[Asterisk-Dev] SIP_CODEC not working anymore or differently in version 1.0.0?

[Andreas Greulich]

Hi all,

I'm using teh SIP_CODEC variable in my extensions.conf to "force" the caling
device to some codec (say, ilbc) for outbound calls:

[sipgate-out]
exten => h,1,Hangup
exten => _X.,1,SetCallerID(${SIPGATEID})
exten => _X.,2,SetVar(SIP_CODEC=ilbc)
exten => _X.,3,Dial(SIP/${EXTEN}@Sipgate,60,)

This is working perfectly under 1.0-RC2:


>....
    -- Executing SetCallerID("SIP/GSIn-131d", "1838074") in new stack
    -- Executing SetVar("SIP/GSIn-131d", "SIP_CODEC=ilbc") in new stack
    -- Executing Dial("SIP/GSIn-131d", "SIP/0041313249211 at Sipgate|60|") in new stack
    -- Called 0041313249211 at Sipgate
    -- SIP/Sipgate-4f13 is making progress passing it to SIP/GSIn-131d
    -- SIP/Sipgate-4f13 answered SIP/GSIn-131d
Sep 26 12:05:52 NOTICE[262159]: chan_sip.c:1817 sip_answer: Changing codec to
'ilbc' for this call because of ${SIP_CODEC) variable
    -- Attempting native bridge of SIP/GSIn-131d and SIP/Sipgate-4f13
    -- Attempting native bridge of SIP/GSIn-131d and SIP/Sipgate-4f13
rufus*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)   Format
217.10.79.9      0041313249  7bb7c0d1525  00103/00000   ILBC  
10.1.41.188      GSIn        59d8eec61ae  00101/43829   ILBC  
2 active SIP channel(s)


After installing 1.0.0, the same process leaves GSIn on codec ULAW (actually
performing codec translation), the rest of teh output is the same (only the
NOTICE message is produced in line 1844 instead of 1817); don't have the precise
logfile at hand right now.

Is this a known problem, or am I doing something wrong? Can otehrs reproduce the
problem?

for the moment, I'm using 1.0-RC2 again because of this (which works for me).

Andy



-- 

Andreas Greulich
E-Mail: andreas.greulich at spl.ch
Skype: klaymen-neverhood

Sermo datur cunctis, animi sapientia paucis.