similar to: IVR sound after dial sip

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Eric Wrote: ----------- The trick is not to use options you don't understand. "show application dial" will show you what the t and T options are for. Most people use the transfer feature of their phone, rather than using the T/t hack on the Dial line. Sounds like you are using CVS-HEAD and so will have to configure stuff in /etc/asterisk/features.conf. /Snip/ Eric, Thanks for

Hi all, I am trying to understand how I can get a simple IVR scenario to work properly (having already removed most of my hair...). The basic requirement is as follows: * Caller arrives at our main number * Caller is greeted and then told they can enter an extension number, if known, or wait and their call will be connected to an available rep. * The IVR then dials a group of extensions (if

I have asterisk 1.0.5 installed via apt on a debian system. It's a custom distrobution called Voyage Linux that runs from a flash card and I have a hard drive installed with mysql installed on it as well as apache. I have been though the AMP install guide (asterisk management portal) and in the interface it has a place for me to record new IVR menus. I have to dial *77 to begin recording

There are no cordless devices in this situation. :) Jerry Geis wrote: >/ I have a normal setup of calls coming in on analog lines (4 of them) />/ coming into />/ an old KX1232 pbx. I have those lines forwarded to a T1 card in the />/ KX1232 />/ going over to my T1 card, it then uses IAX going over to my REAL />/ asterisk pbx. />/ />/ (these steps are there for testing

I have a normal setup of calls coming in on analog lines (4 of them) coming into an old KX1232 pbx. I have those lines forwarded to a T1 card in the KX1232 going over to my T1 card, it then uses IAX going over to my REAL asterisk pbx. (these steps are there for testing and other items) Anyway after a 10 minute call asterisk gives a faint beep, beep, beep and then hangs up the line. Is there

I have a list of CLIDs prefixes that I want to use in a context. Basically, I want to do this but the list of prefix numbers is much longer. List of prefixes (556,557,557,989.....) [custom-inbound] exten => _556,1,answer exten => _556,n,playback(beep) exten => _557,1,answer exten => _557,n,playback(beep) exten => _558,1,answer exten => _558,n,playback(beep) exten =>

Hi fellows.. I have 2 asterisk servers in which the following line exten => _09049.,111,SetVar(LIMIT_PLAYAUDIO_CALLER=YES) exten => _09049.,112,SetVar(LIMIT_WARNING_FILE=beep) exten => _09049.,113,Dial(${TYPE}${DESTINO}|30|L(30000:10000)) works OK on my Asterisk 1.2.9, it plays the beep 10 seconds before the end of the call. doesn't work on my Asterisk 1.2.13, it hungs 10

Hi All; I installed Asterisk 1.8.11-cert1, and it look like the default is ulaw for the sound files. How I can fix this? Athough file beep.gsm is existed under path (/var/lib/asterisk/sounds/en), but when I used the Record function, it gave me the following (so I am sure there is something that can let asterisk accept beep.gsm), what could be? [May 5 00:44:16] WARNING[2262]: file.c:663

Hi, Here is my problem. I am trying to get the Status of the call if the user picked up the phone or not. It is coming as empty. Please help. Here is my extensions_additional.conf file code: [multi-dir-callback] include => multi-dir-callback-custom exten => _X.,1,Answer exten => _X.,n,Playback(beep) exten =>

I'm having a problem with sound output to the console. My basic dial plan is as follows: exten => _1NXXNXXXXXX,1,Dial(IAX2/####@voxee/${EXTEN},30,A(beep)) exten => _1NXXNXXXXXX,2,Playtones(info) exten => _1NXXNXXXXXX,3,Hangup I get the following output in the console: ___*CLI> dial 1#######@voxee -- Executing Dial("ALSA/default",

Hey hey... I'd like to create a new feature code in asterisk so when a user dials... say... *00, it would then call some other extensions and play a sound file to them. So far, this is what I have... [testing-custom] exten => *00,1,Wait(1) exten => *00,2,Playback(beep) exten => *00,3,Playback(beep) exten => *00,4,AGI(festival-script.pl|I will now attempt the call) exten =>

This seems basic but something is missing..... I dial from my cell phone to my DID and enter the context in extensions.conf I am hoping to cascade through the plan and successfully automatically dial the 1444 number listed. But it fails. And, I dpon't know why? Should I removed the Hangup application? Syntax issue somewhere? I have a good SIP registration with the vendor, voipvoip.

I'm unable to get the Dial option 'g' to work with callback agents. The plan is to use it so that I can redirect a customer to a menu so they can rate the call they just had with the agent. However, when the agent hangs up the call does not continue in the dialplan. I login with the agent. Call joins the queue. The agent and call get connected. The agent hangs up and the call

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

Hi I'd like to allow users to transfer a call to a specific park number. This way, the receptionist can tranfer a call park for ext 100 at park number 7100 etc... It seems like this should be fairly simple using the Park(ext) app but it doesn't work for me. No matter what I extension I use, the system just picks the next available park number. I've simplified my dialplan for

Hi All, I have been working on a project where I need to record a call in Asterisk and then split the recording into multiple audio files based on a presence of particular sound (i.e. beep) in a recording. I know this is out of scope for Asterisk but I wanted to benefit from someone else's experience if it has been done earlier. I have googled a bit and seems that Audio fingerprint(

Does anyone have a more clear beep tone for the voicemail? The default one seems to cut off and give the feeling that there is a problem with the vm :) If no one has this available, I may try to create a new one. Thanks. -- respectfully, Joseph - ------=============