<div dir="ltr"><div>thank you so much all works without issue now </div>
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<div class="gmail_quote">2011/10/31 Christian Gansberger <span dir="ltr"><<a href="mailto:christian.gansberger@accm.at">christian.gansberger@accm.at</a>></span><br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Hello,<br><br>You have to disable RTP-Encryption on your Snom under Identity, RTP.<br>It is set to on per default.<br>
<br><br>On 31 October 2011 13:33, salaheddine elharit<br>
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<div class="h5"><<a href="mailto:salah.elharit200@gmail.com">salah.elharit200@gmail.com</a>> wrote:<br>> hello list<br>><br>> i have installed asterisk 1.8.7.1 and i have configured 2 account for sip in<br>
> order to do internal call<br>><br>> when i use x-lite and eyebeam1.5 i can call from 222 to 223 ,and alson from<br>> 223 to 222<br>><br>> but when i use my snom 320 i can call from my x-lite or eyebeam1.5 to<br>
> snom320 but the issue i can not call from my snom<br>><br>> i have this issue just Asterisk 1.8 when i tested with asterisk 1.4 theres<br>> is no problem<br>><br>> see the sip.conf and extenssions.conf below and also the cli when i try to<br>
> call from my snom to x-lite<br>><br>> thanks and regards<br>><br>> CLI<br>> == Using SIP RTP CoS mark 5<br>> [Oct 31 12:29:17] WARNING[16515]: chan_sip.c:8843 process_sdp: We are<br>> requesting SRTP, but they responded without it!<br>
> salaheddine*CLI><br>><br>> sip.conf<br>><br>><br>> [general]<br>> context=agents<br>> allowguest=yes<br>> allowoverlap=no<br>> allowtransfer=yes<br>> allow=alaw<br>> allow=ulaw<br>
> allow=gsm<br>> allow=ilbc<br>> [222]<br>> type=friend<br>> context=agents<br>> host=dynamic<br>> dtmfmode=auto<br>> disallow=all<br>> allow=alaw<br>> allow=ulaw<br>> qualify=yes<br>><br>
><br>> [223]<br>> type=friend<br>> context=agents<br>> host=dynamic<br>> dtmfmode=auto<br>> disallow=all<br>> allow=alaw<br>> allow=ulaw<br>> qualify=yes<br>><br>> extenssions.conf<br>><br>
><br>> [agents]<br>><br>> exten => 222,1,Dial(SIP/222)<br>> exten => 222,n,Hangup()<br>> exten => 223,1,Dial(SIP/223)<br>> exten => 223,n,Hangup()<br>><br></div></div>> --<br>> _____________________________________________________________________<br>
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New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></font></blockquote></div><br></div>