[Asterisk-Users] SIP 404 - circuit busy when dialing out

[Cinoss]

Thanks for reply. Yes i am getting audio. It hangs-up automaticly after
10 secs, or the line goes down. Softphone has the line still open
though.

I dont get this 404 anymore, it was just before the missing canreinvite=



-----Original Message-----
Cinoss,

Are you getting audio during the call? 
Or are you just seeing the call setup?
Is it really 10 seconds? Or just seems like it?

I have seen the SIP 404 when the codec matches were incorrect.

With Debug and Versobose, -dvvvvvgc, on in Asterisk look for messages
about codec matching.

Also turn on sip debugging. "*CLI> sip debug"

Found description format CN
Capabilities: us - 0x10e(GSM|ULAW|ALAW|G729A), peer -
audio=0x105(G723|ULAW|G729A)/video=0x0(EMPTY), combined -
0x104(ULAW|G729A)

Non-codec capabilities: us - 0x1(G723), peer - 0x3(G723|GSM), combined -
0x1(G723)
Urgent handler

If "combined" equals empty then you need to adjust the codecs.




-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Cinoss
Sent: 19 October 2004 13:04
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] SIP 404 - circuit busy when dialing out

Well i have now sorted dialing out. Only needed to add fromdomain= to my
[sipprovider].
Still got small problem with it. The call gets automaticly hang-up after
10secs. I tried both canreinvite=no and yes and my sip.conf but it
doesn't seem to do any difference, well other than fail code.
-- 
  Cinoss
  cinosss at f-m.fm

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-- 
  Cinoss
  cinosss at f-m.fm