I have a quick call routing question that I'm sure is simple, but for the
life of me I can't figure out.
For example, someone dials 94162384000 asterisk trys our first call route
(our sip trunk) as per the extension rule below:
exten => _9NXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@10.10.100.40)
However, this call fails because 94162384000 is one of our phone lines and
our SIP gateway detects a loop and returns a SIP 503 message. Is there a
way to have asterisk stip the '9' and try it as a local extension call
as if
the user didn't dial 9? I try this (see below) and it failed:
exten => _9NXXXXXXXXX,2,Dial(${EXTEN:1})
Thanks in advance, I'm sure it's a simple problem and I'm just
missing
something...
Regards,
M. Gamble
Hi Matt! :)
You can use the "Local" channel driver:
exten => _9NXXNXXXXXX,1,Dial(Local/${EXTEN:1}@${CONTEXT})
Where ${CONTEXT} is set to the local context you want to use.
-wade
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of Matthew M. Gamble
> Sent: Thursday, August 07, 2003 8:36 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Call routing question
>
> I have a quick call routing question that I'm sure is simple, but for
the
> life of me I can't figure out.
>
> For example, someone dials 94162384000 asterisk trys our first call route
> (our sip trunk) as per the extension rule below:
>
> exten => _9NXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@10.10.100.40)
>
> However, this call fails because 94162384000 is one of our phone lines and
> our SIP gateway detects a loop and returns a SIP 503 message. Is there a
> way to have asterisk stip the '9' and try it as a local extension
call as
> if
> the user didn't dial 9? I try this (see below) and it failed:
>
> exten => _9NXXXXXXXXX,2,Dial(${EXTEN:1})
>
> Thanks in advance, I'm sure it's a simple problem and I'm just
missing
> something...
>
> Regards,
>
> M. Gamble
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
Hi All, I have a question about call routing. I currently have a phone number provided by Voicepulse that connects directly to my Asterisk box and another phone number provided by Verizon that I have Remote Call Forwarded to the Voicepulse number. What I'm wondering is if the information about which number is actually dialed available for me to route the calls to different extensions? Thanks for the help and I apologize if this has already been discussed. Thanks, -Herman
Try using the special identifiers ${DNID} or ${RDNIS}. Refer to
http://www.voip-info.org/tiki-index.php?page=Asterisk%20variables for more info.
Regards
Cameron
Original message
------------------------------
Date: Tue, 08 Mar 2005 10:09:37 -0500
From: Herman Sheremetyev <herman@swebpage.com>
Subject: [Asterisk-Users] call routing question
To: asterisk-users@lists.digium.com
Message-ID: <422DC031.4010800@swebpage.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi All,
I have a question about call routing. I currently have a phone number provided
by Voicepulse that connects directly to my Asterisk box and another phone number
provided by Verizon that I have Remote Call
Forwarded to the Voicepulse number. What I'm wondering is if the
information about which number is actually dialed available for me to route the
calls to different extensions? Thanks for the help and I apologize if this has
already been discussed.
Thanks,
-Herman
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050308/75afeb0c/attachment.htm