[asterisk-users] PJSIP does not qualify contacts after starting Asterisk

[Francisco Valentin Vinagrero]

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 escribió:
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 escribió:
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.









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