Displaying 3 results from an estimated 3 matches for "invitesip".
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invites
2020 May 14
0
I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
...;
> Last minute thought: Is it possible that the caller is sending g729 in
> RTP even though the SIP negotiation clearly chooses alaw? Maybe I
> need some RTP debugging.
>
And in fact that is exactly what's happening.
>
> <--- SIP read from UDP:SUPPLIER:5060 --->
> INVITEsip:LOCAL at ASTERISK:5060 SIP/2.0
> Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9
> From:<sip:REMOTE at SUPPLIER>;tag=gK02498cb1
> To:<sip:LOCAL at ASTERISK>
> Call-ID: 205665777_90679951 at SUPPLIER
> CSeq: 539098 INVITE
> Max-Forwards: 70
> Allow:...
2010 Dec 20
2
SIP 420
Hi;
I am running asterisk 1.6 from Fonality (Trixbox PRO).
I am trying to initiate a call FROM a softphone client to asterisk (either
an internal 4 digit extension call) or an outside line via a SIP trunk.
In both cases, asterisk rejects the call with a 420.
In this case, it?s a call from x3992 to x4415
Does this require a change on the softphone for x-call-detail?
<--- SIP read
2020 May 14
6
I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
...l hangs up. Why?
Last minute thought: Is it possible that the caller is sending g729 in
RTP even though the SIP negotiation clearly chooses alaw? Maybe I need
some RTP debugging.
Asterisk 13.14.1 on Debian, using chan_sip.
Here's the trace:
<--- SIP read from UDP:SUPPLIER:5060 --->
INVITEsip:LOCAL at ASTERISK:5060 SIP/2.0
Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9
From:<sip:REMOTE at SUPPLIER>;tag=gK02498cb1
To:<sip:LOCAL at ASTERISK>
Call-ID: 205665777_90679951 at SUPPLIER
CSeq: 539098 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REF...