Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and which has a uri rule that says forward 4 digit starting with 1 to the asterisk sip port the asterisk extensions.conf has an entry for 1999 and dials 999@ser-server, if not answered voicemail runs and so on. ain't there a way to make 666 directly call 999 without using 1999. -- Thx MAG
yes, there is. run everything through asterisk, no matter how long the extensions are. for example, 666 calls 999 goes to asterisk, sees a dial sip:999@ser, goes back to SER. bounces back to ser. If everything is working well asterisk will set up the call and get out of the way. I don't see why you need to prepend digits in order to make this work, if i'm missing something let me know. -yair On 6/16/05, Mohamed A. Gombolaty <mgombolaty@noorgroup.net> wrote:> Dear All, > > I am trying to make the phones always talk to each other (peer to peer) > using SER as a sip proxy, and incase the call is not answered we will > use the voicemail of asterisk and other feautures, I have done that > already, but in order to do so I found that I have to make the users > dial different exten numbers, here is an example: > > user with exten 666 wants to call 999 . > 666 dials 1999 and which has a uri rule that says forward 4 digit > starting with 1 to the asterisk sip port > the asterisk extensions.conf has an entry for 1999 and dials > 999@ser-server, if not answered voicemail runs and so on. > > ain't there a way to make 666 directly call 999 without using 1999. > > > -- > Thx > MAG > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
If these are the only calling rules you could try
if (!lookup(location))
{
t_relay to your asterisk box
break
}
Mohamed A. Gombolaty wrote:
>Dear All,
>
>I am trying to make the phones always talk to each other (peer to peer)
>using SER as a sip proxy, and incase the call is not answered we will
>use the voicemail of asterisk and other feautures, I have done that
>already, but in order to do so I found that I have to make the users
>dial different exten numbers, here is an example:
>
>user with exten 666 wants to call 999 .
>666 dials 1999 and which has a uri rule that says forward 4 digit
>starting with 1 to the asterisk sip port
>the asterisk extensions.conf has an entry for 1999 and dials
>999@ser-server, if not answered voicemail runs and so on.
>
>ain't there a way to make 666 directly call 999 without using 1999.
>
>
>--
>Thx
>MAG
>
>
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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asterisk-users-bounces@lists.digium.com wrote:> Actually what happens is that from SER debug I can see the call is > looping between Asterisk and SER. but adding a number makes no > loops.Check what the origin (IP/DNS name) of the incoming SIP message is. If it's from asterisk, send it to the user, if it is not from asterisk, it must be meant to go to asterisk. Add a couple of other tests (known user, etc) to it and then I think you'll have what you're looking for. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45