Hi gentleman
I've configured SER to forward every call starting with sip uri request
"1" to Asterisk. I need to configure Asterisk as a Sip UAC in order to make it
call to my other SIP Provider outside my network, sending username and password
for authentication.
I've read at www.voip-info.org some articles but found none that could
suit to my needs, but yet I've found an article which explains an
implementation very similiar to what I need
(http://www.voip-info.org/wiki-Asterisk+voicepulse+connect), but in my
solution, I don't use IAX just sip terminatino via Internet.
I've tried to do exactly as this tutorial said, but with one
difference, all the entries at iax.conf I've made at sip.conf. The result is
that I can still connect my sip phone to my server but it doesn't give me an
outside line after I press 1. Have anyone implemented this solution or know
what I may be doing wrong ??
My configurations are following below:
Extensions.conf
exten => 1,1,Dial(SIP/<username>:<password>@go2call,30,rT)
exten => 2,1,Playback(tt-weasels)
exten => 2,2,Hangup()
exten => 3,1,Playback(tt-weasels)
Sip.conf
[go2call]
context = go2call
username=<username>
secret=<password>
auth=md5
type=friend
host=<go2callhost>
--
Felipe Martins
TEP Solution & New Technologies
Mundivox Communications
[EMAIL PROTECTED]
Site: www.mundivox.com
Tel.: +55 +21 +3820 8839
Cel.: +55 +21 +9823 8602
Fax.: +55 +21 +3820 8844
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