Thanks a lot Joshua. That's a great idea :-) I will try it and get back to you.
On Thu, Apr 10, 2014 at 5:50 PM, Joshua Colp <jc...@digium.com> wrote: > Yaron Nachum wrote: > >> Hi everyone, >> > > Kia ora, > > > I am starting to work with PJSIP on release 12.1.0.rc3. >> >> I used to have Asterisk 1.8 with the regular sip channel. I was using >> the usereqphone settings in order to set user=phone on from and to URIs. >> >> Is there a similar config in PJSIP? >> > > There is currently no option which explicitly does this. A core difference > between chan_sip and chan_pjsip, though, is that we treat stuff as SIP URIs > in the first place. To that end it may be possible to simply add > ;user=phone to the SIP URI yourself and have it work. Since ';' is used for > comments though you will need to escape it using \. > > An example for a static contact on an AOR: > > contact=sip:172.16.1.1:5060;user=phone > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- 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
