similar to: 2 step dialing

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Hello, I am trying to get MWI working after integrating Asterisk with CCM.I have followed the instructions in http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Voicemail+IntegrationMy problem is that I don't see externnotify's script being called at all in the logs, and not sure if I miss something here! In Voicemail general I addedpollmailboxes =

Hello, I'm experiencing a weird problem when using the VoiceMailMain application. If I use the application after dialing a Local channel, there's strange beep just after asterisk answers the call and before the first locution. The extensions.conf I'm using is: Ruido extra?o al llamar a la aplicaci?n VoiceMailMain [default] exten => _X.,1,Dial(Local/${EXTEN}@test) [test] exten

Good afternoon list. I am experiencing a problem with the CDR and callfiles. What is happening is this: When generating a call with a callfile, everything works perfectly, but the CDR is recorded in the table when they answer the call destination. The field disposition is being recorded correctly, but the duration field is marked with the ring time and billsec is marked with 0. This just happens

Hello all, We would like to "inform" the caller of the reason for a failed call. For example, when we get a "486 Busy Here", the system accepts it and in the CLI we see "Everyone is busy/congested at this time". Can we use this data to play an announcement to the caller? Thank you in advance for your help. Michael -------------- next part -------------- An HTML

Hi everyone, I have a file in which the dates are subscribed as for instance: 20101020. This is 20th Octobre 2010. My problem is that R won't except this as a date, since there is no sign to seperate the Year, Month and Day and that it will only see it as an origin, which it is not. Does anyone know what to do about this. Greetings, Britt -- View this message in context:

Hiyall. Just wondering how people test your emergency dialing in the UK. Obviously you need to dial the 999 for emergency services, but am a bit unsure if this would go down too well with the operator with a 'sorry just testing' call. (you do all /test/ your emergency dialing dont you!?:-) ) As another thing, what is the correct method when using least cost routing... If you have a

Hi, I am using asterisk via AGI and want to be able to record a call. The scenario is: 1. A call comes in 2. The call is redirected to a mobile number via a local extension and ChannelRedirect 3. The local extension looks like something this: exten => _X.,1,Dial(SIP/${EXTEN},60,?) exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..) I have looked through all arguments of Dial

After giving command "yum update openoffice" the output is: Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: centos.aol.in * updates: ftp.oss.eznetsols.org * addons: centos.aol.in * extras: centos.aol.in Setting up Update Process No Packages marked for Update -------------- next part -------------- An HTML attachment was scrubbed... URL:

Preparing to use * for a 'real' installation shortly. Meanwhile, I've got a single port clone thing, 00:06.0 Communication controller: Motorola Wildcard X100P working to answer my landline and send calls to my laptop or voicemail. Sweet! Trying to "call out" from linphone, I set up this: exten => _X.,1,Dial(DAHDI/1,${EXTEN}) Both SIP client and this extension are in

Should call celt_inner_prod(). --- celt/bands.c | 7 ++++--- celt/bands.h | 2 +- celt/celt_encoder.c | 6 +++--- celt/pitch.c | 2 +- src/opus_multistream_encoder.c | 2 +- 5 files changed, 10 insertions(+), 9 deletions(-) diff --git a/celt/bands.c b/celt/bands.c index bbe8a4c..1ab24aa 100644 --- a/celt/bands.c +++ b/celt/bands.c

Hi, My costumers are logged in on my Asterisk PBX through XLite Softphone (SIP). My server is connected to PSTN. Problem is when SIP phone calls ordinary phone via dahdi I get DAHDI/1-1 ANSWERED SIP/number-number and billsec field from cdr is start counting. Is it normal behavior ? Can I change that ? So channel gets in ANSWERED state and billsec starts as soon as line starts to ring even if no

Hi, I am trying to setup some LCR functions on my Asterisk box and have a cheap call provider that uses various different numbers for landlines and cell phone numbers in various countrys.. I am finding it difficult to find the various codes.. eg. UK Landline - +44[12]. UK Cell - +44[7]. SA Landline - +27[1-6]. SA Cell - +27[78]. Please send me your country's dialing rules similar to how I

I have an issue where a particular dialplan works but another doesn't. I'm not sure why. To me they look identical and it has me stumped. This works: [to-test] exten => _X., 1, SetCallerPres(allowed) exten => _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb) exten => _X., 3, Ringing exten => _X., 4, Dial(SIP/9330 at a-test,20,ro) exten => _X., 5,

Hi, My dialplan looks like this: [from-external] 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() include => test [test] Exten => 8282,1,Noop(--- WE GOT TO

Hi all, I have the following context in extensions.conf: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup exten => _X.,21,Playback(AR_GetGiveToID) exten => _X.,22,Wait(2) exten => _X.,23,Record(/tmp/asterisk-recording:ulaw,,5) exten => _X.,24,Wait(2) exten =>

Everybody, I have a fall though context that, among other things, tests to see if someone it trying to pick up a non-existent parked call (Defined from 90 to 99). I have the following: [not-in-service] exten => _X.,1,Wait(1) exten => _X.,n,ResetCDR() ; ************************************************** ; Check to see if the mis-dialed number was a parking ; slot. If so, jump to the

Dear All, I have the below context defined in extensions.con: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup exten => _X.,21,Wait(2) exten => _X.,22,Record(/tmp/asterisk-recording:ulaw) exten => _X.,23,Wait(2) exten => _X.,24,Playback(/tmp/asterisk-recording) exten =>

Hi team, I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls. Please suggest me the solution. [TB] exten => _X.,1,Wait(${INCOMING_WAIT}) exten =>_X.,2,Verbose(TB) exten =>_X.,3,Answer() exten => _X.,4,Set(mainLoop=0) exten =>