[asterisk-users] Understanding Incoming sip DID handling
John Signorello
jsignorello at ispbx.com
Mon May 19 14:01:23 CDT 2008
You have DID1 on sip trunk 1 (unlimited channels)
You have DID2 on sip trunk 2 (restricted channels)
You want all you outgoing traffic to go out sip trunk 1
==
Sherwood McGowan wrote:
> Joseph L. Casale wrote:
>
>> Hi,
>> What is the method (preferred) way Asterisk handles the incoming
>> sip lines? I am currently trying to setup two lines, one has
>> unlimited in/out channels and the other phone number has only two.
>>
>> The provider has given a macro that manages dialing out on the two
>> possible servers.
>>
>> Would I match on phone number to decide where to send it? Both lines
>> can originate from two different servers so matching by IP wouldn't
>> help as both share either/or server.
>>
>> Thanks!
>> jlc
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
> Yes, in your dialplan you should have one extension set up for the first
> number and where to send it, and a second for the other.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
John Signorello
Managing Partner
ISPBX LLC
Bus: 866 GO ISPBX ext 2000
Dir: 973-841-2061
Cell: 973-534-0888
http://ispbx.com
http://cogoblue.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080519/259ca7d5/attachment.htm
More information about the asterisk-users
mailing list