search for: pjsip_media_off

hi: when using chan_sip, I can use set SIP_CODEC in dialplan to change the codec of endpoint. this method didn't work with pjsip in asterisk 12/13. I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems can set codec, but the document didn't offer any example. i try to use something like PJSIP_MEDIA_OFFER(alaw) but didn't work. can someone give an example for the function? thanks for the help. Regards, tbskyd

Hello, I finally got to look at chan_sip to chan_pjsip migration again. This time I’m having problems with influencing codec selection on originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only works on 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) cha...

...e 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_pjsip migration again. This > time I’m having problems with influencing codec selection on > originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only > works on 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...

On 03.10.19 15:08, Administrator TOOTAI wrote: > Before calling the gatreway add > > same = n,set(SIP_CODEC=alaw) > > [...] > Hey there, that doesn't work as it seems to be implemented for chan_sip only; I'm using chan_pjsip; sorry if I didn't explain myself properly. Anyway, in my case that would not really be an acceptable solution anyway, because I need the

...Michael > > On 6/30/23 09:30, Michael Ulitskiy wrote: >> >> Hello, >> >> I finally got to look at chan_sip to chan_pjsip migration again. This >> time I’m having problems with influencing codec selection on >> originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only >> works on 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_MED...

Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data

...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...

...) * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-29029 - Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe (Reported by Karsten Wemheuer) * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion...

...) * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-29029 - Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe (Reported by Karsten Wemheuer) * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion...

...events (Reported by Ove Aursand) * ASTERISK-29043 - app_queue: Leave empty sometimes not recorded as abandoned (Reported by Kfir Itzhak) * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion...

...release of Asterisk 13.35.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Reported by Jaco Kroon) * ASTERISK-28957 - chan_...

...release of Asterisk 13.35.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Reported by Jaco Kroon) * ASTERISK-28957 - chan_...

...release of Asterisk 16.12.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Reported by Jaco Kroon) * ASTERISK-28886 - chan_...

...release of Asterisk 16.12.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Reported by Jaco Kroon) * ASTERISK-28886 - chan_...

...release of Asterisk 17.6.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Reported by Jaco Kroon) * ASTERISK-28886 - chan_...

...release of Asterisk 17.6.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Reported by Jaco Kroon) * ASTERISK-28886 - chan_...

...ections on shutdown (Reported by Corey Farrell) * ASTERISK-24369 - res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations (Reported by Matt Jordan) * ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK (Reported by Matt Jordan) * ASTERISK-24370 - res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd (Reported by Matt Jordan) * ASTERISK-242...

...ections on shutdown (Reported by Corey Farrell) * ASTERISK-24369 - res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations (Reported by Matt Jordan) * ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK (Reported by Matt Jordan) * ASTERISK-24370 - res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd (Reported by Matt Jordan) * ASTERISK-242...

...events (Reported by Ove Aursand) * ASTERISK-29043 - app_queue: Leave empty sometimes not recorded as abandoned (Reported by Kfir Itzhak) * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion...

...surrogate channel appears in list and call count is inflated. (Reported by Mark Michelson) * ASTERISK-24370 - res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd (Reported by Matt Jordan) * ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK (Reported by Matt Jordan) * ASTERISK-24369 - res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locat...