[arm-allstar] Polycom SIP phone hangs up 32 seconds after nodestatus (solved)
Phil Visalli
phil at philv.net
Thu Apr 14 21:13:21 EST 2016
I dont know exactly the cause but i've got a solution. On the phone, under
lines, for the "address" field i was just using the server IP, and filling
in the userid in the "Authentication User ID" field. Doing that everything
worked except that disconnect issue when calling one of my two allstar
nodes.
Turns out you need to use userid at serverIP (not just serverIP) in the
"address" field (and also userid in the "Authentication User ID" field) and
everything now works as it should. I've kept a call up for 30 min and a
bunch of 5 min calls so I think im good.
Thanks for the sugguestions.
Phil V
K2ELV
On Thu, Apr 14, 2016 at 4:59 PM, <arm-allstar-request at hamvoip.org> wrote:
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> Today's Topics:
>
> 1. Re: Polycom SIP phone hangs up 32 seconds after nodestatus
> (Stanley Stanukinos)
> 2. Time Announcement Cron (Marty)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 14 Apr 2016 15:51:00 -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
> nodestatus
> Message-ID: <526F5A9D-4D01-49AA-8445-755112583DB4 at swbell.net>
> Content-Type: text/plain; charset="us-ascii"
>
> Phil, no problem. Sometimes more data is better. Wireshark has caught
> things in my work world that uses SIP and RTP that were not behaving as
> expected.
>
> Stan
>
> Sent from my iPhone
>
> > On Apr 14, 2016, at 3:41 PM, Phil Visalli via arm-allstar <
> arm-allstar at hamvoip.org> wrote:
> >
> > 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:
> >> Send arm-allstar mailing list submissions to
> >> arm-allstar at hamvoip.org
> >>
> >> To subscribe or unsubscribe via the World Wide Web, visit
> >> http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar
> >> or, via email, send a message with subject or body 'help' to
> >> arm-allstar-request at hamvoip.org
> >>
> >> You can reach the person managing the list at
> >> arm-allstar-owner at hamvoip.org
> >>
> >> When replying, please edit your Subject line so it is more specific
> >> than "Re: Contents of arm-allstar digest..."
> >>
> >>
> >> Today's Topics:
> >>
> >> 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
> >> -------------- next part --------------
> >> An HTML attachment was scrubbed...
> >> URL: <
> http://lists.hamvoip.org/pipermail/arm-allstar/attachments/20160414/c4bad582/attachment-0001.html
> >
> >>
> >> ------------------------------
> >>
> >> 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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> >> ------------------------------
> >>
> >> Subject: Digest Footer
> >>
> >> _______________________________________________
> >> arm-allstar mailing list
> >> arm-allstar at hamvoip.org
> >> http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar
> >>
> >>
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> >>
> >> End of arm-allstar Digest, Vol 23, Issue 16
> >> *******************************************
> >
> > _______________________________________________
> >
> > 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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> ------------------------------
>
> Message: 2
> Date: Thu, 14 Apr 2016 16:59:46 -0400
> From: Marty <kd4hlv at gmail.com>
> To: arm-allstar at hamvoip.org
> Subject: [arm-allstar] Time Announcement Cron
> Message-ID:
> <CAO=2xhof5OG=O38U49e07=
> DDJXfbE5Fcyo3eerqfKH7_mL5m2g at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> I have a question about Time Announcement Cron. The Announcem plays 24
> hours a day I've been looking for the cron that I can change to say 7AM
> till 11PM I know how to change it on an IRLP computer under custom crons
> but allstar does nit use them. I seen where it might be in the
> say24time.pl
> or saytime.pl but can't find the
> 00 0-23 * * * is located.
>
>
> Marty KD4HLV
>
> 28523
>
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>
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