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

Phil Visalli phil at philv.net
Thu Apr 14 10:26:09 EST 2016

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

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


Phil V
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