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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, I am hoping that someone can assist me with a work around. I have an IVR system that I am attempting to connect asterisk to, however the IVR was written Some time ago, and requires tones to be approximately 2 seconds in length. I am using SIP Phones, Is there any way to adjust the Length of the Tones that asterisk sends out the audio path? Are there any phones that would allow

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

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)

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 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

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

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> | | | | | |

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, I have my asterisk box setup to be an ntp server, and my zultys 4X4 phone is able to get the time, however I must first select the TimeZone Offset and then it will use the time setting from my server. This is a hassle because every time the phone reboots the user must answer this question and as you can imagine End users do not know what to do and since the phone is not

Good Day List, I have several different SIP Phones running on my asterisk box. I have recently purchased a snom 190 and a snom 220. They work great, EXCEPT I have to have the volume turned all the way up in order to hear the conversation on the other end. All the other phones Zultys 4x5, 4x4, 2x2, Cisco 7960, Wooksung 2000... all behave fine and volume is set to middle of the range and I

Good Day list, I am having some issues with my card in that it wants to share IRQ's with everything else in my box. I am running WhiteBox Linux 3.0 Is there a way to tell linux to assign a specific IRQ to a card. (unfortunately my MB does not have feature of assigning IRQ to slot) Further, I know this is silly to even ask but I

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

Anyone know if the INTEL/Dialogic announcement will become available to us who do not use the asterisk BE? Just curious as I have used those cards in the past and they are VERY STABLE and VERY dependable with excellent quality. ~ron

Anyone know if it is possible to control how aggressively the "Aggressive" mode behaves. Meaning, is it possible to dial back the aggressive mode to have a happy medium between Regular and the Aggressive defaults. I have a situation where Normal echo cancellation is not quite enough, however when I turn on aggressive mode We are attacking it to hard and I am unhappy with the walkie

Good Day list I am looking to buy a few Wooksung Video phones to try with my asterisk box.... http://www.wooksung.com/eng/html/pro/pro_001.html has anyone had any experience using these with asterisk? Thanks Ron

It has been narrowed down to the CONTEXT the call was originally located in. I need to figure a way within the parking application to set the extension I want the call to ring back to. Specifically I want to set it to always ring back to the phone that parked it. Example, below .. When the call rings only a specific phone http://pastebin.ca/2452 the context it originates from aa_1,2204,1

Ok I have read every line in the wiki for call parking. HELP PLEASE I need to perform a simple call parking that upon timeout will ALWAYS rings back to the extension that actually parked the call. Am I just crazy or is this not possible. >It has been narrowed down to the CONTEXT the call was originally located in. >I need to figure a way within the parking application to set the

Good Day All, I am experiencing some weirdness using the E&M channel and hope you can offer a little assistance with the problem I am having. 1) call comes into channel 25 (Second Span first channel of a Digium Quad PRI from SBC-PRI) 2) Call is sent to channel 1 (First Span first channel on the Digium Quad PRI connecting an ADTRAN via E&M Feature Group D) 3) Between rings one and two