similar to: Group dial, first phone cannot pickup call. Cisco 7905 hangs.

I installed Asterisk 1.0.9 in a Freebsd 5.4 ( with no zaptel card); I have 2 zoom x5v and works great ( in extensions 123 and 321 ) but I was trying to test cmd Playback, MusicOnHold, MP3Player but when I call to extension 100 I don't hear the sound ( mp3 or gsm that I put) , I only hear noise If I leave a message in a mailbox the same, all the record is noise --------- extensionns.conf

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

Hi I have two small question, if you can help me ;=) Problems with Answer+Music my extension: [Cal-In] exten => _811XXXX20,1,Goto(C-Internal,100,1) exten => _811XXXX21,1,Goto(C-Internal,200,1) [C-Phibee] exten => 100,1,Ringing exten => 100,2,Wait,1 exten => 100,3,Answer exten => 100,4,Dial(SIP/201&SIP/200,30) exten => 100,5,Hangup exten =>

All, I have been beating my head against a wall trying to figure out how I would implement a separate moderator code and participant code for the same conference using meetme, the deal is I dont want the participants to be able to join until the moderator is in the conference. Is it possible to do this using the apps as they are , or is their a way to use an Agi script, is that the only way?

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

Not sure this is known behaviour but it seems that if want to set the environment to, for example, test, and you want to use Active Record you have to both explicitly set the RAILS_ENV and the BackgrounDRb environment. So, if you have a config file backgroundrb_test.yml (as well as the default) and set the environment to test in that, this is what *seems* to happen: $

Dear users, I'll appreciate your help with this (hopefully) simple problem. I have a model object which was fitted to inputs X1, X2, X3. Now, I'd like to use this object to make predictions on a new data set where only X1 and X2 are available (just use the estimated coefficients for these variables in making predictions and ignoring the coefficient on X3). Here's my attempt but, of

Hola, Can someone please point me to the right direction on how-to set up proper routing on PPPoE connection and multiple external IP''s. Thank you kindly. ~Andrew Nady.

This is a follow/find me script that I can't quite get to work, asterisk wont save forward/${calleridnum} to AstDB... any comments or thoughts on how to make this work or change it to work differently are appreciated. The voice prompts to go with all playback/background extensions are commented appropriately. I hope this code is of use to some of you and any help with a perfected

Hi, Is there a way to detect if the caller will be entering an agentless queue? I'd like to be able to redirect any caller who tried to join a queue with no logged in agents, to be redirected to the groups voicemail. Is this possible? I know I could create a menu and an announcement for voicemail (should the user wish to drop from the queue); but they wouldn't know no one was taking

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for

Hello, I've reccorded a voice message for the IVR. (.wav, 16 bits, 8kHz) The file is /var/lib/asterisk/sound/11ivrrecording.wav. When asterisk (1.2.5) starts this file i can't hear it on my phone. Here is the log : Apr 6 17:00:16 VERBOSE[845] logger.c: -- Executing SetCallerID("SIP/11-97b9", ""Patrice" <11>") in new stack Apr 6 17:00:16

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

Hello, I wrote a wrapper for symbols() that produces a bivariate bubble plot, for use when plot(x,y) hides multiple occurrences of the same x,y combination (e.g. if x,y are integers). Circle area ~ counts per bin, and circle size is controlled by 'scale'. Question: how can I automatically make the smallest circle the same size as a standard plot character, rather than having to

Here's what I'm looking to do.... exten => 10,1,Dial(SIP/1000&SIP/1001,15,wW) exten => 10,2,Voicemail(u1000) This should ring both phones and they should keep ringing for the alloted time before moving on. However, it appears that if one of the phones is Busy, it returns with a busy and moves on without really ringing the second phone. Short of checking if the channels are

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

Hi, I have a simple dialplan. [test] exten => _X.,1,Dial(SIP/1000&SIP/1002) I have registered user whose context is test. Now I am dialing any number, so it will enter into test context. It will dial 1000 & 1002 both. Both keeps ringing. Now the problem is, when any of them answer, another one keeps ringing for 10 to 15 sec. Please help me , what's wrong here. Thanks, Krunal

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

Is there a way to strip the normal features out of FollowMe (call acceptance, etc), and just set followme up to to blind transfer any call to an extension's associated cell number if it is not answered on the extension after 4 rings? Users want followme calls to wind up in their cellphone voicemail and I'm having some issues with ring/answer timing and Asterisk wants to pull the call

Hi All Curious will this work Std. PSTN line ---x------ X100p | ------ Fax Machine Using a standard "home phone" pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten => s,1,Wait(1) exten => s,2,IfFax continue to ring, so that the Fax Machine gets it exten