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...kes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI> == Using SIP RTP CoS mark 5 -- Executing [6003 at myphones:1] Set("SIP/6001-0000000c", "_SIPSRTP_CRYPTO=enable") in new stack -- Executing [6003 at myphones:2] Dial("SIP/6001-0000000c", "SIP/6003") in new stack == Using SIP RTP CoS mark 5 -- Called 6003 -- SIP/6003-0000000d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallth...

Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the /encryption=no/ in /sip.conf /and the /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have/ noload=res_srtp.so/ on my /modules.conf /otherwise I cannot place SIP calls (cause other ends don't support it) Regards, Miguel Baptista -------------- next part -------------- An HTML attachment was...

...gt; sip.conf as the shared secret. http://www.voip-info.org/wiki/view/Asterisk+SRTP updated test srtp server (asterisk SVN-trunk-r61760 + latest SRTP patch) voice2.fpf.slu.cz test sip accounts 700:700 701:701 702:702 extensions.conf exten => 600,1,Set(_SIPSRTP=optional) exten => 600,n,Set(_SIPSRTP_CRYPTO=enable) exten => 600,n,Playback(demo-echotest) ; Let them know what's going on exten => 600,n,Echo ; Do the echo test exten => 600,n,Playback(demo-echodone) ; Let them know it's over exten => 600,n,hangup exten => 610,1,Set(_SIPSRTP=require) exten => 610,n,Set(_SIPSRTP_MI...

...sip.conf: [Asterisk_SRTP_2] type=peer secret=123 host=dynamic defaulthost=192.168.69.176 port=5070 dtmfmode=rfc2833 insecure=invite,port qualify=no canreinvite=no extensions.conf: exten => 355,1,NoOP(Encrypted Call) exten => 355,n,Set(_SIPSRTP=require) exten => 355,n,Set(_SIPSRTP_CRYPTO=enable) exten => 355,n,Dial(SIP/Asterisk_SRTP_2,30) exten => 355,n,Hangup() If you need anymore info please let me know! Thanks in advance for your answers. Any help will be appreciated. Best regards, ?scar Patr?cio

...Content-Type: text/plain; charset="iso-8859-1" Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the /encryption=no/ in /sip.conf /and the /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have/ noload=res_srtp.so/ on my /modules.conf /otherwise I cannot place SIP calls (cause other ends don't support it) Regards, Miguel Baptista -------------- next part -------------- An HTML attachment was...