VOIP-info.org Comments

Asterisk n-way call HOWTO / res_features.c :1478 ast_bridge_call : Bridge failed on channels [ID: 41870]

Wed, 01 Aug 2007 07:26:00 GMT

Can any one help me? When i try to add a person in the established call, the first person redirected in the confernce room (OK) i dial the number of the second personne i talk with him (OK) when i try to accept him(**) i have a hangup and the 2 other party can talk in the confÃ©rence room the same think was reproduced when i refused to invite him! this error was genereted when i passed a normal call first and try to invite other party :" res_features.c:1478 ast_bridge_call: Bridge failed on channels SIP/XXXXXXXXXXXXXXX and AsyncGoto/SIP/XXXXXXXXXXXXXXX " if confernece established and one of the party try to invite some one there is no probleme!!! Help please
bahbouh (karray) at 2007-08-01 07:26

MixMonitor / Re: Can you show me a sample...?! [ID: 41866]

Wed, 01 Aug 2007 05:24:23 GMT

I know it!! Thank U~~?!
ksdgeni (Geni kim) at 2007-08-01 05:24

TelIAX / Re: Service and Support given to Mr. Ogali [ID: 41861]

Wed, 01 Aug 2007 02:03:31 GMT

Update: Mr. Ogali was contacted on Monday morning (the first business day after he sent in his inquiry) and all issues were resolved, ostensibly to his satisfaction. He remains a Teliax subscriber...
daldworth (David) at 2007-08-01 02:03

Asterisk UNISTIM channels / i2004 black - one way audio [ID: 41860]

Wed, 01 Aug 2007 01:52:31 GMT

Hi All I have a nortel i2004, and I have tried every RTP mode between 0 and 3. For 1 and 2 I get audio to the phone, but I cannot send audio or dtmf tones. I cannot get it to work. This is my config. Anyone please help! general keepalive=120 ; in seconds, default = 120 bindaddr=192.168.205.5 [black] device=000ae4765d8a rtp_port = 10000 ; RTP port used by the phone, default = 10000. RTCP = rtp_port+1 rtp_method=1 ; If you don't have sound, you can try 1 or 2, default = 0 titledefault=Eastern ; default = "TimeZone (your time zone)". 12 characters max maintext0="jonathans phone2" ; default = "Welcome", 24 characters max maintext1="JC pbx" ; default = the name of the device, 24 characters max maintext2="(main page)" ; default = the public IP of the phone, 24 characters max dateformat=1 ; 0 = month/day, 1 (default) = day/month timeformat=0 ; 0 = 0:00am ; 1 (default) = 0h00, 2 = 0:00 contrast=8 ; define the contrast of the LCD. From 0 to 15. Default = 8 nat=0 ; control ast_rtp_setnat, default = 0 callerid="JC" <555-234-5678> context=default ; context, default="default" mailbox=7104 ; Specify the mailbox number. Used for Message Waiting Indication linelabel="Support" ; Softkey label for the next line=> entry, 9 char max. line => 101 bookmark=jonathan@7101 ; Use a softkey to dial 123. ;bookmark=Mailbox@011@54 ; 54 shows a mailbox icon. See #define FAV_ICON_ for other values (32 to 63) ;bookmark=Test@*@USTM/violet ; Display an icon if violet is connected (dynamic), only for unistim device Thanks
jc179 (Jonathan) at 2007-08-01 01:52

Asterisk priorities / n Priority Labels to other extensions? [ID: 41858]

Wed, 01 Aug 2007 01:10:18 GMT

Is there a way to jump from one extension to a label on another extension using n Priority? I'm updating my DialPlan Compiler (http://ast-dpc.sourceforge.net/) to include the option to compile using this syntax, but for this to work, there must be a way to jump to a label on one extension from another extension.
chochos () at 2007-08-01 01:10

MixMonitor / Re: Can you show me a sample...?! [ID: 41855]

Tue, 31 Jul 2007 23:53:50 GMT

I know it!! Thank U~~?!
ksdgeni (Geni kim) at 2007-07-31 23:53

Web based Asterisk Database maintenance / Issue displaying existing values [ID: 41850]

Tue, 31 Jul 2007 21:37:20 GMT

I am able to add/delete values but existing ones are not showing in the table. They show fine from the Asterisk CLI when I type "database show": /blacklist/01223949492 : 1 /blacklist/01223949432 : 2 etc.
jbassett (Jason Bassett) at 2007-07-31 21:37