[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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