search for: codec_prefs_incoming_answer

...ctually follow the described logic which is likely a bug. Can you please follow with me through a simple 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_outgoi...

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

...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 / ulaw Asterisk receives OK from Bob alaw B: codec_prefs_incoming_answer = prefer: configured or pending, operation: intersect, keep: first or all, transcode: prevent Asterisk sends OK to Alice G722 / alaw / ulaw A: codec_prefs_outgoing_answer = prefer: pending, operation: intersect, keep: first or all, transcode: prevent => I would have expected alaw to be sent...

...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? -------------- next part -------------- An HTML attachmen...

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: