Hi,
We're holding ourselves back from moving to PJSIP as we don't appear to
have figured out how to force codec preference in a dial plan. The 'PJSIP
Advanced Codec Negotiation' document
(https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation)
appears to ultimately be what we're after, but we're not comfortable
running Asterisk 18 in production just yet. Is there no way to mimic
functionality we previously had in chan_sip?
I don't appear to be able to set an inheritable variable for the subsequent
PJSIP leg of the call, to exclusively only offer the codec we negotiated for the
first leg of the call. If for example we have chan_iax2 incoming that we wish to
send out via pjsip.
With chan_sip, this works:
exten => s,n,Set(_SIP_CODEC_OUTBOUND=${CHANNEL(audioreadformat)})
With pjsip, this gives an error:
exten =>
s,n,Set(_PJSIP_MEDIA_OFFER(audio)=!all,${CHANNEL(audioreadformat)})
Error:
ERROR[26925][C-00020b9c] pbx_functions.c: Function _PJSIP_MEDIA_OFFER not
registered
I'd image things haven't changed since 2018 where this appears to have
been discussed in the following thread:
Re: Pjsip migration - SIP_CODEC and SIP_CODEC_OUTBOUND
URL:
https://community.asterisk.org/t/pjsip-migration-sip-codec-and-sip-codec-outbound/73342/7
No way besides learning whatever code the AGI's written in, temporarily
passing it up as a temporary variable and then calling a pre-dial handler?
I presume incoming shouldn't have this problem as the channel would be
pjsip, right? With chan_sip we simply set SIP_CODEC as one of the first inbound
context dial plan rules, before the channel is answered:
exten => s,n,Set(SIP_CODEC=alaw)
I presume the following would work with pjsip:
exten => s,n,Set(PJSIP_MEDIA_OFFER(audio)=!all,alaw)
PS: I can't find the reference again but recall a recommendation to call
Progress() due to nuances with some systems, is this still relevant?
Regards
David Herselman
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