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
fmartins@mundivox.com
Site: www.mundivox.com
Tel.: +55 +21 +3820 8839
Cel.: +55 +21 +9823 8602
Fax.: +55 +21 +3820 8844
