search for: sip_codec_outbound

...(called) channel and has no affect on calling channel. My experiments and function documentation (which says “Media and codec offerings to be set on an outbound SIP channel prior to dialing.”) seem to confirm it. So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s equivalent of ${SIP_CODEC_INBOUND}? Or, in other words, what are we supposed to do to influence /calling/ channel codec selection from dialplan? I’m working with asterisk 20.3.0. Thank you, Michael -------------- next part -------------- An HTML attachment was scrubbe...

...ndles per minute and it never goes down. The dialplan is actually a four liner look at the audiowritecodec select an outbound endpoint based on that The idea is to bridge calls based on the codec to avoid any transcoding, so I have two outbound codecs and I dial like this: exten => _X.,1,Set(_SIP_CODEC_OUTBOUND=${CHANNEL(audiowriteformat):0:4}) exten => _X.,n,Goto(${SIP_CODEC_OUTBOUND}) exten => _X.,n(ulaw),Dial(PJSIP/alawoutbound/sip:${EXTEN}@X.X.X.X) exten => _X.,n(g729),Dial(PJSIP/g729outbound/sip:${EXTEN}@X.X.X.X) As you can see, "Houston, we have a problem"

...no affect on calling > channel. My experiments and function documentation (which says “Media > and codec offerings to be set on an outbound SIP channel prior to > dialing.”) seem to confirm it. > So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s > equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s > equivalent of ${SIP_CODEC_INBOUND}? Or, in other words, what are we > supposed to do to influence /calling/ channel codec selection from > dialplan? > I’m working with asterisk 20.3.0. > > Thank you, > Michael > > > -------------- next p...

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

...as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP codec change after dahdi answered the channel. so everything is broken. the call log like below: [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c: -- Executing [s at macro-dialout-trunk-predial-hook:2] Set("SIP/222-00000004", "SIP_CODEC=alaw&q...

...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...

...ction of > sip.conf. Depending on your Asterisk version only the order in [general] > would be respected, but not the order in the individual sip peer/user > definition > > * look at the variable SIP_CODEC for the inbound (first) call leg, and in > Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND > > * many Asterisk operators have applied the third party "codec negotiation > patch" > > Philipp > > > > > ------------------------------ > > Message: 15 > Date: Tue, 3 Aug 2010 07:26:41 -0400 > From: C F <shmaltz at gmail.com> > Subjec...

...>> channel. My experiments and function documentation (which says “Media >> and codec offerings to be set on an outbound SIP channel prior to >> dialing.”) seem to confirm it. >> So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s >> equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s >> equivalent of ${SIP_CODEC_INBOUND}? Or, in other words, what are we >> supposed to do to influence /calling/ channel codec selection from >> dialplan? >> I’m working with asterisk 20.3.0. >> >> Thank you, >> Michael >>...