:) On 31 October 2011 15:36, salaheddine elharit <salah.elharit...@gmail.com> wrote: > thank you so much all works without issue now > > > > 2011/10/31 Christian Gansberger <christian.gansber...@accm.at> >> >> 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 >> <salah.elharit...@gmail.com> 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 >
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