[asterisk-users] call rejected because extension not found in context 'internal
Justin Hester
jhester at digium.com
Mon Feb 3 08:00:03 CST 2014
Howdy,
Your sip.conf file looks fine for some testing, though I would recommend
_not_ using an extension number to name a sip endpoint. Instead, name the
sip endpoint something more descriptive of the device. [Linphone-01]
[Linphone-02] for example. Then you'll want to configure extensions.conf to
Dial() the sip endpoint whenever the extension is dialed.
Justin Hester
Digium, Inc. · Technical Trainer
445 Jan Davis Drive NW · Huntsville, AL 35806 · USA
ph: +1 256 428 6238
Check us out at: http://digium.com · http://asterisk.org
On Mon, Feb 3, 2014 at 5:45 AM, Raghav Goud <raghavgoud.g at gmail.com> wrote:
> Hi all,
>
> I want to two sip clients connect through Asterisk in local network for
> testing. My sip.conf file looks like this
>
> [general]
> context=internal
> allowguest=no
> allowoverlap=no
> bindport=5060
> bindaddr=0.0.0.0
> srvlookup=no
> disallow=all
> allow=ulaw
> alwaysauthreject=yes
> canreinvite=no
> nat=yes
> session-timers=refuse
> localnet=192.168.1.0/255.255.255.0
>
> [7001]
> type=friend
> host=dynamic
> secret=123abcd
> context=internal
>
> [7002]
> type=friend
> host=dynamic
> secret=456abcd
> context=internal
>
>
> Am using linphone as sip client and create account on linphone with user
> name 7001 and 7002
> 7001 is running on 192.168.2.15:5060
> 7002 is running on 192.168.2.45:5060
>
> when i try to call from 7002 to 7001 i specified sip:7001 at 192.168.2.15 it
> working fine as i know ip adress i specified it as url. if i dnt know the
> ipadress how can i call to 7001? i try to call sip:7001 at 192.168.2.20 it
> through call rejected because extension not found in context 'internal,
> error.
>
> How can call to sip id with out knowning ipadress where it is runnning?
> Any modification required for sip.conf file?
>
> Thanks,
> Raghav
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> 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/20140203/528779f4/attachment.html>
More information about the asterisk-users
mailing list