[asterisk-users] Re: some simple newbie help with dialplan needed...

[Evert]

Hi!

I have SER running on 5060, and Asterisk on 5061, so if Asterisk passes
it to @localhost there should be no problem there.

The remaining problem seems to be the passing on of the call. The
mentioned dialplan does send the entered extension to SER, and the phone
on that SER-account rings. However, when the called phone anwers the
call, the calling phone does not notice this, and keeps on ringing...
No connection is established.

I am not sure whether this is caused by Asterisk or SER, so I am sending
this to the SER-group as well...  :-)

Regards,
  Evert



William Piper wrote:
> Does your variable ${SERADDRESS} have the port number defined? If you
> replace it with localhost, I believe port 5060 will pick up the call by
> default.
>  
> Try making sure that your ${SERADDRESS} has the IP like the following:
> 127.0.0.1:5070 <http://127.0.0.1:5070> (replace 5070 with whatever port
> your SER is listening on).
>  
> I've never tried sending the call through loopback but it seems like it
> should work.
>  
> bp
> 
>  
> On 11/6/06, *Evert* <evert at poboxes.info <mailto:evert at poboxes.info>> wrote:
> 
>     SERADDRESS is a variable that points to a SER-server (currently running
>     on the same IP as Asterisk, but on a different port).
> 
>     I noticed that it helped to replace @${SERADDRESS} with @localhost
> 
>     I now have:
> 
>     exten => 998,1,Answer
>     exten => 998,2,Background(agent-newlocation)
>     exten => 998,3,Set(TIMEOUT(digit)=15)
>     exten => 998,4,WaitExten(10)
>     exten => _XXXX,1,Dial(SIP/${EXTEN}@localhost,60,o)
> 
> 
>     This does send the entered extension to SER, and the phone rings.
>     However, when the called phone anwers the call, the calling phone does
>     not notice this, and keeps on ringing...
> 
> 
> 
> 
>     Regards,
>     Evert
> 
>     Leo Ann Boon wrote:
>     > Evert wrote:
>     >> Hi! :)
>     >>
>     >> Thanks for the tip. I'm almost there now, the only problem that I
>     have
>     >> left is that I do NOT want Asterisk to check whether the extension
>     >> entered is valid. In the current setup Asterisk will refuse to
>     forward
>     >> the call since it thinks the extension is invalid...  :-/
>     >>
>     >>
>     > Is ${SERADDRESS} the name of a valid SIP peer or just plain ole IP
>     > address? It should be a SIP peer.
>     >
>     > Leo