[asterisk-users] Snom, Asterisk and Patton SN1400 - sending bye instead of hold

[Philipp Kempgen]

Stefan Agethen schrieb:
> Hey Everyone, once again - last time to publish this..

Hey, you posted this on Jun 8, 10 and 12.

> i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM
> 300,320,360 Devices.

Which firmware version is on the phones?

I can't remember having seen any problems on the Snoms related to
putting calls on hold.

You could try to post on http://groups.google.de/group/gemeinschaft-users
(gemeinschaft-users at googlegroups.com) even if you're not using
Gemeinschaft. Maybe somebody happens to know why that doesn't work.
Just make sure to say "Ich verwende zwar nicht Gemeinschaft, aber
vielleicht hat trotzdem jemand zufällig einen Tipp für mich."
And I'd start the subject with "OT" (off-topic).

> In the combination with asterisk and the patton, there are occuring some
> strange behaviour, due to the calling and answering everything works
> good, clear voice, great availability.
> All the devices have to use ulaw, alaw and slinear is available but
> never the first choice since i use my asterisk in europe. (slinear is
> available for debugging supposes)
> 
> But if a calls comes from or go to the SN1400 and someone tries to HOLD
> a call, the snoms are sending bye instead of hold, Asterisk plays his
> MOH until the bye reveives, the snoms doesnt understand this and thinks
> the caller is still on hold. In the SIP Debug i found some things which
> i cant handle, so i try to ask you whats going on there :
> 
> The call comes in, the patton routes it to asterisk and the codec invite
> starts :
> 
> --FROM PATTON TO ASTERISK--
> Found audio description format PCMU for ID 0
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0x4
> (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> 
> The last line is mysterious to me.
> 
> --ASTERISK IS INVITING  MY SNOM AT HOME--
> Audio is at [ I P - A S T E R I S K ] port 11576
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x40 (slin) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> 
> --SNOM IS ANSWERING THE CALL--
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 101
> Peer audio RTP is at port [ I P - A N G E R U F E N E R ]:13790
> Found audio description format pcmu for ID 0
> Found audio description format pcma for ID 8
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0xc
> (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> 
> The same as above..
> 
> --NOW I PRESS HOLD ON THE SNOM, THE SIP STATEMENT TO THE ASTERISK IS--
> 
> <--- SIP read from [ I P - A N G E R U F E N E R ]:5060 --->
> BYE sip:[ TEL. CALLER ]@[ I P - A S T E R I S K ] SIP/2.0
> Via: SIP/2.0/UDP [ I P - A N G E R U F E N E R
> ]:5060;branch=z9hG4bK-o6cb4olp9iv1;rport
> From: <sip:44@[ I P - A N G E R U F E N E R
> ]:5060;line=7anx8ofw>;tag=e8yr1936gy
> To: "[ MyName in the Snom ], " <"[ MyName in the Snom ], >;tag=as6fec2de7
> Call-ID: 055f1d8f752fcd8b52f0f3b71f89ef36@[ MyName in the Snom ].dyndns.org
> CSeq: 2 BYE
> Max-Forwards: 70
> Contact: <sip:44@[ I P - A N G E R U F E N E R
> ]:5060;line=7anx8ofw>;reg-id=1
> User-Agent: snom320/7.3.14
> Content-Length: 0
> 
> As you can see - a BYE is sent.
> 
> I tested it out many times, it only occures if a call comes from the
> patton, only sip calls can greatly be holded and transferred.
> The whole SIP DEBUG is available here, i dont wanted to post this
> stuff.. ( http://www.agethen.com/sip-debug-patton-snom.txt )


    Philipp Kempgen
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