I have set SIP debug on but it is too much output to post on the mailinglist. I have tried to understand the SIP-messages between my Grandstream and my Asterisk-server and my Asterisk server and the ITSP. This is some output that's a bit shorter :
debug log : [Dec 27 12:11:32] DEBUG[14035] chan_sip.c: Oooh, we need to change our audio formats since our peer supports only 0x2 (gsm) and not 0x8 (alaw) [Dec 27 12:11:32] DEBUG[14035] chan_sip.c: Strict routing enforced for session 60eeb653339f6caa003dceab7ce38...@ip-asterisk-server [Dec 27 12:11:32] DEBUG[30447] chan_sip.c: Strict routing enforced for session 60eeb653339f6caa003dceab7ce38...@ip-asterisk-server [Dec 27 12:11:32] DEBUG[30447] rtp.c: Got a FRAME_CONTROL (8) frame on channel SIP/09277xxx7-1e547fc0 [Dec 27 12:11:33] DEBUG[14035] chan_sip.c: Strict routing enforced for session 60eeb653339f6caa003dceab7ce38...@ip-asterisk-server [Dec 27 12:11:33] DEBUG[14035] chan_sip.c: Strict routing enforced for session 60eeb653339f6caa003dceab7ce38...@ip-asterisk-server messages log : [Dec 27 12:11:32] NOTICE[14035] chan_sip.c: Failed to authenticate on INVITE to '<sip:09277x...@ip-itsp>;tag=as4ca755d8' Is it normal that I am able to call out ? Making calls is not a problem at all. There is also codec negotiation there, huh ?! Also : when I do not use the SIP proxy, receiving calls is not a problem. I just open up a range of SIP and RTP ports and forward these to the subnet my IP-phone is on. As I read your reactions, it seems not to be my SIP proxy ? I doubt that the SIP-proxy influences the codec negotiation. This some output of "sip show peer my-grandstream" : SIP Options : (none) Codecs : 0xa (gsm|alaw) Codec Order : (alaw:20,gsm:20) Auto-Framing: No Status : OK (43 ms) In my Grandstream I have the codecs : alaw - gsm (in that order) Jonas. On Sun, 2009-12-27 at 14:52 +0200, Tzafrir Cohen wrote: > > Sounds like the dialplan hangs up immediately. What do you see in the > CLI trace? >
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