[Asterisk-Users] (no subject)
reacend at gmx.net
reacend at gmx.net
Tue May 25 08:56:41 MST 2004
USE SUBJECTS!
On Tue, May 25, 2004 at 03:51:17PM +0100, Graham Turner wrote:
> was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing ..
>
> given the example i work from on john todd's www.loligo.com site ;
>
> exten => _3.,1,SetCallerID(${IPTELUSERID})
> exten => _3.,2,SetCIDname(${IPTELUSERNAME})
> exten => _3.,3,Dial(SIP/${EXTEN:1}@iptel)
> exten => _3.,4,Playback(invalid)
> exten => _3.,5,Hangup
>
> are the first two statements mandatory to connection to iptel.org ??
>
> i understand the dial plan of any numbers prefixed by 3 are interpreted as iptel extensions, and that the third extension priority strips off the prefix
>
> presumably the call is then 'processed' by the [iptel] section of sip.conf and generates sip call with the credentials in this section as those passed to the iptel.org server ??
>
> how (iassuming they do) these relate in any way to the IPTELUSERID / IPTELUSERNAME variables defined in extensions.conf ??
>
> presumably (and i am using a 7940 sip device) i can dial an asterisk extension of targetiptelusername prefixed with 3 to call the sip user registered with
>
> qu. does the target iptel username need the iptel.org domain appended to it or is it somehow 'implied' by the above ??
>
> another wiki example suggests the use of a 'fromdomain' statement in an [iptel] section of sip.conf ??
>
> TIA
>
> GT
>
>
>
>
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