thank you so much all works without issue now
2011/10/31 Christian Gansberger <[email protected]> > Hello, > > You have to disable RTP-Encryption on your Snom under Identity, RTP. > It is set to on per default. > > > On 31 October 2011 13:33, salaheddine elharit > <[email protected]> wrote: > > hello list > > > > i have installed asterisk 1.8.7.1 and i have configured 2 account for > sip in > > order to do internal call > > > > when i use x-lite and eyebeam1.5 i can call from 222 to 223 ,and alson > from > > 223 to 222 > > > > but when i use my snom 320 i can call from my x-lite or eyebeam1.5 to > > snom320 but the issue i can not call from my snom > > > > i have this issue just Asterisk 1.8 when i tested with asterisk 1.4 > theres > > is no problem > > > > see the sip.conf and extenssions.conf below and also the cli when i try > to > > call from my snom to x-lite > > > > thanks and regards > > > > CLI > > == Using SIP RTP CoS mark 5 > > [Oct 31 12:29:17] WARNING[16515]: chan_sip.c:8843 process_sdp: We are > > requesting SRTP, but they responded without it! > > salaheddine*CLI> > > > > sip.conf > > > > > > [general] > > context=agents > > allowguest=yes > > allowoverlap=no > > allowtransfer=yes > > allow=alaw > > allow=ulaw > > allow=gsm > > allow=ilbc > > [222] > > type=friend > > context=agents > > host=dynamic > > dtmfmode=auto > > disallow=all > > allow=alaw > > allow=ulaw > > qualify=yes > > > > > > [223] > > type=friend > > context=agents > > host=dynamic > > dtmfmode=auto > > disallow=all > > allow=alaw > > allow=ulaw > > qualify=yes > > > > extenssions.conf > > > > > > [agents] > > > > exten => 222,1,Dial(SIP/222) > > exten => 222,n,Hangup() > > exten => 223,1,Dial(SIP/223) > > exten => 223,n,Hangup() > > > > -- > > _____________________________________________________________________ > > -- 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 >
-- _____________________________________________________________________ -- 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
