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francisco.valentin.vin... Guest
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Posted: Mon Jun 13, 2016 3:22 am Post subject: [asterisk-users] PJSIP does not qualify contacts after start |
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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
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Posted: Mon Jun 13, 2016 7:11 am Post subject: [asterisk-users] PJSIP does not qualify contacts after start |
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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
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Posted: Mon Jun 13, 2016 7:18 am Post subject: [asterisk-users] PJSIP does not qualify contacts after start |
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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
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Posted: Mon Jun 13, 2016 7:34 am Post subject: [asterisk-users] PJSIP does not qualify contacts after start |
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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
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Posted: Mon Jun 13, 2016 10:43 am Post subject: [asterisk-users] PJSIP does not qualify contacts after start |
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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
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Posted: Tue Jun 14, 2016 1:36 pm Post subject: [asterisk-users] PJSIP does not qualify contacts after start |
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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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