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

[Stefan Agethen]

Hey Everyone,

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

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 )

I would be glad if someone can take a look...

Kindly regards,

Stefan