[asterisk-users] REALY strange issue with making calls biside 2 phones
Thats my issue, i hope someone could suggest something: Phone A - Phone B == Using SIP RTP CoS mark 5 -- Executing [01@default:1] Dial(SIP/00-0076, SIP/01) in new stack == Using SIP RTP CoS mark 5 -- Called 01 -- SIP/01-0077 is ringing -- SIP/01-0077 answered SIP/00-0076 -- Locally bridging SIP/00-0076 and SIP/01-0077 == Spawn extension (default, 01, 1) exited non-zero on 'SIP/00-0076' Phone B - phone A == Using SIP RTP CoS mark 5 -- Executing [00@default:1] Dial(SIP/01-0078, SIP/00) in new stack [Jul 12 19:08:35] WARNING[2965]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [00@default:2] Hangup(SIP/01-0078, ) in new stack == Spawn extension (default, 00, 2) exited non-zero on 'SIP/01-0078' -- -- Best regards Matiss Jekabsons Procerto Ltd. -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with setting up fresh 1.8.5 Asterisk
Is there some detailed documentation for 1.8.5? I am tryin to make Asterisk 1.8.5 with MySQL backend, TLS transport and SRTP encryption. For now with no success :-(-- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with setting up fresh 1.8.5 Asterisk
Cool thx :) dont know why i didnt found it myself :D Quoting Patrick Lists asterisk-l...@puzzled.xs4all.nl: On 07/10/2011 05:02 PM, Matiss Jekabsons wrote: Is there some detailed documentation for 1.8.5? I am tryin to make Asterisk 1.8.5 with MySQL backend, TLS transport and SRTP encryption. For now with no success :-( https://wiki.asterisk.org/wiki/display/AST/Asterisk+1.8+Documentation Regards, Patrick -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Best regards Matiss Jekabsons Procerto Ltd. ICT project manager GSM: (+371) 22440298 E-Mail: mat...@procerto.lv Dzelzavas Str. 117. Riga, Latvia-- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help anybody - how to manage SRTP with TLS trasport
Working on that about a week and not getting closer. Now upgrading to Asterisk 1.8.5 with a bit of hope that will work. TLS is working just fine, but not SRTP. Module is not loading and thats it. Asterisk 1.8.4.4 -- Best regards Matiss Jekabsons Procerto Ltd. -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange behavior with Asterisk TLS
Hallo there! 1. i have Asterisk 1.8.2.2 2. installed on Debian 3. MySQL used as backend 4. I have configured TLS openssl genrsa -out key.pem 1024 openssl req -new -key key.pem -out request.pem rtc in sip.conf tlsenable=yes tlsbindaddr=0.0.0.0 [i am begind NAT] tlscertfile=/etc/asterisk/certificates/hereismyfilename.pem tlsdontverifyserver=no tlscipher=DES-CBC3-SHA tlsclientmethod=tlsv1 And to one test client i added: transport=tls Then i tryed to connect with BRIA softphone with enabled transport TLS and there is nothing happening... the phone is not connecting anv there is no message in CLI. But when i try to connect thru UDP then CLI says: Device not configured to use this transport type Only with sip cinfigured to udp all is working. Where could be my dumb mistake? Thx for any suggestion! -- Best regards Matiss Jekabsons Procerto Ltd.-- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About recompile reinstall couse of SRTP
Hi there! My issue is that i have (bouht to me) a box with Asterisk 1.8.2.2 but its seems that with no SRTP support. So i added a libsrtp libraries and like i understand now i need to recompile/reinstall Asterisk... is that safe to all data and MySQL backand data? -- Best regards Matiss Jekabsons Procerto Ltd. Dzelzavas Str. 117. Riga, Latvia-- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
