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...isk will stop setting the variable MEETME_RECORDINGFILE and start placing the recordings in the sounds directory named meetme-conf-rec.######.wav. Which is the default is MEETME_RECORDINGFILE is not set. Anyone seen this issue before? Thanks! Forrest Beck jonforrest.beck at gmail.com www.shift8.biz #!/bin/bash #Set some variables USFACULTY="ast-phonepages at somedomain.com" LSFACULTY="ast-phonepages at somedomain.com" USFACULTY="ast-phonepages at somedomain.com" MONTH=`date +%B` DAY=`date +%d` YEAR=`date +%Y` HOUR=`date +%I` MINUTE=`date +%M` ZONE=`date +%Z...

...c as a prefered codec in the Polycom provisioning files for the 550's? I couldn't find a pref for voice.codecPref.IP_550. What needs to be put into the allow field (sip.conf) for asterisk to allow the codec? -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck at gmail.com http://www.shift8.biz

...recorded somewhere, Ast DB or MySQL DB. Then when the audio file is played back to a caller, the system will say something like. This message was recorded January 14th at 10 42 pm Thanks for any ideas you may have. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck at gmail.com http://www.shift8.biz

Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan

Hi folks, How I can change default level in asterisk from 3 level to 7level, using the script /etc/init.d/asterisk

...ic? I am looking for something like the old norstar beep every few seconds. I tried 3 seconds silence, beep.wav, beep.wav. But it just didn't sound right. I need one that has a "softer" beep. Thanks! -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck at gmail.com http://www.shift8.biz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080221/c736648e/attachment.htm

I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call "22" and the phone rang it did not auto answer. Did I miss something? exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten => 22,n,SipAddHeader(Alert-Info: Ring Answer) exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten =>

I am currently using a T1 PRI from TWTelecom for DID and outgoing calls, but I recently discovered that they're offering call termination/origination over SIP trunks in my area now. If they could deliver these SIP trunks to me over a guaranteed-QoS circuit, this would be of great interest to me. We're already using a DS3 circuit from TW for our internet uplink, so I'd imagine it

Hello Everyone! According to voip-info.org the correcy syntax for the realtime function is: REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read REALTIME(family|fieldmatch|value|field) on write It seems from the syntax that it is only possible to retrieve a full row according to the value of only of column. This translates in SQL language as Select * from family where fieldmath =

Hi, I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The add-ons pack has been installed for a while, so now I'm trying to add the Mysql config. I've created a mysql database, added the grants for a user acces, and can run a mysql -u asteriskcdruser -p and can connect to the database. I've been using this as a guide:

Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] ??? exten => 99,1,ANSWER() ??? exten => 99,2,PLAYBACK(tt-monkeys) ??? exten => 99,3,HANGUP() The phone

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim