search for: sip_codec_inbound

...riments 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 scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users...

...xtensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between internal extensions naturally happen over g722 as its their preferred codec - for external calls I now set SIP_CODEC_INBOUND=ulaw to influence codec selection on calling channel and the calls set up using ulaw end-to-end Can somebody please advise how to achieve the same with chan_pjsip? Thanks, Michael On 6/30/23 09:30, Michael Ulitskiy wrote: > > Hello, > > I finally got to look at chan_sip to chan_pjs...

...her using g722. so their > codec setting (with chan_sip now) is "allow=g722,ulaw" > - I have carriers trunks that handle ulaw only (allow=ulaw) > - calls between internal extensions naturally happen over g722 as its > their preferred codec > - for external calls I now set SIP_CODEC_INBOUND=ulaw to influence codec > selection on calling channel and the calls set up using ulaw end-to-end > > Can somebody please advise how to achieve the same with chan_pjsip? > > Thanks, > > Michael >

...her using g722. so their > codec setting (with chan_sip now) is "allow=g722,ulaw" > - I have carriers trunks that handle ulaw only (allow=ulaw) > - calls between internal extensions naturally happen over g722 as its > their preferred codec > - for external calls I now set SIP_CODEC_INBOUND=ulaw to influence > codec selection on calling channel and the calls set up using ulaw > end-to-end > > Can somebody please advise how to achieve the same with chan_pjsip? > > Thanks, > > Michael > > On 6/30/23 09:30, Michael Ulitskiy wrote: >> >> Hello,...

On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> wrote: > Hello, > > After I have re-read the "PJSIP Advanced Codec negotiation" document, it > occurred to me that the desired behavior should actually happen > automatically, just due to the codec negotiation logic, but it looks like > asterisk doesn't actually follow the described

Oh, that's great. It wasn't clear from that page, at least not for me. :-( Having it clearly stated on the document would save me (and probably others) lots of time. Thanks for clarifying it. Any idea on the timeframe of implementation? *Michael Ulitskiy* Ace Innovative Networks, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-1203 https://www.aceinnovative.com On 7/6/23 12:47, Joshua

Well, I'm trying to migrate to chan_pjsip so that I don't have to do that. It's so surprising that the issue so seemingly obvious and trivial hasn't been addressed yet that I wanted to query the collective wisdom of this list to verify my observations. Thanks for github pointer. Michael On 7/5/23 16:46, asterisk at phreaknet.org wrote: > On 7/5/2023 4:19 PM, Michael

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So

...e common format. Looking into this deeper Device 100 sends INVITE to Asterisk offering g722,ulaw,alaw,g729 Asterisk sends INVITE to device 101 offering ulaw Device 101 sends 200 OK to Asterisk offering ulaw Asterisk sends 200 OK to device 100 offering g722,ulaw I can prevent transcoding by adding SIP_CODEC_INBOUND=ulaw to the dialplan for extension 101. This causes Asterisk to send 200 OK to device 100 offering ulaw. Am I missing why Asterisk wouldn't just offer the highest priority codec they have in common to prevent transcoding? Ryan -------------- next part -------------- An HTML attachment was scru...

Hello I use asterisk realtime, and I can set the order of codec preference on my realtime allow column. If I could disable transcoding, then I can always ensure a passthrough of the common codec from origin to destination without transcoding (expensive on CPU) - and more or less, force the codec to use by setting the codec preference So, can I disable transcoding?

...) always return 0 (Reported by Aaron An) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-26655 - [patch]pjsip:...

...26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-26655 - [patch]pjsip:...

...0 (Reported by Aaron An) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-266...

...0 (Reported by Aaron An) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-266...

...0 (Reported by Aaron An) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-266...

...chard Mudgett) - [ASTERISK-26670 <https://issues.asterisk.org/jira/browse/ASTERISK-26670>] - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) - [ASTERISK-26691 <https://issues.asterisk.org/jira/browse/ASTERISK-26691>] - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) - [ASTERISK-26673 <https://issues.asterisk.org/jira/browse/ASTERISK-26673>] - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua C. Colp) - [ASTERISK-26684 <https://issues.asterisk.org/jira/browse/ASTER...