Then you sip.conf is not defined properyly, or your extensions.conf does
not have any end, the only way the call should really go back is if
asterisk is telling it to, see what is called in what order in your
dialplan, remeber the dialplan is not called in the order it is written
in the file
Iqbal
Mohamed A. Gombolaty wrote:
Dear Yair,
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.
Thx
MAG
Yair Hakak wrote:
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:[EMAIL PROTECTED], 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 <[EMAIL PROTECTED]> 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
[EMAIL PROTECTED], 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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Thx
MAG
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