[asterisk-users] PJSIP does not qualify contacts after starting Asterisk
George Joseph
gjoseph at digium.com
Tue Jun 14 13:36:08 CDT 2016
pjsip realtime, especially related to contacts, has been worked on a bunch
over the last few releases. Honestly I don't remember what went into which
release but do try 13.9.1 and let us know.
You can also try using the following statements in sorcery.conf and see if
it changes anything. Don't use them in production because
full_backend_cache and allow_unqualified_fetch have performance
implications.
aor/cache=memory_cache,maximum_objects=150,expire_on_reload=yes,object_lifetime_maximum=3600,full_backend_cache=yes
aor=realtime,ps_aors,allow_unqualified_fetch=warn
On Mon, Jun 13, 2016 at 9:42 AM, Francisco Valentin Vinagrero <
francisco.valentin.vinagrero at cern.ch> wrote:
> 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
> <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> <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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--
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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