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:4...@[ 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: 055f1d8f752fcd8b52f0f3b71f89e...@[ MyName in the Snom ].dyndns.org CSeq: 2 BYE Max-Forwards: 70 Contact: <sip:4...@[ 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
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