[asterisk-users] Call ends when picked up

jonas kellens jonas.kellens at telenet.be
Sun Dec 27 07:54:30 CST 2009


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 60eeb653339f6caa003dceab7ce38d17 at ip-asterisk-server
[Dec 27 12:11:32] DEBUG[30447] chan_sip.c: Strict routing enforced for
session 60eeb653339f6caa003dceab7ce38d17 at 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 60eeb653339f6caa003dceab7ce38d17 at ip-asterisk-server
[Dec 27 12:11:33] DEBUG[14035] chan_sip.c: Strict routing enforced for
session 60eeb653339f6caa003dceab7ce38d17 at ip-asterisk-server

messages log :

[Dec 27 12:11:32] NOTICE[14035] chan_sip.c: Failed to authenticate on
INVITE to '<sip:09277xxx7 at 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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