asterisk users - Feb 2008 - Dialing SIP server user extension... Dial string issue...

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);

User on sip server (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
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") in new
stack
    -- Called 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080209/1551c934/attachment.htm

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:
> 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
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080210/69130abc/attachment.htm