similar to: Dial Plan Logic Problem

Hi all, Having fixed my problems with the call waiting ringing on the GS phones, I needed to extend that with a campon facility (available on some legacy systems - sort of callwaiting without phone ringing). I've managed to implement that by adding/modifying app_queue.c. Basically, when calling the SIP phone, and if busy, I can camp the call onto that extension, with MOH, allowing the caller

Hi, A potential customer would like to be able to do this: If a call comes in for an employee who is on the phone, allow the front-desk to push the caller in a queue directly to the employee. Now, this is easily done by using queues, but I am curious: What is the performance impact on a system if _every_ employee (phone) has their own queue. How scalable is that in comparison to

I am attempting to get fax detection to work. I am using a NETjet-s card under ISDN4Linux. Asterisk does not seem to be detecting the fax tone. I have tried following as a test: [MainMenu] exten => s,1,Answer exten => s,2,DigitTimeout(3) exten => s,3,ResponseTimeout(5) exten => s,4,Background(Welcome) exten => s,5,Background(MainMenu) exten => fax,1,Dial(Zap/1,,d) [FaxTest]

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

I think I am the first to use the $135 Avaya 4602 SIP phone, but I need some support from the community to fix one problem I have with it. The phone stops working after about 20-30mins if I have mailbox=context in Asterisk; when I do have mailbox=contect in asterisk the sip debug returns "481 extension does not exist." Anyone willing to help me figure out why? I.E. Is it an Asterisk

Hi, I am trying to implement the callback feature of our old phone system. This feature may go by a different name in asterisk? It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B

I'm developing asterisk to work in a small Call Center for a Mobile Communications Provider. I'm looking for references for Agent Logins and Huntgroups. If any of you have already configured Asterisk for such a Task and would be willing to let me take a look at your config files could you please email them to me at mailto:jhelmich@bluesky.as Thanks, Jason Helmich MIS, Blue Sky

Hi all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... i have run basic testing of asterisk like as shown in website http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp when i have copied sip.conf and extensions.conf from the site and run the client and server i

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Hi list I encountered problem in using Background command. Below is my extensions.conf. [mainmenu] exten => 4,1,Wait(1) exten => 4,2,Background(thank-you-for-calling) exten => 4,3,Goto(n01|s|1) [n01] exten => s,1,NoOp(${CONTEXT}) exten => s,2,Background(thank-you-cooperation|m) exten => s,3,WaitExten() exten => s,4,Playback(digits/pound) exten => 1,1,Playback(digits/1)

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

Hi Guys. I've had a lay off from Asterisk for 12 months but I am starting to look into it again. I am not very Linux savvy and found it hard going the last time. I've started playing with it in the last 3 weeks and I have to admit to making more head way this time. The first problem I'm stuck on and I cant find a solution to is that sound files that I have recorded (be it by

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

Hello People, I'm newbie in * 1.0.1, running a Linux 2.6.7 in a Debian Sarge, and this is my situation: +------------+ +-------------+ | Sip Server |-------------|CISCO PSTN GW| +------------+ +-------------+ \ || \ || \ +----------+ || | Asterisk |=========

If I have a global variable in my dialplan and I change it, does that change immediately take affect for all calls that are active? Here is my situation. The company I work for has two office groups that share a building. The two offices are separate companies but support one another and want to be able to transfer calls as if they were all on the same phone system. Each company has 4

Hello, I am trying to walk through the following document: http://homepages.lu/charlesschwartz/radius/freeRadius_AD_tutorial.pdf in order to authenticate Cisco router and switch logins against FreeRadius/Active Directory. Using the HowTo, I have successfully joined a FC2 box to our Windows 2003 AD for testing purposes. I have also successfully used the manual ntlm_auth command to authenticate

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Hi , i worked with h324m gateway for 3g video calling .It? configured successfully . my code in extensions.conf is [from-zaptel] exten => _X.,1,h324m_gw(0 at mainmenu) exten=>_X.,n,Hangup [mainmenu] exten => 0,1,h324m_gw_answer() exten => 0,2,mp4play(/tmp/menu/menu.mp4,'n(1)') when i make a video call (either sip or through pri) , asterisk cli shows the following error --