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


 
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francisco.valentin.vin...
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PostPosted: Mon Jun 13, 2016 3:22 am    Post subject: [asterisk-users] PJSIP does not qualify contacts after start Reply with quote

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 : [url=sip:myurl:5060]sip:myurl:5060[/url]
default_expiration : 3600
mailboxes :
max_contacts : 0
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy : [url=sip:myurl:5060]sip:myurl:5060[/url]
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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annusfictus at gmail.com
Guest





PostPosted: Mon Jun 13, 2016 7:11 am    Post subject: [asterisk-users] PJSIP does not qualify contacts after start Reply with quote

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:

Quote:
<![endif]--> <![endif]-->
Hi all,
(sending this again from the correct address)

Im 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.

Ive 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/[url=sip:myurl:5060]sip:myurl:5060[/url] 771bf6a7d4 Created 0.000


ParameterName : ParameterValue
===================================================
authenticate_qualify : false
contact : [url=sip:myurl:5060]sip:myurl:5060[/url]
default_expiration : 3600
mailboxes :
max_contacts : 0
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy : [url=sip:myurl:5060]sip:myurl:5060[/url]
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/[url=sip:myurl:5060]sip:myurl:5060[/url] 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/[url=sip:myurl.ch:5060]sip:myurl.ch:5060[/url] 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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francisco.valentin.vin...
Guest





PostPosted: Mon Jun 13, 2016 7:18 am    Post subject: [asterisk-users] PJSIP does not qualify contacts after start Reply with quote

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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Annus Fictus
Sent: 13 June 2016 14:11
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@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:
Quote:

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/[url=sip:myurl:5060]sip:myurl:5060[/url] 771bf6a7d4 Created 0.000


ParameterName : ParameterValue
===================================================
authenticate_qualify : false
contact : [url=sip:myurl:5060]sip:myurl:5060[/url]
default_expiration : 3600
mailboxes :
max_contacts : 0
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy : [url=sip:myurl:5060]sip:myurl:5060[/url]
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/[url=sip:myurl:5060]sip:myurl:5060[/url] 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/[url=sip:myurl.ch:5060]sip:myurl.ch:5060[/url] 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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annusfictus at gmail.com
Guest





PostPosted: Mon Jun 13, 2016 7:34 am    Post subject: [asterisk-users] PJSIP does not qualify contacts after start Reply with quote

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:

Quote:
<![endif]--> <![endif]-->
Hi,

Yes, were 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@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Annus Fictus
Sent: 13 June 2016 14:11
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> (asterisk-users@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:
Quote:

Hi all,
(sending this again from the correct address)

Im 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.

Ive 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/[url=sip:myurl:5060]sip:myurl:5060[/url] 771bf6a7d4 Created 0.000


ParameterName : ParameterValue
===================================================
authenticate_qualify : false
contact : [url=sip:myurl:5060]sip:myurl:5060[/url]
default_expiration : 3600
mailboxes :
max_contacts : 0
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy : [url=sip:myurl:5060]sip:myurl:5060[/url]
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/[url=sip:myurl:5060]sip:myurl:5060[/url] 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/[url=sip:myurl.ch:5060]sip:myurl.ch:5060[/url] 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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francisco.valentin.vin...
Guest





PostPosted: Mon Jun 13, 2016 10:43 am    Post subject: [asterisk-users] PJSIP does not qualify contacts after start Reply with quote

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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Annus Fictus
Sent: 13 June 2016 14:34
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@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:
Quote:

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@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Annus Fictus
Sent: 13 June 2016 14:11
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> (asterisk-users@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:
Quote:

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/[url=sip:myurl:5060]sip:myurl:5060[/url] 771bf6a7d4 Created 0.000


ParameterName : ParameterValue
===================================================
authenticate_qualify : false
contact : [url=sip:myurl:5060]sip:myurl:5060[/url]
default_expiration : 3600
mailboxes :
max_contacts : 0
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy : [url=sip:myurl:5060]sip:myurl:5060[/url]
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/[url=sip:myurl:5060]sip:myurl:5060[/url] 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/[url=sip:myurl.ch:5060]sip:myurl.ch:5060[/url] 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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gjoseph at digium.com
Guest





PostPosted: Tue Jun 14, 2016 1:36 pm    Post subject: [asterisk-users] PJSIP does not qualify contacts after start Reply with quote

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@cern.ch (francisco.valentin.vinagrero@cern.ch)> wrote:
Quote:

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@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Annus Fictus
Sent: 13 June 2016 14:34
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com (asterisk-users@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:
Quote:

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@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Annus Fictus
Sent: 13 June 2016 14:11
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> (asterisk-users@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:
Quote:

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 JosephDigium, 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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