search for: codec_prefs_incoming_off

...ple sip call and see if I'm missing something or asterisk actually doesn't do what it's supposed to do? Here's the codec negotiation config: CLI> pjsip show endpoint A  ...  codec_prefs_incoming_answer        : prefer:pending, operation:intersect, keep:all, transcode:allow  codec_prefs_incoming_offer         : prefer:pending, operation:intersect, keep:all, transcode:allow  codec_prefs_outgoing_answer        : prefer:pending, operation:intersect, keep:all, transcode:allow  codec_prefs_outgoing_offer         : prefer:pending, operation:union, keep:all, transcode:allow All endpoints have t...

Hello, Anyone? I have hard time to believe this is not possible with chan_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions 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

...re and unfortunately wasn't able to get it working as expected. I tried several configurations - but none has been working - the result has always been the same. Use case: Alice calls Bob - sends INVITE G722 / alaw / ulaw Configured in Asterisk for this device: G722 / alaw / ulaw / gsm A: codec_prefs_incoming_offer = prefer: configured, operation: intersect, keep: all, transcode: prevent Bob: Configured in Asterisk for this device: alaw / ulaw B: codec_prefs_outgoing_offer = prefer: configured or pending, operation: intersect, keep: first or all, transcode: prevent Asterisk sends INVITE to Bob alaw / u...

...aw,g729 Alice calls into Asterisk on ext 100 and then we dial Bob I want to wait until Bod side codec is chosen to answer Alice and have each channel use the codec chose on Bob side. I see these options on this link, https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_pjsip codec_prefs_incoming_offer codec_prefs_outgoing_offer codec_prefs_incoming_answer codec_prefs_outgoing_answer but I dont see them on my pjsip.conf file. I only see these tow: incoming_call_offer_pref outgoing_call_offer_pref Do I have to use the 4 of them on each endoint Alice and Bob? Or just one side should be enough?...

The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: