Francisco Valentin Vinagrero
2016-Jun-13 12:16 UTC
[asterisk-users] PJSIP does not qualify contacts after starting Asterisk
Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Annus Fictus Sent: 13 June 2016 14:11 To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] PJSIP does not qualify contacts after starting Asterisk Hello Francisco, you have to use: extensions => odbc,asterisk only if you want use dialplan in Realtime can you share your sorcery.conf file? Regards El 13/06/2016 a las 10:21, Francisco Valentin Vinagrero escribi?: Hi all, (sending this again from the correct address) I'm running Asterisk 13.8.0 (I need to check if that happens with 13.9.1 too when I have the time to build it) with PJSIP realtime config. I've defined several aors in the table ps_aors, like this (real url replaced by myurl): *CLI> pjsip show aor pbx-node-1 Aor: <Aor..............................................> <MaxContact> Contact: <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)..> ======================================================================================== Aor: pbx-node-1 0 Contact: pbx-node-1/sip:myurl:5060 771bf6a7d4 Created 0.000 ParameterName : ParameterValue ================================================== authenticate_qualify : false contact : sip:myurl:5060 default_expiration : 3600 mailboxes : max_contacts : 0 maximum_expiration : 7200 minimum_expiration : 60 outbound_proxy : sip:myurl:5060 qualify_frequency : 30 qualify_timeout : 3.000000 remove_existing : false support_path : false So I think that those aors should be qualified automatically when I run Asterisk, but if I do "pjsip show contacts", I get that it was just Created but not qualified: *CLI> pjsip show contacts Contact: <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..> ======================================================================================== Contact: pbx-node-1/sip:myurl:5060 771bf6a7d4 Created 0.000 And not a single OPTIONS message if I take a trace... If I want Asterisk to start sending OPTIONS, I need to do pjsip reload and after that, they are qualified and their status changes dynamically: *CLI> pjsip show contacts Contact: <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..> ======================================================================================== Contact: pbx-node-1/sip:myurl.ch:5060 771bf6a7d4 Avail 8.833 The extconfig.conf file looks like this: [settings] ps_endpoints => odbc,asterisk ps_auths => odbc,asterisk ps_aors => odbc,asterisk ps_domain_aliases => odbc,asterisk ps_endpoint_id_ips => odbc,asterisk ps_contacts => odbc,asterisk extensions => odbc,asterisk Any idea why I need to reload PJSIP if I want the aors to be qualified? Cheers, Francisco. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160613/577b00d8/attachment.html>
Annus Fictus
2016-Jun-13 12:33 UTC
[asterisk-users] PJSIP does not qualify contacts after starting Asterisk
Hello, in which moment Asterisk leave to qualify the realtime endpoint? When you restart Asterisk? On my asterisk 13.9.1, qualify on realtime endpoints works correctly. My sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints endpoint=config,pjsip.conf,criteria=type=endpoint auth=realtime,ps_auths auth=config,pjsip.conf,criteria=type=auth aor=realtime,ps_aors aor=config,pjsip.conf,criteria=type=aor domain_alias=realtime,ps_domain_aliases domain_alias=config,pjsip.conf,criteria=type=domain_alias contact=realtime,ps_contacts contact=config,pjsip.conf,criteria=type=contact [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips identify=config,pjsip.conf,criteria=type=identify Regards El 13/06/2016 a las 14:16, Francisco Valentin Vinagrero escribi?:> > Hi, > > Yes, we?re implementing the dialplan in realtime too. > > Here the contents of sorcery.conf: > > [res_pjsip] > > endpoint=realtime,ps_endpoints > > aor=realtime,ps_aors > > contact=realtime,ps_contacts > > [res_pjsip_endpoint_identifier_ip] > > identify=realtime,ps_endpoint_id_ips > > Cheers, Francisco. > > *From:*asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Annus > Fictus > *Sent:* 13 June 2016 14:11 > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > *Subject:* Re: [asterisk-users] PJSIP does not qualify contacts after > starting Asterisk > > Hello Francisco, > > you have to use: > > extensions => odbc,asterisk > > only if you want use dialplan in Realtime > > can you share your sorcery.conf file? > > Regards > > El 13/06/2016 a las 10:21, Francisco Valentin Vinagrero escribi?: > > Hi all, > > (sending this again from the correct address) > > I?m running Asterisk 13.8.0 (I need to check if that happens with > 13.9.1 too when I have the time to build it) with PJSIP realtime > config. > > I?ve defined several aors in the table ps_aors, like this (real > url replaced by myurl): > > *CLI> pjsip show aor pbx-node-1 > > Aor: <Aor..............................................> > <MaxContact> > > Contact: <Aor/ContactUri............................> > <Hash....> <Status> <RTT(ms)..> > > ========================================================================================> > > Aor: pbx-node-1 0 > > Contact: pbx-node-1/sip:myurl:5060 771bf6a7d4 Created > 0.000 > > ParameterName : ParameterValue > > ==================================================> > authenticate_qualify : false > > contact : sip:myurl:5060 > > default_expiration : 3600 > > mailboxes : > > max_contacts : 0 > > maximum_expiration : 7200 > > minimum_expiration : 60 > > outbound_proxy : sip:myurl:5060 > > qualify_frequency : 30 > > qualify_timeout : 3.000000 > > remove_existing : false > > support_path : false > > So I think that those aors should be qualified automatically when > I run Asterisk, but if I do ?/pjsip show contacts?/, I get that it > was just Created but not qualified: > > *CLI> pjsip show contacts > > Contact: <Aor/ContactUri..............................> > <Hash....> <Status> <RTT(ms)..> > > ========================================================================================> > Contact: pbx-node-1/sip:myurl:5060 771bf6a7d4 Created 0.000 > > And not a single OPTIONS message if I take a trace? > > If I want Asterisk to start sending OPTIONS, I need to do pjsip > reload and after that, they are qualified and their status changes > dynamically: > > *CLI> pjsip show contacts > > Contact: <Aor/ContactUri..............................> > <Hash....> <Status> <RTT(ms)..> > > ========================================================================================> > Contact: pbx-node-1/sip:myurl.ch:5060 771bf6a7d4 Avail > 8.833 > > The extconfig.conf file looks like this: > > [settings] > > ps_endpoints => odbc,asterisk > > ps_auths => odbc,asterisk > > ps_aors => odbc,asterisk > > ps_domain_aliases => odbc,asterisk > > ps_endpoint_id_ips => odbc,asterisk > > ps_contacts => odbc,asterisk > > extensions => odbc,asterisk > > Any idea why I need to reload PJSIP if I want the aors to be > qualified? > > Cheers, Francisco. > > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160613/2e164df5/attachment-0001.html>
Francisco Valentin Vinagrero
2016-Jun-13 15:42 UTC
[asterisk-users] PJSIP does not qualify contacts after starting Asterisk
Hi, So basically you're doubling all the lines with a failover to the pjsip.conf file. What do you have in that file? For me it didn't work. Whenever I add or update a contact in the ps_aors table, I get that the contacts are created but not qualified. Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Annus Fictus Sent: 13 June 2016 14:34 To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] PJSIP does not qualify contacts after starting Asterisk Hello, in which moment Asterisk leave to qualify the realtime endpoint? When you restart Asterisk? On my asterisk 13.9.1, qualify on realtime endpoints works correctly. My sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints endpoint=config,pjsip.conf,criteria=type=endpoint auth=realtime,ps_auths auth=config,pjsip.conf,criteria=type=auth aor=realtime,ps_aors aor=config,pjsip.conf,criteria=type=aor domain_alias=realtime,ps_domain_aliases domain_alias=config,pjsip.conf,criteria=type=domain_alias contact=realtime,ps_contacts contact=config,pjsip.conf,criteria=type=contact [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips identify=config,pjsip.conf,criteria=type=identify Regards El 13/06/2016 a las 14:16, Francisco Valentin Vinagrero escribi?: Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Annus Fictus Sent: 13 June 2016 14:11 To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com><mailto:asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] PJSIP does not qualify contacts after starting Asterisk Hello Francisco, you have to use: extensions => odbc,asterisk only if you want use dialplan in Realtime can you share your sorcery.conf file? Regards El 13/06/2016 a las 10:21, Francisco Valentin Vinagrero escribi?: Hi all, (sending this again from the correct address) I'm running Asterisk 13.8.0 (I need to check if that happens with 13.9.1 too when I have the time to build it) with PJSIP realtime config. I've defined several aors in the table ps_aors, like this (real url replaced by myurl): *CLI> pjsip show aor pbx-node-1 Aor: <Aor..............................................> <MaxContact> Contact: <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)..> ======================================================================================== Aor: pbx-node-1 0 Contact: pbx-node-1/sip:myurl:5060 771bf6a7d4 Created 0.000 ParameterName : ParameterValue ================================================== authenticate_qualify : false contact : sip:myurl:5060 default_expiration : 3600 mailboxes : max_contacts : 0 maximum_expiration : 7200 minimum_expiration : 60 outbound_proxy : sip:myurl:5060 qualify_frequency : 30 qualify_timeout : 3.000000 remove_existing : false support_path : false So I think that those aors should be qualified automatically when I run Asterisk, but if I do "pjsip show contacts", I get that it was just Created but not qualified: *CLI> pjsip show contacts Contact: <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..> ======================================================================================== Contact: pbx-node-1/sip:myurl:5060 771bf6a7d4 Created 0.000 And not a single OPTIONS message if I take a trace... If I want Asterisk to start sending OPTIONS, I need to do pjsip reload and after that, they are qualified and their status changes dynamically: *CLI> pjsip show contacts Contact: <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..> ======================================================================================== Contact: pbx-node-1/sip:myurl.ch:5060 771bf6a7d4 Avail 8.833 The extconfig.conf file looks like this: [settings] ps_endpoints => odbc,asterisk ps_auths => odbc,asterisk ps_aors => odbc,asterisk ps_domain_aliases => odbc,asterisk ps_endpoint_id_ips => odbc,asterisk ps_contacts => odbc,asterisk extensions => odbc,asterisk Any idea why I need to reload PJSIP if I want the aors to be qualified? Cheers, Francisco. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160613/02be7380/attachment.html>