search for: cfmc_cdi_private

...Are there any ideas how to troubleshoot this? I am not touching anything so I am not sure what caused the cancel to occur. This is with Asterisk Version:1.6.0.10 compiled from source running under CentOS 5.3. Here is the console output: [2009-06-26 08:12:47.896] -- Executing [retry_number at cfmc_cdi_private:1] UserEvent("Local/retry_number at cfmc_cdi_private-4ff4;2", "BeforeRetry,ActionID:RetryAndQueue & asterisk-16-1.oregon.cfmc.com-1246029167.102 & Local/retry_number at cfmc_cdi_private-4ff4;2 & 1001 & SIP/GXP280_18 & 10 & another-time & 10 & 7")...

Hello User-List, is it possible to play a sound file directly to a caller channel? Like this AMI command Action: Originate Channel: SIP/20-00001d41 Application: Playback Data: /path/to/audio/file I get an Error Message. My intension is to play a sound file to a caller and the other callers don't hear this. Can someone help me ? Thanks a lot Bye Daniel

Hi all, How can I playback a file within an active call? I've tried with ChanSpy whisper mode like this (using AMI): Action: Originate Channel: Local/9999 at default Priority: 0 Variable: MSG=test Application: ChanSpy Data: SIP/1234-123 Async: 1 and in the dialplan: [default] exten => 9999,1,Answer() exten => 9999,n,Wait(2) exten => 9999,n,Playback(${MSG}) Where

Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel variables when I originate a call via AMI. All the documentation I have found says to do it like this: Variable: variable1=value|variable2=value|variable3=value However when I do this it runs them all together and I end up with:

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

Hello! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and now want to add a SIP-Diversion header that changes the CallerID of a number that is not available on the trunk, the CallerID to be visible externally

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

...like this: exten => do_monitor,1,Answer() exten => do_monitor,n,NoOp(Just got '${CfMC_ActionID}') exten => do_monitor,n,ChanSpy(${CfMC_WhoHear},qX) exten => do_monitor,n,Hangup() I use an AMI packet like this: Action: Originate Channel: Agent/1001 Exten: do_monitor Context: cfmc_cdi_private Priority: 1 Variable: CfMC_ActionID=callE1334 Variable: CfMC_WhoHear=SIP/dickenson ActionID: callE1324 Async: true It seems as if the X option to the ChanSpy application causes asterisk to get into a loop executing priorities 1-3 of the extension until I press a 1 to go to extension 1 in the same...