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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 :<BR>
<BR>
debug log :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[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)</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Dec 27 12:11:32] DEBUG[14035] chan_sip.c: Strict routing enforced for session <A HREF="mailto:60eeb653339f6caa003dceab7ce38d17@89.31.97.186">60eeb653339f6caa003dceab7ce38d17@ip-asterisk-server</A></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Dec 27 12:11:32] DEBUG[30447] chan_sip.c: Strict routing enforced for session <A HREF="mailto:60eeb653339f6caa003dceab7ce38d17@89.31.97.186">60eeb653339f6caa003dceab7ce38d17@ip-asterisk-server</A></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Dec 27 12:11:32] DEBUG[30447] rtp.c: Got a FRAME_CONTROL (8) frame on channel SIP/09277xxx7-1e547fc0</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Dec 27 12:11:33] DEBUG[14035] chan_sip.c: Strict routing enforced for session <A HREF="mailto:60eeb653339f6caa003dceab7ce38d17@89.31.97.186">60eeb653339f6caa003dceab7ce38d17@ip-asterisk-server</A></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Dec 27 12:11:33] DEBUG[14035] chan_sip.c: Strict routing enforced for session <A HREF="mailto:60eeb653339f6caa003dceab7ce38d17@89.31.97.186">60eeb653339f6caa003dceab7ce38d17@ip-asterisk-server</A></FONT></FONT><BR>
<BR>
messages log :<BR>
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<FONT COLOR="#0000ff"><FONT SIZE="2">[Dec 27 12:11:32] NOTICE[14035] chan_sip.c: Failed to authenticate on INVITE to '<<A HREF="sip:092779077@85.119.188.3">sip:09277xxx7@ip-ITSP</A>>;tag=as4ca755d8'</FONT></FONT><BR>
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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 ?!<BR>
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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.<BR>
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As I read your reactions, it seems not to be my SIP proxy ? I doubt that the SIP-proxy influences the codec negotiation.<BR>
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This some output of "sip show peer my-grandstream" :<BR>
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<FONT COLOR="#0000ff"><FONT SIZE="2">SIP Options : (none)</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> Codecs : 0xa (gsm|alaw)</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> Codec Order : (alaw:20,gsm:20)</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> Auto-Framing: No </FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> Status : OK (43 ms)</FONT></FONT><BR>
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In my Grandstream I have the codecs : alaw - gsm (in that order)<BR>
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<BR>
Jonas.<BR>
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<BR>
On Sun, 2009-12-27 at 14:52 +0200, Tzafrir Cohen wrote:
<BLOCKQUOTE TYPE=CITE>
<PRE>
Sounds like the dialplan hangs up immediately. What do you see in the
CLI trace?
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