search for: asterisk_config

...eers still have to be declared in sip/iax2.conf to make calls... However, when I restart Asterisk: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 192.168.0.10:5060 == Using TOS bits 0 Connected to database 'asterisk_config' on 'localhost' as 'asterisk_user' == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Apr 16 14:10:34 WARNING[1074449120]: chan_iax2.c:6218 load_...

...or various Astlinux platform builds to complete on making the contents of asterisk/config a little more complete, a little more useful, a little more real-world... I started looking at the possibility of taking JTodd's ISN Freenum cookbook example and updating it: http://freenum.org/cookbook/#ASTERISK_CONFIG but ran into some problems having to do with the following: we use SIP handsets in house, and those operate in the redfish-solutions.com domain... but anonymous SIP (freenum) calls coming in off the internet would also be in the same domain. I can't figure out how to separate calls in the sa...

.... 00a09a 00 99 00 45 00 44 c0 0e c0 04 00 2f 00 96 00....E.D...../... 00b041 00 15 00 12 00 09 00 14 00 11 00 08 00 06 00A............... 00c0ff 02 01 00 00 12 00 23 00 00 00 0f 00 01 01 00.......#........ 00d00e 00 05 00 02 00 01 00........ Setup of Asterisk PBX is described in the attachment asterisk_config.txt Any help is HIGHLY appreciated !! I wonder if anyone has ever really made calls from DTLS-SRTP WebRTC-client to Asterisk.. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140707/1e806c...

I made a little write-up that attempts to synthesise a lot of the information out there about how to get HylaFAX working with Asterisk by way of IAXmodem for inbound faxing: http://blog.evaristesys.com/?p=24 Of course, there are bound to be some things I've left out or are grossly in need of correction. So, before I link it off the voip-wiki I am extremely eager to solicit the input of

...could not be handled, bad request: 1832465624 at 192.168.0.25 This error is different to the error that is received if a username that is not in the MySQL sip_peers / sip_users table is specified. Therefore at least the MySQL connection appears to be working. extconfig.conf: sipusers => mysql,asterisk_config sippeers => mysql,asterisk_config I have also tried explicitly adding ',sip_users' and ',sip_peers' to these lines, but asterisk behaved similarly. res_mysql.conf dbhost = 127.0.0.1 dbname = asterisk_config dbuser = asterisk dbpass =<snip> ;dbport = 3306 dbsock = /tmp/mys...