search for: 1234,1

I'm running 1.2.9.1, and I can't get caller id dialplan matching to work. When calling from 9220370 to 1234, the following does not match. exten => 9220370/1234,1,NoOp(${CALLERIDNUM}) exten => 9220370/1234,2,Answer exten => 9220370/1234,3,Playback(tt-weasels) However, when calling from 9220370 to 1234, this DOES match. exten => 1234,1,NoOp(${CALLERIDNUM}) exten => 1234,2,Answer exten =&...

Hello, I am trying to run festival, it is running but I am getting this when I run tts_ping.agi WARNING[278546]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Enter the eye-p address you wish to ping. WARNING[278546]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... WARNING[278546]: File app_festival.c, Line 400 (festival_exec):

...these routines are also coming in via SIP. Voice works fine on the channels, once answered. Does anyone know if there is something wrong with the "r" modifier on SIP Dial application calls, or have you had experience doing this a better way? None of these methods work: exten => 1234,1,Dial(${PHONE1},25,r) exten => 1234,1,Answer exten => 1234,2,Dial(${PHONE1},25,r) exten => 1234,1,Answer exten => 1234,2,Ringing exten => 1234,3,Dial(${PHONE1},25,r) exten => 1234,1,Ringing exten => 1234,2,Dial(${PHONE1},25,r) exten => 1234,1,Ringing exten => 123...

...ample of what is currently happening for an attended transfer when DTMF sequences are allowed Call from outside: [from-pstn] exten => _X.,1,Dial(SIP/....,,...t...) ; fine -- only callee can transfer Attended transfer (Asterisk uses a Local channel to connect): [from-internal] exten => 1234,1,Dial(Local/....,,Tt...) ; bad -- from here on the outside caller can do whatever he wants in this case it should be exten => 1234,1,Dial(Local/....,,t...) Call from inside: [from-internal] ; e.g. exten => _X.,1,Dial(DAHDI/r2/${FILTER(0-9,${EXTEN})},,...T...) ; fine -- only caller c...

Hi All, I'm having trouble setting up a queue: I'm using AgentCallBackLogin to login in the queue, but: 1 - When an agent answer the call and another call arrive his phone rings again. 2 - When no there are no one answer the queue the system goes to voicemail of agent 1234 I'm using asterisk-1.2.0-beta1. My configuration is below, Any ideas? Many thanks, Joao Antunes ;extensions.conf [demo] exten => 1005,1, Answer exten => 1005,2, AgentCallBackLogin(${CALLERIDNUM}|${CALLERIDNUM}@agentes) exten => 1005,3, AddQueueMember(test1|local/${CALLERIDNUM}@age...

...directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty application=http://pubint.ic.llnwd.net/stream/pubint_wnpr and this is how I am testing it: exten => 1234,1,Answer exten => 1234,2,SetMusiconHold(stream2) exten => 1234,3,WaitmusiconHold(60) exten => 1234,4,Hangup and this is the console output I get when I dial 1234: Asterisk Ready. *CLI> -- Executing Answer("SIP/3250072-ed28", "") in new stack -- Executing Se...

hi folks. is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? suppose i want to do something like this: exten => 1234,1,dial(SIP/1234) exten => 1234,2,<do something> but when the dial command hangs up normally, line 2 won't get executed. -- Edwin Lam <edwin@officegeneral.com> Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/look...

I'm trying to set a dialing rule in my dialplan. As a part of it, from my point of view, this works wrong priorityjumping=no [test_context] exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag exten => 1234,2,Playback(digits/2) exten => 1234,3,Playback(digits/3) exten => 1234,102,Playback(digits/4) In this case, if I dial the extension, and it's answered, the caller can hear "Two, Three) and the ca...

It appears that my X100P card is only answering after two rings. Ideally, I'd like it to answer on the first ring. Here is the incoming section of my extensions.conf file: [incoming] exten => s,1,Answer exten => s,2,BackGround(demo-congrats) ; Play a congratulatory message exten => 1234,1,Goto(jgunther,1234,1) exten => 4321,1,Goto(mgunther,4321,1) exten => s,3,Goto(mgunther,4321,1) exten => s,4,Hangup include => default Any ideas on how to correct this issue? Thanks. Jeff

Hi there, I am configuring a small voicemail server and I am facing the following problem. Executing this command: exten => 1234,1,VoiceMail(${NUMBER}@test) When a user does not have a customized temporary greeting vm-intro message is played asking for the message to the user but when the user has already a temporary greeting both the temporary greeting and vm-intro are played. Basically what I would like to do is to avoid...

...gest it), but has anyone had any problems with Asterisk accepting a PIN number for a conference room. At this point in time I have established the conference definition in the meetme.conf file as well as specifying the appropriate lines in the extensions.conf file. meetme.conf file: conf => 1234,1574 extensions.conf file: exten => 1234,1,MeetMe(5678) OR exten => 1234,1,MeetMe,5678 In both cases I can get the SIP client "SJPhone" to actually dial the conference number and I hear the operator say, enter the pin number followed by a "pound sign". After entering...

...e one can't register to the one on the inside, since it can't be reached on the private network. Now to my problem: * How do I dial from outside to the inside over the existing IAX connection? When I dial from the outside to the inside by using the registred loginname like exten => 1234,1,Dial(IAX/loginname/12345) The outside server seems to dial the one on the inside, but I see nothing on the inside. The log on the outside mysteriously enough claims it can't authenticate to the inside server - but how do I authenticate, all authentication in IAX is based on hostname or IP nu...

...'s me, the CPE device, Asterisk, my dialplan code or a combination of all. I am running Asterisk 1.6.0.21, FreePBX 2.6, and the latest version of DAHDI. Here's the dialplan logic I am trying to execute, this is in /etc/asterisk/extensions_custom.conf: [from-internal-custom] exten => 1234,1,Dial(DAHDI/2) ; DAHDI channel 2, FXO exten => 1234,n,SendDTMF(2) ; I expect to send DTMF digit 2 after the channel answers? exten => 1234,n,Wait(1) ; I added a wait statement exten => 1234,n,Flash() ; Send the hookflash After the hookflash is...

Hi there, whenever I use a macro to dial out I see only "s" recorded in the dst field of the CDR. Is there anyway to get around that problem except for not using a macro? Example: [default] exten => 1234,1,macro(dial-out) [macro-dial-out] exten => s,1,Dial(SIP/test,30,r) Now, I can probably catch the "unavailable" and the "busy" case using the MACRO_OFFSET variable, but what do I do if the caller hangs up himself? The Dial option 'g' won't help, and I always e...

...w stack == Spawn extension (from-external, 8282, 5) exited non-zero on 'SIP/192.168.200.32-00000000' It seems that the 8282 in test is being ignored now if I comment out all lines starting with " Exten => _X.,1" so my dialplan looks like this: [from-external] Exten => 1234,1,Noop() ;Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) ;Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) ;Exten => _X.,n,Noop(CALLED NUMBER: ${EXTEN}) ;Exten => _X.,n,Ringing ;Exten => _X.,n,WaitExten(15) ;Exten => _X.,n,Congestion() ;Exten => _X.,n,Hangup() in...

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 ${DIALSTATUS}) Why, if 5551234 is busy, is DIALSTATUS set to CHANUNAVAIL? Should it not be BUSY? And now with IAX2 (I am...

I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=>#9 in my features.conf, I have Dial(....,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is

...ist - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Streaming Music On Hold > > > Try this: > > musiconhold.conf: > > [stream2] > mode=mp3 > directory=http://pubint.ic.llnwd.net/stream/pubint_wnpr > > > extensions.conf: > > exten => 1234,1,Answer > exten => 1234,2,MusicOnHold(stream2) > exten => 1234,3,Hangup > > > On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote: > > Ok, I'm tearing my hair out trying to get Asterisk moh streaming to > work. After several hours jerking around with iceca...

..._X.,n,WaitExten(15) > Exten => _X.,n,Congestion() > Exten => _X.,n,Hangup() > > [test] > > Exten => 8282,1,Noop(--- WE GOT TO CONTEXT TEST!--------- ) > > -- > > Doug, I tried that as well. Even with my dialplan looking like this: [from-external] Exten => 1234,1,Noop() include => test Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) Exten => _X.,n,Noop(CALLED NUMBER: ${EXTEN}) Exten => _X.,n,Ringing Exten => _X.,n,WaitExten(15) Exten => _X.,n,Congestion() Exten => _X.,n,H...

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current