similar to: ChanSpy problem

I would like to be able to stop the chanspy application and go to the next step in the dialplan but I do not see a way to do that. I have looked at the code and I do not see a way to stop the chanspy application. Even if there are no channels that match the chanprefix pattern the chanspy application is not exited. Hitting the * key stops spying on a channel but then starts spying on the same

I issue this command: RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ ueue^SIP/GXP280_18)) Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds. Asterisk rings again for 10 seconds. I would expect this to happen a total of 4 times. The problem is that after the second ring for 10 seconds Asterisk exits the RetryDial step with HANGUPCAUSE=0 and

Is there any way that I can use AMI to play a sound file to both legs of a call without either issuing two commands, one per leg, or setting up meeting rooms? I would like to be able to play a sound file that can be heard by the caller and the person called using AMI. The only way so far I have been able to do this is the following. One problem with this is that people will hear sound out of

I have this extension: exten => 1322,1,Answer() exten => 1322,n,Set(CfMC_AMDValue="NotChecked") exten => 1322,n,GotoIf($["${CfMC_DoAMD}" != "Yes"]?NOAMD) exten => 1322,n,AMD() exten => 1322,n,Set(CfMC_AMDValue = ${AMDSTATUS}) exten => 1322,n(NOAMD),Wait(1) exten => 1322,n,UserEvent(E1322-1,${CfMC_ActionID}=${CHANNEL} & ${CfMC_AgentToUse}

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

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

I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches

I have a dialplan that looks like this: [dorecord] exten => _*99XX,1,Verbose(2,Doing custom record) exten => _*99XX,n,Answer() exten => _*99XX,n,Verbose(2,Doing custom record - before wait) exten => _*99XX,n,Wait(0.5) exten => _*99XX,n,Verbose(2,Doing custom record - before record) exten => _*99XX,n,Record(/tmp/prompt${EXTEN:3}.gsm) exten => _*99XX,n,Verbose(2,Doing custom

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for testing. In addition I register a zoiper SIP soft phone. For the Grandstream I have busylevel=1 in sip.conf. If I place a call from the GXP280 to zoiper and then put that call on hold from the zoiper side and then call GXP280's extension, asterisk indicates the phone is ringing. As the GXP280 is a single line phone it

I sent this email a few days ago but did not see any responses to it: > I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for > testing. In addition I register a zoiper SIP soft phone. > > For the Grandstream I have busylevel=1 in sip.conf. > > If I place a call from the GXP280 to zoiper and then put that call on hold > from the zoiper side and then

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

We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup. What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to

I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser

I have an i686 cpu and when compiling from source I get this error: touch /usr/src/dahdi-linux-2.2.0/drivers/dahdi/xpp/init_fxo_modes.verified Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32. o.cmd for /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o Anyone else seeing this?

Is there a way to inject sound from a sound file into an established call using AMI? I have an established call from which I can record either or both legs. I can additionally "spy" on the call. Is there any way I can play a sound file into the call and not loose the ability for the people to continue talking while listening to the sound file? -- Jim Dickenson mailto:dickenson at

[2009-05-27 02:06:16.294] WARNING[6971] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [2009-05-27 02:06:16.295] VERBOSE[6971] logger.c: [2009-05-27 02:06:16.295] == Primary D-Channel on span 1 up [2009-05-27 02:06:16.301] ERROR[6971] chan_dahdi.c: !! Got a UA, but i'm in state 7 I noticed the above error many days after this at around 2AM. This

Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can

I have been fighting with the ability to set the caller ID when I make outbound calls via a PRI line as well as via my SIP provider. The more I play around the less I understand. There is a setting in chan_dahdi.conf that seems to say do not pay attention to the CALLERPRES value and just allow the ID to be set. This setting is usecallingpres. If this is set to yes then the value of CALLERPRES