similar to: res_rtp_asterisk.so problem with minimal (ish) chan-sip based Asterisk

Hi! Installing new asterisk server and decided to use chan_pjsip. While module reload I get: y 12 15:33:04] ERROR[21137]: config_options.c:715 aco_process_var: Could not find option suitable for category '3567' named 'inband_progress' at line 867 of [May 12 15:33:04] ERROR[21137]: res_sorcery_config.c:317 sorcery_config_internal_load: Could not create an object of type

Hello, 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, I can't compile asterisk having error    [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct pj_ice_sess_cb’} has no member named ‘on_valid_pair’   .on_valid_pair = ast_rtp_on_valid_pair,    ^~~~~~~~~~~~~ res_rtp_asterisk.c:2674:19: warning:

Le 13/03/2020 à 13:30, Joshua C. Colp a écrit : > On Fri, Mar 13, 2020 at 9:27 AM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > Hello, > > 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of > them, > I can't compile asterisk having error > >     [CC] res_rtp_asterisk.c ->

Hello, In Asterisk 11 it is possible to set extensions on DB table (sipppers) and also in sip.conf. But in Asterisk 13 apparently this is not possible: as I tried to set in ps_endpoints and also in pjsip.conf but only the realtime endpoints are loaded. Is there a way to use realtime + static modes at the same time for the ps_endpoints lookup using PJSIP. Thanks -------------- next part

asterisk-16.13.0-rc2. Fedora 32 pjsip won't load because of undefined symbols: [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'func_pjsip_aor.so': /usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module

I have a CentOS 6.3 machine I installed Asterisk 11, worked fine... I then tried to install on Cents 5.8, seemed to go fine... Then when I placed a call I got this: ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? Did a search and found issues with ARM and this problem but did not help me, not using gtalk or anything. Just call between two polycom phones on local network.

On 2 February 2015 at 15:12, Joshua Colp <jcolp at digium.com> wrote: > Sunny wrote: > >> Hello, >> >> >> In Asterisk 11 it is possible to set extensions on DB table (sipppers) >> and also in sip.conf. >> >> But in Asterisk 13 apparently this is not possible: as I tried to set in >> ps_endpoints and also in pjsip.conf but only the

The Asterisk Development Team would like to announce the release of asterisk-18.20.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of Certified asterisk-18.9-cert7. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert7 and https://downloads.asterisk.org/pub/telephony/certified-asterisk This release resolves issues reported by the community and would have not been possible

The Asterisk Development Team would like to announce the release of asterisk-21.0.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of Certified asterisk-18.9-cert7. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert7 and https://downloads.asterisk.org/pub/telephony/certified-asterisk This release resolves issues reported by the community and would have not been possible

The Asterisk Development Team would like to announce the release of asterisk-18.20.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-21.0.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

On 2/15/16 12:50 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> Is it possible to use serveral protocols for a single transport section >> in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you >> cound use webrtc along with your phones but if I try: >> >> [transport-udp] >> type=transport >> protocol=udp,ws,wss >> bind=0.0.0.0 >

On Fri, Mar 13, 2020 at 9:27 AM Administrator <admin at tootai.net> wrote: > Hello, > > 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, > I can't compile asterisk having error > > [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o > res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct > pj_ice_sess_cb’} has no member named

Hi, I compiled the asterisk 14.0.2 to my ARMv5 NAS, however I just have enough time to test it now. But with the default config (I only edited the http.conf), it won't start, but gives the following: Sorcery registered wizard 'bucket' Sorcery registered wizard 'bucket_file' Parsing /ffp/etc/asterisk/sorcery.conf Parsing '/ffp/etc/asterisk/sorcery.conf': Found Cannot

Dear list, I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know. 1) Ports and IP addresses which PJSIP bind to I have configured one transport like that: [tr_wZCMk5MvC2ATNzAr] type = transport protocol = udp bind = 192.168.20.48 Nevertheless, PJSIP

The Asterisk Development Team has announced the first beta of Asterisk 14.0.0. This beta is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.0.0-beta1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this beta: New