[asterisk-users] Asterisk not using common codec between (SIP) endpoints

Andreas Wehrmann a.wehrmann at yandex.com
Thu Oct 3 09:10:05 CDT 2019

On 03.10.19 15:08, Administrator TOOTAI wrote:

> Before calling the gatreway add
> same = n,set(SIP_CODEC=alaw)
> [...]

Hey there,

that doesn't work as it seems to be implemented for chan_sip only;
I'm using chan_pjsip; sorry if I didn't explain myself properly.

Anyway, in my case that would not really be an acceptable solution anyway,
because I need the called party to be able to pick from the range of 
codecs presented to it
because the codec chosen by the destination might change (my example is 
a simplified version).

I don't think putting the burden of worrying about audio codecs on the 
dialplan writer is a good idea,
since this should be dealt with automatically with respect to what is 
configured and negotiated.
This is also because in the systems I have to work with, the 'engineers' 
usual provide the configuration (endpoints, NAT config and the like)
while the technicians implement the dialplan (or the business logic so 
to speak) according to customer needs.
They (the technicians) usually don't know (much) about codecs or how the 
channels techs work exactly...


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