similar to: Another "Unable to create channel of type 'Zap' (cause 0)" error

Can someone tell me why this is not valid... [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@Local) exten => 2000,1,Macro(DialProxy,115551212) exten => 3000,1,Queue(testq||||45) while this is: [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@start) exten =>

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

That could be it. What I want to do is set a group of callers and have the event cause the phone to ring them in order. I will tie it to my IVR portion and thus I can make sure peole in sales get calls based on our hierarchy in the office. So if I am reading your example right the syntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause

Hi, I am running Asterisk on Fedora Core 3. I am trying to use DDD to debug Asterisk. However, when I attach the debugger to the Asterisk Process, the Asterisk CLI promt hangs. Does not give any output, and Asterisk stops processing calls... What could be wrong and what is the best way to debug Asterisk...? Appreciate pointers.. Thx a lot, J --------------------------------- Do you

Does anyone have an update patch file to get Spandsp installed? I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0 I installed spandsp-0.0.2 when runnig the patch I get patching file Makefile Hunk #1 FAILED at 41. Hunk #2 FAILED at 69. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej -------------- next part -------------- An HTML attachment was scrubbed...

Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes

Folks, I've been running asterisk successfully using the extensions.conf and voicemail.conf. Now that I've got asterisk happily looking up MySQL tables for the VM configuration, I decided to try out the contributed script /usr/src/asterisk/contrib/scripts/retrieve_extensions_from_mysql.pl I edited the script so that its output goes to a separate extensions_from_mysql.conf file. The

OK. I now have call recording working for both incoming and outgoing calls. Now I want to make those wavs into mp3. I could launch a script from cron that checks for new wavs and converts them. But that wouldn't be so elegant. Launching it from * on hangup would be nicer. How is it done? [outgoing] exten => _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten =>

Full_Name: susscorfa Version: 2.7.1 OS: ubuntu Submission from: (NULL) (129.125.177.31) Incorrect implementation of the grouping variable in the function glmer crashes R a small example: require(lme4); a<-data.frame(b=rpois(1000,10), c=gl(20,50), d=rnorm(1000,3), e=rnorm(1000,5), f=rnorm(1000,2)+5); glmer(b~d+f|c+(e), family=poisson, data=a) It crashes R on debian linux (2 independant

I'm trying to record a Meetme conference to disk, but the Monitor application doesn't seem to play nicely with Meetme. In extensions.conf, I have this: exten => 1000,1,Answer exten => 1000,2,Monitor exten => 1000,3,Meetme This starts up the monitoring OK, and it records the prompts that Meetme gives, but as soon as the user enters the conference, the -out WAV file stops

I have a voicemail application that users can listen to messages and leave messages. I am looking for a way to play a beep tone to a user when a new message is received when they are on the phone. Here is what I have come up with: in extensions.conf: [beepvoicemail] exten => 1000,1,answer() exten => 1000,2,NoCDR() exten => 1000,3,wait(2) exten => 1000,4,Set(TIMEOUT(absolute)=5)

I'm having some issues with Background() and Read() commands. See the example below. This is when I wait for Background to finish playing the sound file, before entering '12345#'. All works fine. hestia*CLI> -- Executing Answer("SIP/2944093-3366", "") in new stack -- Executing Wait("SIP/2944093-3366", "1") in new stack --

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

Hi all, I'm unable to put app_festival to work. I successfully patched, installed and tested festival (interactive logon and telnet to server port) which seems to work without problems. But when I test it in asterisk I got the following trace in console: -- Executing Answer("SIP/bsenicar-850b", "") in new stack -- Executing

I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message: Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22' -- Got SIP response 404 "Not Found"

Hello, Has anyone noticed that if you don't pick up a BT101 phone in 60 seconds it stops ringing and acts like it was never called ? Or is it just something I missed ? If it matters for something I have call waiting enabled on the phone. Product Model: BT100 Software Version: Program--1.0.5.11 Bootloader--1.0.0.18 HTML--1.0.0.37 VOC--1.0.0.6 Custom Ring Tone:

Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been connected for few minutes) when an incoming call comes in. The log reads the following when this happens: [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963, having received INVAL [Nov 17 15:25:04]

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

Hi all. I'm a very computer literate person, but an a bit of a noob when it comes to linux etc. I've a got a brand new PC. I've stuck in a TDM400P, with one FXO port. I've installed Fedora Core 3. Everything good so far. Now for Asterisk. GO to a shell then... md src cd src I'm now at /home/alan/src. Type the appropriate cvs commands (as per asterisk.org), and now i have