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Since you've specified that the gs102 peer has a dynamic IP address,
you'll need to ensure that this peer registers with Asterisk, otherwise
it'll default to the 192.168.2.1 address in the config file.<br>
<br>
<br>
ast guy wrote:
<blockquote
cite="mid:c28c863a0802100304i4d04eff7uf5caf79badcf0081@mail.gmail.com"
type="cite">Will it require to add register statement in sip.conf. I
have all sip buddies in Database. so will that work in this scenario ?<br>
-ag<br>
<br>
<div class="gmail_quote">On Feb 10, 2008 11:55 AM, Rob Hillis <<a
moz-do-not-send="true" href="mailto:rob@hillis.dyndns.org">rob@hillis.dyndns.org</a>>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
Why are you specifying the password and server IP in the dial string
when it's included in sip.conf? It's unnecessary.<br>
<br>
I believe that Dial(SIP/gs102/1234) will achieve what you want.<br>
<br>
ast guy wrote:
<blockquote type="cite">
<div>
<div class="Wj3C7c">Hi,<br>
<br>
I'm trying to call a SIP server while providing the SIP server
username/password in dial string but it's not working ...<br>
<br>
Dial(<a moz-do-not-send="true" href="mailto:SIP/gs102:test@192.168.2.81"
target="_blank">SIP/gs102:test@192.168.2.81</a>);<br>
<br>
User on sip server (<a moz-do-not-send="true" href="http://192.168.2.81"
target="_blank">192.168.2.81</a>):<br>
<br>
[gs102]<br>
disallow=all<br>
allow=ulaw<br>
allow=alaw<br>
type=friend<br>
username=gs102<br>
secret=test<br>
host=dynamic<br>
dtmfmode=inband<br>
defaultip=<a moz-do-not-send="true" href="http://192.168.2.1"
target="_blank">192.168.2.1</a><br>
qualify=1000<br>
mailbox=102<br>
context=context-gs102<br>
<br>
Extensions.conf entry<br>
<br>
[context-gs102]<br>
<br>
exten => s,1, Answer();<br>
exten => s,n, Playback(demo-congrats);<br>
exten => s,n, Meetme(8600051);<br>
<br>
exten => 1234,1, Answer();<br>
exten => 1234,n, Playback(demo-congrats);<br>
exten => 1234,n, Meetme(8600051);<br>
<br>
<br>
When I dial I get following error on console<br>
<br>
-- Executing Dial("SIP/331-6263", "<a moz-do-not-send="true"
href="mailto:SIP/gs102:test@192.168.2.81" target="_blank">SIP/gs102:test@192.168.2.81</a>")
in new stack<br>
-- Called <a moz-do-not-send="true"
href="mailto:gs102:test@192.168.2.81" target="_blank">gs102:test@192.168.2.81</a><br>
-- SIP/192.168.2.81-0343 is circuit-busy<br>
== Everyone is busy/congested at this time (1:0/1/0)<br>
-- Executing Hangup("SIP/331-6263", "") in new stack<br>
== Spawn extension (default, 1234, 2) exited non-zero on
'SIP/331-6263'<br>
<br>
<br>
I want to call extension 1234 defined under gs102 defined context-gs102
context... what should be the exact Dialed SIP URL ?<br>
<br>
<br>
-ag<br>
</div>
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