similar to: DTMF Tone Lengths

Good Day list, Anyone know if there is a way to have the AgentCallBackLogin function play a voice file after the agent picks up the phone? If this is not an available feature, any ideas on the difficulty in making this feature? Example: Extensions.conf exten?=>?700,1,AgentCallbackLogin(${CALLERIDNUM}|?AnnounceCAllQue-TechSu pport?); ....... exten => s,6,Queue(queue1)

Good Day again list, Encountered another problem in the ACD queue... If I use the ADDQueueMember to dynamically add members as foolows, exten => 403,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM}) lets assume I called extension 403 from my extension 2204. then a caller (extension 2203) enters into the techsupport queue I am able to receive the support call on my phone (extension 2204

Good Day list, My head is pounding from google overload. Does anyone know if there is any way to park a call and specify the ("context", "Extension", and Priority) of where to call back to if the call times out and is parked too long? ParkAndAnnounce does this feature, however I do not like that it has to call you back to announce the location. I want my cake and eat it

Good day list, I am feeling extra stupid this Monday morning and am hoping someone can come to the rescue. I am trying to use the ztmonitor utility on my wildfire 4 FXO card. and have read the following from the wiki. *********Wiki start******** If you set this to yes, use ztmonitor to adjust the rxgain and txgain. Ztmonitor isn't installed by default; but it is included with the Zaptel

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... -- Best regards, Duane http://www.cacert.org - Free Security Certificates

Good Day list, I have a friend who is interested in implementing an asterisk implementation at his offices. The configuration would consist of the following Site A ---- Asterisk Box With 12 incoming lines and 15 phones Extensions 101-115 Site B ---- Asterisk Box With 4 incoming lines and 7 phones Extensions 201-207 Site C ---- Asterisk Box With 4 incoming lines and 6 phones

I have been getting quite a bit of PRI Resets using my Quad PRI Digium card. Prior to the resets I am getting similar notices to the following chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 Telco claims the PRI's are fine on their end and that it is my unit. Is this timing? (google somewhat leads to this) I am running 1.08 asterisk zaptel

Good Day list, Need assistance determining the best place to read up on whether Asterisk can help me out. I have a situation where I need to do the following <PRI from Telco> ------- <Analog Channel Bank>------------<Proprietary Box> | | | | | | <PRI Port 1 of Digium Quad T1> <PRI Port 2 of Digium Quad T1> | | | | | |

Please point me in the right direction as to where to increase the length of time I am sending DTMF tones on my T100P. I can't remember where/how I set it last time, and I need to do a fresh install. I'd sure appreciate it :) Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977

Is there a way to get my X100P card to dial "slower" on the line? Mine seems to dial the digits too short/fast for the switch to catch all of the digits and roughly 25% of outbound calls fail to complete. I've monitored on the line with a test set and the audio sounds clean, and I hear the X100P send out digits really fast, but sometimes the switch seems to time out and issue

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

Hi, We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do

Hello, I'm having trouble working out how to send DTMF tones to an external IVR. My system has an analog phone connected to a TDM400P card, a SIP software phone (Zultys LIPZ4) and is connected to a BRI in Australia with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched with the ISDN audio patch from Traverse (which allows the card to do voice). DTMF works fine between

I have 1.2.12.1 Voicepulse using IAX I get about 30-40% issues with not having the DTMF tones work. I have 3 questions #1. Voicepulse says they are sending them, Is there some setting I can adjust to make sure my end is working? #2. I have set the Dialplan to play a sound "Operator" then go to a SIP phone. I would like it to write Caller ID Time .... to a file I can read and find

Hello, I'm a senior statistician at Affinnova, a market research firm outside of Boston, MA. We are looking to hire a strong R programmer. Here is the description of the job: Come join a growing company in the market innovation field. Affinnova is changing innovation and product development by applying Interactive Genetic Algorithms to bridge the chasm between the worlds of the designer, the

Good Day list, Does anyone know if it is possible to setup asterisk such that it passes DTMF Tones through from One channel to the next transparently. I have a situation where asterisk is answering the phone on Channel 1 (first channel of a PRI) and then bridges this call to Channel 25 (first channel of T1 connecting in a channel bank). I need to have asterisk NOT do anything to the

Hi, I am attempting to create a intercom buzzer system using asterisk as a back end. Most is figured out except the actual action of buzzing the door. I need to detect whether a DTMF key was pressed by the the called party (the resident). Is this possible to do using just a dialplan? I can't see any options on the Dial command that would lead to this, am I looking in the wrong place? I looked

Hello all, I am using voicepulse DID's to receive calls via IAX to and asterisk IVR dial plan I have put together. The problem is after 3-5mins the system cant pickup the DTMF tones I am sending... I have tried different telephones... It just repeats menu options over and over.... I have to call back and then it works again for another few mins... Any ideas... iax.conf? issue? Thanks, J.C.

Dear to all someone know how is possible to have a DTMF tone like "C" AKA Alpha Tone (connect tone) to the caller? Thanks in advance Dimitri