F82-9A03-8602245BCE8E@swbell.net" target="_blank">841D1E0C-0F5E-4F82-9A03-8602245BCE8E@swbell.net</a>><br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
Phil, have you ran Wireshark and captured the data on the Lan to see what is going on?<br>
<br>
Stan<br>
<br>
Sent from my iPhone<br>
<br>
> On Apr 14, 2016, at 10:26 AM, Phil Visalli via arm-allstar <<a href="mailto:arm-allstar@hamvoip.org" target="_blank">arm-allstar@hamvoip.org</a>> wrote:<br>
><br>
> I know this might be a little bit outside the scope of this mailing list but i figured I'd try and see if anyone can point me in the right direction.<br>
><br>
> For starters, Im still running the RPi2 V1.0 version, but i dont think that has anything to do with the problem im having.<br>
><br>
> I've got 2 nodes, one radio node, and one hub node. One Polycom 550 SIP phone and 2 sip softphone clients running on android devices. I've also got autopatch set up using a gvoice number for a sip trunk. The three sip extensions can call each other, call either allstar node, or use the gvoice number to call out. Also, from the radio node I can use autopatch to dial out on the gvoice line or "call" the polycom sip phone (*611 rings the phone ext). Audio and all the call routing in all these scenarios works great.<br>
><br>
> Now to the problem. When i call from the Polycom to either of my two nodes, the sip phone seems to be hanging up at what appeared to be random intervals. After further testing I've figured out that its hanging up about 30-32 seconds after the node does something like playing an ID or *70 to play node status. Watching the asterisk console I see this:<br>
><br>
> -- <DAHDI/pseudo-672364238> Playing 'rpt/node' (language 'en')<br>
> -- <DAHDI/pseudo-672364238> Playing 'digits/4' (language 'en')<br>
> -- <DAHDI/pseudo-672364238> Playing 'digits/2' (language 'en')<br>
> -- <DAHDI/pseudo-672364238> Playing 'digits/1' (language 'en')<br>
> -- <DAHDI/pseudo-672364238> Playing 'digits/9' (language 'en')<br>
> -- <DAHDI/pseudo-672364238> Playing 'digits/0' (language 'en')<br>
> -- <DAHDI/pseudo-672364238> Playing 'rpt/repeat_only' (language 'en')<br>
> -- Hungup 'DAHDI/pseudo-672364238'<br>
><br>
> Which leads me to believe that the phone is seeing that "hungup" at the end of the server status message and thinks that the call is over and therefore it hangs up. Its worth mentioning that this ONLY happens on the polycom phone and not on the softphone clients (zoiper). I've gone over all the settings on the phone and nothing seems to change this behavior. I can be passing audio back and forth for several min with no problem, but as soon as I do something like *70, 30ish seconds later the polycom hangs up.<br>
><br>
> While i'd love it if someone has an answer, id be happy to find out if anyone has run into this before.... or if someone has a similar setup that could discuss with me (either on the mailing list, offline or on the radio). Im going to try to get my hands on another brand sip phone to see if it happens on others too. Again, sorry if this is the wrong place for this but it seems to be the most active allstar specific community so hopefully no one minds me dropping my problem here :p<br>
><br>
> Thanks<br>
><br>
> Phil V<br>
> K2ELV<br>
> _______________________________________________<br>
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><br>
> Visit the BBB and RPi2 web page - <a href="http://hamvoip.org" rel="noreferrer" target="_blank">http://hamvoip.org</a><br>
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