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() -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111031/3cee5835/attachment.htm>
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.elharit200 at 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 >
thank you so much all works without issue now 2011/10/31 Christian Gansberger <christian.gansberger at 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.elharit200 at 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111031/98d5f424/attachment.htm>
:) On 31 October 2011 15:36, salaheddine elharit <salah.elharit200 at gmail.com> wrote:> thank you so much all works without issue now > > > > 2011/10/31 Christian Gansberger <christian.gansberger at 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.elharit200 at 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 >