[OpenSIPS-Users] Opensips + asterisk 1.4
Willian Mazzardo - SYSSVOIP
willian at syssvoip.com.br
Wed Jul 17 03:32:34 CEST 2013
Hi all..
I know this is a very simple scenario, all PSTN calls be routed to asterisk
to do the billing job, but im having some problems, this is my scenario:
Sip Client (10.0.0.3) > Opensips (10.1.1.2) > Asterisk (10.1.1.247) ..... >
PSTN
Calls between sip clients on Opensips are working, but when I try to call
over Asterisk, I have Proxy authentication problem.
Here is my logs:
Opensips: http://pastebin.com/SWpuRHku
Asterisk: http://pastebin.com/6jp50LSS
[opensips]
host=10.1.1.2
type=friend
context=callingcard
qualify=no
insecure=very
fromdomain=10.1.1.2
Route: http://pastebin.com/mLgpXiNx
Can someone help me on this?
Thanks
Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br
55 3537 2030
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