[asterisk-users] asterisk as a Media Gateway
I am planning to use asterisk with Digium TDM2404E card as a media gateway to terminate traffic to Cell phones. Anyone got this working before with no problems, specially with Answer/Disconnect supervision? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk as a Media Gateway
I am planning to use asterisk with Digium TDM2404E card as a media gateway to terminate traffic to Cell phones. Anyone got this working before with no problmes, specially with Answer/Disconnect supervision? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Media Gateway
I am planning to use asterisk with Digium TDM2404E card as a media gateway to terminate traffic to Cell phones. Anyone got this working before with no problmes, specially with Answer/Disconnect supervision? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sending flash using DTMF
Is there is a way to send Asterisk FLASH using DTMF? I am trying to redial or dialing a new number without hangingup and start the whole process again. Thanks, Osama ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transcoding problem
I am having a problem with asterisk transcoding GSM and G729 codecs, the error message is below: Jun 14 09:38:12 WARNING[18292]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/3004-fcfb(256) to SIP/3003-c1c3(2) Jun 14 09:38:12 WARNING[18292]: app_dial.c:1586 dial_exec_full: Had to drop call because I couldn't make SIP/3004-fcfb compatible with SIP/3003-c1c3 == Spawn extension (test, 3003, 1) exited non-zero on 'SIP/3004-fcfb' sip.conf [3004] type=friend secret=x context=test callerid=test1 3004 nat=yes disallow=all allow=g729 host=dynamic canreinvite=yes dtmfmode=rfc2833 [3003] type=friend secret=x context=test callerid=test2 3003 nat=yes disallow=all allow=gsm host=dynamic canreinvite=yes dtmfmode=rfc2833 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] syslog server
I am using syslog-ng, with mysql, and php-syslog-ng, so you get a web interface to search for logs, and a huge capacity on the mysql databse, I have a syslog-ng with the above configuration, and is handleing 5 million syslog message per day. On 6/6/06, Matthew Warren [EMAIL PROTECTED] wrote: Does anyone know a good syslog server to use for grandstream phones?I wantto set this up to see what is happening with the grandstreams.Easy andFree preferably.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reinvite
does thia apply on SIP only or also IAX?On 6/5/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - Osama Kamal [EMAIL PROTECTED] wrote: I am running asterisk behind nat, and 2 sip phones on 2 different adsl neted connections, asterisk is staying always in rtp media path, while canreinvite=yes is configured in both extensions. I need asterisk to stay away from the rtp media path, what is wrong with that setup?It is nearly impossible to get a direct media path between two endpoints that are both behind NATs, regardless of the SIP server/proxy you use. Asterisk is no different in this regard.--Kevin P. FlemingSenior Software EngineerDigium, Inc.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reinvite
I am running asterisk behind nat, and 2 sip phones on 2 different adsl neted connections, asterisk is staying always in rtp media path, while canreinvite=yes is configured in both extensions. I need asterisk to stay away from the rtp media path, what is wrong with that setup? Regards, Osama Kamal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
