Raghav Goud
2014-Feb-03 11:45 UTC
[asterisk-users] call rejected because extension not found in context 'internal
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140203/6aed1f1d/attachment.html>
Justin Hester
2014-Feb-03 14:00 UTC
[asterisk-users] call rejected because extension not found in context 'internal
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>