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:*[email protected] [mailto:[email protected]] *On Behalf Of *Annus Fictus
*Sent:* 13 June 2016 14:11
*To:* Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> *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.






-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to