similar to: Problems with CISCO, SIP and Asterisk

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)

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

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

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

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 --

Before I open a bug I'll ask again if anyone else is having trouble with receiving MWI on SIP devices in 1.4. My configuration was working fine in 1.2 but as soon as I change to any build of 1.4 I don't get notification on any of several SIP devices. I can post my configuration but since it was working I can only assume it would break if something in voicemail.conf has changed or

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

I have set up an auto attendant and all is working but I am bothered by a long pause when switching between menus. This pause is between 5 and 7 seconds and is quite annoying. Is there anyway to address this. One other thing I find interesting is that when I move from the main menu to the sub menu the delay is there but when I move from the sub menu to the main menu the delay is not there.

I'm trying to setup a VERY simple mainmenu ivr but can't seem to get the tones to be recognized during the background( ) the playback and background files play, but asterisk doesn't do anything when I start pushing keys - I've tried it from softphones and pstn line phones Can anyone tell me what I'm doing wrong? Required contexts Exentions.conf below [from-broadvoice]

Here is some dialog from the Console: -- Starting simple switch on 'Zap/13-1' Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack -- Goto (mainmenu,s,1) -- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker -support") in new stack

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, I'm now the proud owner of an X100P and am struggling to set up a CVS-compiled Asterisk to do my bidding. I checked zaptel/zapata/asterisk out today and pretty much did a straight make install on all packages. So far the only consistent trick I can make it perform is calling from one SIP phone to another. Could I get a bit of

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also 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]

Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use 555@mainmenu and 666@mainmenu) for the Icon extensions. IAX softphone seems to work

I have the latest Asterisk installed however the analog line card I was using doesn't work in the new server, that is ok. Instead what I want to do is allocate one of my numbers to be the lead number for the Asterisk system. User who call this will have their call delivered to our site specific version of the Asterisk mainmenu via an IP connection. I already have SIP connections coming into