[arm-allstar] Polycom SIP phone hangs up 32 seconds after node status

[Stanley Stanukinos]

Phil, have you ran Wireshark and captured the data on the Lan to see what is going on?

Stan

Sent from my iPhone

> On Apr 14, 2016, at 10:26 AM, Phil Visalli via arm-allstar <arm-allstar at hamvoip.org> wrote:
> 
> 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.
> 
> For starters, Im still running the RPi2 V1.0 version, but i dont think that has anything to do with the problem im having. 
> 
> 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.
> 
> 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:
> 
> -- <DAHDI/pseudo-672364238> Playing 'rpt/node' (language 'en')
> -- <DAHDI/pseudo-672364238> Playing 'digits/4' (language 'en')
> -- <DAHDI/pseudo-672364238> Playing 'digits/2' (language 'en')
> -- <DAHDI/pseudo-672364238> Playing 'digits/1' (language 'en')
> -- <DAHDI/pseudo-672364238> Playing 'digits/9' (language 'en')
> -- <DAHDI/pseudo-672364238> Playing 'digits/0' (language 'en')
> -- <DAHDI/pseudo-672364238> Playing 'rpt/repeat_only' (language 'en')
> -- Hungup 'DAHDI/pseudo-672364238'
> 
> 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.
> 
> 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
> 
> Thanks
> 
> Phil V
> K2ELV
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