[Asterisk-Users] Asterisk SIP B2BUA without RTP proxy

Gustavo Garcia Bernardo ggb at tid.es
Mon Mar 22 05:08:36 MST 2004


I would like to use Asterisk as a SIP B2BUA, for CDR generation. I prefer to
avoid doing RTP proxy in Asterisk for SIP UAs for increasing performance.

Could i configure sip channel for that? Some kind of dont_touch_sdp=1?

Thank you very much.


-----Mensaje original-----
De: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]En nombre de Matteo
Enviado el: lunes, 22 de marzo de 2004 12:58
Para: asterisk-users at lists.digium.com
Asunto: [Asterisk-Users] setvar CALLERIDNUM

Is it possible to change the var CALLERIDNUM?
I need to put the 0 in front of the incoming number to be able to make a
redial on a missing call.
We need the 0 to rich the external line

I tried with
exten => s,1,Answer
exten => s,2,Setvar(CALLERIDNUM=0${CALLERIDNUM})

On console I see the new variable but in reality it does not change the
callerid number on the phone display.

Any suggestions?

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