rob at hillis.dyndns.org Guest
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Posted: Sun Feb 10, 2008 10:44 am Post subject: [asterisk-users] Dialing SIP server user extension... Dial s |
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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.
ast guy wrote:
Quote: | Will it require to add register statement in sip.conf. I have all sip
buddies in Database. so will that work in this scenario ?
-ag
On Feb 10, 2008 11:55 AM, Rob Hillis <rob at hillis.dyndns.org
<mailto:rob at hillis.dyndns.org>> wrote:
Why are you specifying the password and server IP in the dial
string when it's included in sip.conf? It's unnecessary.
I believe that Dial(SIP/gs102/1234) will achieve what you want.
ast guy wrote:
Quote: | Hi,
I'm trying to call a SIP server while providing the SIP server
username/password in dial string but it's not working ...
Dial(SIP/gs102:test at 192.168.2.81
<mailto:SIP/gs102:test at 192.168.2.81>);
User on sip server (192.168.2.81 <http://192.168.2.81>):
[gs102]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=gs102
secret=test
host=dynamic
dtmfmode=inband
defaultip=192.168.2.1 <http://192.168.2.1>
qualify=1000
mailbox=102
context=context-gs102
Extensions.conf entry
[context-gs102]
exten => s,1, Answer();
exten => s,n, Playback(demo-congrats);
exten => s,n, Meetme(8600051);
exten => 1234,1, Answer();
exten => 1234,n, Playback(demo-congrats);
exten => 1234,n, Meetme(8600051);
When I dial I get following error on console
-- Executing Dial("SIP/331-6263", "SIP/gs102:test at 192.168.2.81
<mailto:SIP/gs102:test at 192.168.2.81>") in new stack
-- Called gs102:test at 192.168.2.81
<mailto:gs102:test at 192.168.2.81>
-- SIP/192.168.2.81-0343 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("SIP/331-6263", "") in new stack
== Spawn extension (default, 1234, 2) exited non-zero on
'SIP/331-6263'
I want to call extension 1234 defined under gs102 defined
context-gs102 context... what should be the exact Dialed SIP URL ?
-ag
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