To: sip:[EMAIL PROTECTED];tag=as2ae322ec
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 192.168.1.13:5060
DEBUG[311314]: File chan_sip.c, Line 1025 (sip_hangup): find_user(7601)
- decrement inUse counter
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060
From: User ID
sip:[EMAIL PROTECTED];tag=000b5f800a9b010359818116-229fbd39
To: sip:[EMAIL PROTECTED];tag=as2ae322ec
Call-ID: [EMAIL PROTECTED]
Date: Tue, 21 Oct 2003 17:38:37 GMT
CSeq: 102 ACK
Content-Length: 0
On Tue, 2003-10-21 at 12:39, Martin Pycko wrote:
[extensions.conf]
exten = 123456,1,SetVar,SIP_CODEC=ulaw
exten = 123456,2,Dial(${TRUNK}/${EXTEN})
The problem is with the SetVar function, the debug shows that the
function is executed, but after that, * sends the media capability to
the phone with g729 as preferred codec.
SIP_CODEC is was supposed to only change the codec of the incoming call,
eg: asterisk responds with ANSWER with ulaw codec ...
But it won't change anything with the 2nd call.
regards
Martin
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Luis Benavente [EMAIL PROTECTED]
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