[Asterisk-Users] G723 error

[yusuf]

Matt,

I you dont define a sip user/peer and just use a dial, asterisk will 
automatically use the codec that it prefers,  in my experince whenever i 
dial SIP without defining a sip user/peer it always dials g711alaw/ulaw.

So in sip.conf in [general] (which would set codec choice for ALL sip 
calls) or in the defined section for sip device/user/peer have
disallow=all
allow=g723


Matt wrote:
> Well... correct except that there is no [sipdevice].. it is all done
> through IP registration on the other person's end.    So.. all I have
> is the dial statement.  Is there a way to set a variable or something
> right before the dial? (To my knowledge there isn't).
> 
> On 2/15/06, yusuf <yusuf at ecntelecoms.com> wrote:
> 
>>I am assuming you made a profile in sip.conf like so
>>
>>[sipdevice]
>>type=peer
>>host=x.x.x.x
>>...
>>.
>>.
>>disallow=all
>>allow=ulaw
>>
>>and in extensions.conf
>>
>>exten => _X.,1,Dial(SIP/sipdevice/${EXTEN})
>>
>>then this MUST work.  :)
>>
>>you can do a sip debug or set debug 10
>>
>>yusuf
>>
>>Matt wrote:
>>
>>>Hi,
>>>How do I specify a codec to use for a SIP call?
>>>
>>>IE.. If I'm doing Dial(SIP/blah) for some reason the call is
>>>connecting using the codec at the bottom of my allow list rather then
>>>top (G711u)... and I'd like to force it to G711u if possible.
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>>
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> 
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