search for: pjsip_resolv

Hello. I have two records in dialplan: exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org) Calling testA works fine while testB fails with "CONGESTION". Adding debug for console shows that pjsip_resolver.c does `New queries added, performing parallel resolution again` for linphone after discovering SRV records, and does not for iptel. It looks like a bug. All modules are loaded manually, with autoload=no, res_resolver_unbound is not loaded. Asterisk 16.2.1~dfsg-1+deb10u2 debian stable. I hav...

I see that pjsip_resolver.c tries unsuccessfuly to resolve the hostname each 10 seconds: [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Wrapper created [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Set timer to 2000 msec [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_resolver.c: Performing SIP DNS re...

On Thu, Aug 27, 2020 at 8:58 AM Leonid Fainshtein < leonid.fainshtein at xorcom.com> wrote: > I see that pjsip_resolver.c tries unsuccessfuly to resolve the > hostname each 10 seconds: > > [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Wrapper created > [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Set timer to > 2000 msec > [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_reso...

...terisk 16.2 installed from Debian 10 repository. Best regards, Leonid Fainshtein On Thu, Aug 27, 2020 at 3:01 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Thu, Aug 27, 2020 at 8:58 AM Leonid Fainshtein < > leonid.fainshtein at xorcom.com> wrote: > >> I see that pjsip_resolver.c tries unsuccessfuly to resolve the >> hostname each 10 seconds: >> >> [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Wrapper created >> [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Set timer to >> 2000 msec >> [Aug 27 07:51:36] DEBUG[5...

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

...n [0, 4294967295] gives [32](0) [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip_outbound_registration.c:699 handle_client_registration: Outbound REGISTER attempt 2 to 'sip:orange-obs.fr' with client 'sip:+3313445XXXX at orange-obs.fr' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:475 sip_resolve: Performing SIP DNS resolution of target 'orange-obs.fr' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:502 sip_resolve: Transport type for target 'orange-obs.fr' is 'UDP transport' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_r...

...YJeSn0ELEYAVnkOVCUwou7pxO len 40 [Jun 8 12:28:09] DEBUG[4180] res_pjsip_sdp_rtp.c: Stream msid: 0x7f0578077610 audio 23eb03ca-f0ee-406a-b7cd-5fb19fc33fa2 ddca7927-ff8d-45ab-a61f-9474f8b7a9df [Jun 8 12:28:09] DEBUG[4180] res_pjsip_session.c: Method is INVITE [Jun 8 12:28:09] DEBUG[4180] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '10.215.144.48' [Jun 8 12:28:09] DEBUG[4180] res_pjsip/pjsip_resolver.c: Transport type for target '10.215.144.48' is 'ws' [Jun 8 12:28:09] DEBUG[4180] res_pjsip/pjsip_resolver.c: Target '10.215.144.48' is an IP address,...

...dialplan: > > exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org) > exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org) > > Calling testA works fine while testB fails with "CONGESTION". > > Adding debug for console shows that pjsip_resolver.c does > `New queries added, performing parallel resolution again` > for linphone after discovering SRV records, and does not for iptel. > > It looks like a bug. > > All modules are loaded manually, with autoload=no, res_resolver_unbound > is not loaded. > > Asterisk 16...

...) at ../src/pjsip/sip_util.c:1276 #11 0x00007f777bc71b3a in stateless_send_resolver_callback (status=<optimized out>, token=0x7f773c635950, addr=<optimized out>) at ../src/pjsip/sip_util.c:1377 #12 0x00007f77380c3398 in sip_resolve_invoke_user_callback (data=0x7f773c4563c8) at res_pjsip/pjsip_resolver.c:206 #13 0x000055e7c0a6f0e3 in ast_taskprocessor_execute (tps=tps at entry=0x7f775416ebc0) at taskprocessor.c:1302 #14 0x000055e7c0a76d28 in execute_tasks (data=0x7f775416ebc0) at threadpool.c:1352 #15 0x000055e7c0a6f0e3 in ast_taskprocessor_execute (tps=0x55e7c2697720) at taskprocessor.c:1302 #...

...in dtls_srtp_check_pending (Reported by Dade Brandon) * ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c (Reported by Carl Fortin) * ASTERISK-25115 - Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c (Reported by John Bigelow) * ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early replaces call pickup (Reported by Walter Doekes) * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c (Reported by Walter Doekes) * ASTERISK-25219 - [patch]Source and destination...

On Wed, Aug 9, 2023 at 3:20 PM Dan Cropp <dcropp at amtelco.com> wrote: > I have a customer who just encountered a crash while running Asterisk > 18.17.1 version. > > > > I’m trying to adapt to the changes so not sure where best to look or how > to possibly report this. > > > > I started by going through >

...in dtls_srtp_check_pending (Reported by Dade Brandon) * ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c (Reported by Carl Fortin) * ASTERISK-25115 - Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c (Reported by John Bigelow) * ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early replaces call pickup (Reported by Walter Doekes) * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c (Reported by Walter Doekes) * ASTERISK-25219 - [patch]Source and destination...

...ted line updates (Reported by Joshua Colp) * ASTERISK-25087 - Asterisk segfault when using Directory application with alias option and specific mailbox configuration (Reported by Chet Stevens) * ASTERISK-25115 - Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c (Reported by John Bigelow) * ASTERISK-25096 - [patch]Segfault when registering over websockets with PJSIP (in ast_sockaddr_isnull at /include/asterisk/netsock2.h) (Reported by Josh Kitchens) * ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS (Reported by B...

...>] - Asterisk segfault when using Directory application with alias option and specific mailbox configuration (Reported by Chet Stevens) - [ASTERISK-25115 <https://issues.asterisk.org/jira/browse/ASTERISK-25115>] - Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c (Reported by John Bigelow) - [ASTERISK-25096 <https://issues.asterisk.org/jira/browse/ASTERISK-25096>] - [patch]Segfault when registering over websockets with PJSIP (in ast_sockaddr_isnull at /include/asterisk/netsock2.h) (Reported by Josh Kitchens) - [ASTERISK-24963 <ht...