[asterisk-users] sip issue

salaheddine elharit salah.elharit200 at gmail.com
Mon Oct 31 09:36:12 CDT 2011


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()
> >
> > --
> > _____________________________________________________________________
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> >               http://www.asterisk.org/hello
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> >
>
> --
> _____________________________________________________________________
> -- 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
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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