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 <[email protected]> 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:[email protected] 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:[email protected] 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 >
-- _____________________________________________________________________ -- 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
