similar to: internal sounds of asterisk / freePBX

HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has

I have the following in the dialplan: [macro-systemrecording] exten => s,1,Goto(${ARG1},1) exten => dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav) exten => dorecord,n,Wait(1) exten => dorecord,n,Goto(confmenu,1) exten => docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording) exten => docheck,n,Wait(1) exten => docheck,n,Goto(confmenu,1) exten =>

Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg "Im-sorry&an-error-has-occured" and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some

Hello, I am trying to alter the outbound callerID for extensions within a context I have created. I wrote the following: exten => _9.,2,ExecIf($[$["${REALCALLERIDNUM}" = "360"] | $["$ {REALCALLERIDNUM}" = "670"]]|Set|CALLERID(num)=581560) exten => _9.,3,ExecIf($[$["${REALCALLERIDNUM}" = "361"] | $["$

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

I have a dialplan that looks like this: [dorecord] exten => _*99XX,1,Verbose(2,Doing custom record) exten => _*99XX,n,Answer() exten => _*99XX,n,Verbose(2,Doing custom record - before wait) exten => _*99XX,n,Wait(0.5) exten => _*99XX,n,Verbose(2,Doing custom record - before record) exten => _*99XX,n,Record(/tmp/prompt${EXTEN:3}.gsm) exten => _*99XX,n,Verbose(2,Doing custom

Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31

Hello everybody, and first of all sorry for my poor English. I'm having trouble with Quadbri installed on Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling to switched off or "out of coverage" cell phones. In this case I have to wait 40 seconds until Asterisk tell me that "all circuits are busy now" instead of receive cell phone

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP

Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.

Hi, I use: Trixbox-2.2.4 FreePBX-2.3.1.0 Asterisk-1.2.17 BRIstuffed-0.3.0-PRE-1y-e Zaptel-1.2.19 ..with two ISDN cards, often but occasionally the dial out failed but is possible to receive external call. My zapata.conf conf is: [trunkgroups] [channels] language=it context=from-pstn signalling=bri_cpe_ptmp rxwink=300 pridialplan=unknown prilocaldialplan=local switchtype=euroisdn

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Paso&view_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package "pear.php.net/db" Cannot initialize 'db' , invalid or missing package files Package "db" is not valid install

Trixbox 2.2 Beta with freePBX 2.2.0rc1 I have a setup that looks something like this in ASCII art: Teliax IAX Trunk ------+ | V Embarq PRI ----> Tandem switch ----> Ottawa Office Server------+ +--------------> Lima Office Server -----+|

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for