search for: pjsip_options

...0x7f75282eb150: Cancelling timer [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Timer cancelled [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Callbacks executed [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: wrapper destroyed [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_options.c: Contact g145/sip: rpi6.in.xorcom.com status didn't change: Unreachable, RTT: 0.000 msec [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_options.c: AOR 'g145' now has 0 available contacts But a DNS server is available at this time: # host rpi6.in.xorcom.com rpi6.in.xorcom.com has addres...

...int [1000 at sip1.domain.com] type=endpoint I can register the two 1000 endpoints using different domain but on the Asterisk console: ERROR[1748]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to create outbound OPTIONS request to endpoint 1000 at sip.domain.com ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact: Unable to create request to qualify contact sip:1000 at 95.250.29.3:53570;rinstance=d90827763e4353c0 in the aor section I'm using: qualify_frequency=30 Any hint? Regards

...mer > [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Timer cancelled > [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Callbacks > executed > [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: wrapper > destroyed > [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_options.c: Contact g145/sip: > rpi6.in.xorcom.com status didn't change: Unreachable, RTT: 0.000 msec > [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_options.c: AOR 'g145' now has > 0 available contacts > > But a DNS server is available at this time: > # host rpi6.in.xorcom.com...

...different domain but > on the Asterisk console: > > ERROR[1748]: res_pjsip.c:2946 create_out_of_dialog_request: Unable > to create outbound OPTIONS request to endpoint 1000 at sip.domain.com > <mailto:1000 at sip.domain.com> > > ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact: Unable > to create request to qualify contact sip:1000 at 95.250.29.3:53570 > <http://sip:1000 at 95.250.29.3:53570>;rinstance=d90827763e4353c0 > > in the aor section I'm using: > > qualify_frequency=30 > > > If you set qua...

...51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Timer cancelled >> [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Callbacks >> executed >> [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: wrapper >> destroyed >> [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_options.c: Contact g145/sip: >> rpi6.in.xorcom.com status didn't change: Unreachable, RTT: 0.000 msec >> [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_options.c: AOR 'g145' now >> has 0 available contacts >> >> But a DNS server is available at this time: >> #...

Hi, I have Asterisk 16.x with a trunk configured with a hostname in PJSIP AOR. The registration is not required for this trunk. I paid attention that Asterisk performs DNS resolving of the hostname that is configured in the AOR 'contact' parameter only upon the Asterisk start only. Thus, if Asterisk is started when the DNS server is unreachable due to the Internet connection failure then

...orted by Jaco Kroon) * ASTERISK-26570 - Macro allows an infinite loop of dialplan inclusion resulting in a crash (Reported by Tzafrir Cohen) * ASTERISK-27801 - Asterisk got stuck while enabling "ari set debug all on" (Reported by shaurya jain) * ASTERISK-26806 - pjsip_options: rework to make more efficient (Reported by Kevin Harwell) * ASTERISK-27814 - translate: interpolated frames are not passed through (Reported by Kevin Harwell) * ASTERISK-27812 - When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk...

...#39; (Reported by Chico Manobela) * ASTERISK-20841 - fromdomain not honored on outbound INVITE request (Reported by Kelly Goedert) * ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f) at astobj2.c:120 (Reported by Jamuel Starkey) * ASTERISK-23254 - Bad ao2_find() usage in pjsip_options.c (Reported by Richard Mudgett) * ASTERISK-23509 - [patch]SayNumber for Polish language tries to play empty files for numbers divisible by 100 (Reported by zvision) * ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find (Reported by JoshE) * ASTERISK-23391 - Audit...

...#39; (Reported by Chico Manobela) * ASTERISK-20841 - fromdomain not honored on outbound INVITE request (Reported by Kelly Goedert) * ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f) at astobj2.c:120 (Reported by Jamuel Starkey) * ASTERISK-23254 - Bad ao2_find() usage in pjsip_options.c (Reported by Richard Mudgett) * ASTERISK-23509 - [patch]SayNumber for Polish language tries to play empty files for numbers divisible by 100 (Reported by zvision) * ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find (Reported by JoshE) * ASTERISK-23391 - Audit...

...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-24224 - When using Bridge() dialplan application, surrogate channel appears in list and call count is inflated. (Reported by Mark Michelson) * ASTERISK-24354...

...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-24224 - When using Bridge() dialplan application, surrogate channel appears in list and call count is inflated. (Reported by Mark Michelson) * ASTERISK-24354...

...ASTERISK-27795 - chan_sip: one way / no audio with srtp (Reported by Florian Kaiser) * ASTERISK-27800 - One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP (Reported by Artur Pires) * ASTERISK-26806 - pjsip_options: rework to make more efficient (Reported by Kevin Harwell) * ASTERISK-27814 - translate: interpolated frames are not passed through (Reported by Kevin Harwell) * ASTERISK-27812 - When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk...

...* ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27465 - CLI Completion Not Working (Reported by Ross Beer) * ASTERISK-27460 - CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... (Reported by Richard Mudgett...

...* ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27465 - CLI Completion Not Working (Reported by Ross Beer) * ASTERISK-27460 - CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... (Reported by Richard Mudgett...

...* ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27465 - CLI Completion Not Working (Reported by Ross Beer) * ASTERISK-27460 - CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... (Reported by Richard Mudgett...

...* ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27465 - CLI Completion Not Working (Reported by Ross Beer) * ASTERISK-27460 - CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... (Reported by Richard Mudgett...

...ASTERISK-27795 - chan_sip: one way / no audio with srtp (Reported by Florian Kaiser) * ASTERISK-27800 - One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP (Reported by Artur Pires) * ASTERISK-26806 - pjsip_options: rework to make more efficient (Reported by Kevin Harwell) * ASTERISK-27814 - translate: interpolated frames are not passed through (Reported by Kevin Harwell) * ASTERISK-27812 - When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk...

...get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27465 - CLI Completion Not Working (Reported by Ross Beer) * ASTERISK-27460 - CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... (Reported by Richard Mudgett...

...ASTERISK-27795 - chan_sip: one way / no audio with srtp (Reported by Florian Kaiser) * ASTERISK-27800 - One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP (Reported by Artur Pires) * ASTERISK-26806 - pjsip_options: rework to make more efficient (Reported by Kevin Harwell) * ASTERISK-27814 - translate: interpolated frames are not passed through (Reported by Kevin Harwell) * ASTERISK-27812 - When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk...

...y Farrell) * ASTERISK-27382 - crash after an invalid rtcp packet from GT48 FXS gateway (Reported by Tzafrir Cohen) * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should (Reported by Vitezslav Novy) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27460 - CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... (Reported by Richard Mudgett) * ASTERISK-27421 - RTP source learning not working with devices that have...