[asterisk-users] sip issue

[Christian Gansberger]

:)

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()
>> >
>> > --
>> > _____________________________________________________________________
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>> >
>>
>> --
>> _____________________________________________________________________
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>>
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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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