[arm-allstar] Polycom SIP phone hangs up 32 seconds after nodestatus
Phil Visalli
phil at philv.net
Thu Apr 14 15:41:43 EST 2016
Stanley,
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.
Thanks for the nudge .... i'll report back when i get some more info
On Thu, Apr 14, 2016 at 1:00 PM, <arm-allstar-request at hamvoip.org> wrote:
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> 1. Polycom SIP phone hangs up 32 seconds after node status
> (Phil Visalli)
> 2. Re: Polycom SIP phone hangs up 32 seconds after node status
> (Stanley Stanukinos)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 14 Apr 2016 11:26:09 -0400
> From: Phil Visalli <phil at philv.net>
> To: arm-allstar at hamvoip.org
> Subject: [arm-allstar] Polycom SIP phone hangs up 32 seconds after
> node status
> Message-ID:
> <
> CAAMk9h_hXSQ2URNXkFh5eL331BGdeJztLTdSiOHkGvLv_NrTFQ at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> 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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> ------------------------------
>
> Message: 2
> Date: Thu, 14 Apr 2016 10:52:42 -0500
> From: Stanley Stanukinos <ka5iid at swbell.net>
> To: ARM Allstar <arm-allstar at hamvoip.org>
> Subject: Re: [arm-allstar] Polycom SIP phone hangs up 32 seconds after
> node status
> Message-ID: <841D1E0C-0F5E-4F82-9A03-8602245BCE8E at swbell.net>
> Content-Type: text/plain; charset="us-ascii"
>
> 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
> > _______________________________________________
> >
> > arm-allstar mailing list
> > arm-allstar at hamvoip.org
> > http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar
> >
> > Visit the BBB and RPi2 web page - http://hamvoip.org
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