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...able-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY ------------------------------------------------------------------------ ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-us...

...[7000] type=endpoint context=outgoing-kamailio disallow=all allow=g722,alaw,ulaw,h264,h263p,h263,h261 transport=transport-udp auth=auth7000 aors=7000 direct_media=no disable_direct_media_on_nat=yes do I have to turn on the Video Support somewhere else ? -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140507/00685383/attachment.html>

...callid: 1bfa1fcfee1e20dbe9bbbcac5d7bdffc) - No matching endpoint found and here the RegEx for fail2ban to catch this log: |NOTICE.* .*: Request from '.*' failed for '<HOST>(:[0-9]{1,5})?' (.*) - No matching endpoint found Regards| -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 <callto:004922897167161> P2P: sip:rainer at sip.soho-piper.de:5072 (pjsip-test) XMPP: rainer at xmpp.soho-piper.de -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-user...

Hi Guys, I have recently moved my database servers to a different database cluster that runs on haproxy. Every minute or so I get the following error in asterisk MySQL RealTime: Ping failed (2006). Trying an explicit reconnect The strange thing is if I do realtime mysql status It shows as connected just the timer resets. Any idea why this is occurring? -------------- next

First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: <[TK]D-Fender> Note for elfranne's situation, : nat=force_rport,comedia" should have returned the public IP the call arrived on, but it

I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. thanks, Thufir

Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone