Test
--Mensaje original--
De: tracinet
Remitente:asterisk-users-boun...@lists.digium.com
Para:Asterisk Users Mailing List - Non-Commercial Discussion
Responder a:Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On 25/03/2009 11:08 a.m., darwin.sol...@gmail.com wrote:
Test
failed :)
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
SIP
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
arob...@pharmacentra.com wrote:
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
] On Behalf Of tracinet
Sent: Friday, March 06, 2009 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
SIP
On Wed, Mar
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins arob...@pharmacentra.comwrote:
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade.
On the server that SENDS the call, I have the following in SIP.CONF:
[192.168.1.2_OB]
no
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tracinet
Sent: Friday, March 06, 2009 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins arob...@pharmacentra.com
wrote:
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues
On 6 Mar 2009, at 19:29, tracinet wrote:
That stinks... We are migrating to SIP from IAX2 at the moment and
running into the same exact problem. No way to control the
destination context unless you use the fromuser. Of course that
is rendering Caller ID useless as you pointed out.
[mailto:asterisk-users-boun...@lists.digium.com
] On Behalf Of tracinet
Sent: Friday, March 06, 2009 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
SIP
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
Basically, Server 1 is the main customer PBX where we have multiple
customers running (each on their own virtual PBX separated by their
contexts). Each customer has their own accountcode that we use to track
calls for billing purposes, etc. The customer uses a SIP phone to register
to Server 1
2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
SIP
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
arob...@pharmacentra.com wrote:
I am switching from IAX2 to SIP for my inter-Asterisk
wrote:
no
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com
] On Behalf Of tracinet
Sent: Friday, March 06, 2009 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Asterisk-Users] Inter
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade.
On the server that SENDS the call, I have the following in SIP.CONF:
[192.168.1.2_OB]
type=peer
fromuser=OB
host=192.168.1.2
And in EXTENSIONS.CONF
exten =
13 matches
Mail list logo
