similar to: mysql-vm-routines does not use the context properly

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Hello, I have setup a test server with asterisk/AMP and have several 7960's connected to it. The asterisk server has a public ip and all the 7960's are behind nat'd routers. When I try to call from extension to extension I get directed straight to voicemail. I do not have any cards installed and instead direct everything to an Ondo server. I have been told it's not an AMP

I have quite an old version of Chameleon Mail, currently the prompts played when leaving a message are ? -- Executing VoiceMail("SIP/209-3b0e", "u5") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing

Hey all, I just tried running a 'make rpm' on a fresh install of Fedora Core 4 and ran into an error near the end of the build process. This is the output of the build when the error occurs: done rm -f /tmp/asterisk/var/lib/asterisk/mohmp3/sample-hold.mp3 mkdir -p /tmp/asterisk/var/spool/asterisk/voicemail/default/1234/INBOX :>

I have installed my first Asterisk implementation using the Asterisk@home ISO. I am using the SJPhone software. Using the setup page, I have been able to configure two extensions. Whne I dial from one to the other, the other does not answer even though it is registered. Watching the log in the CLI, I can see that recorded messages are being played;: == No one is available to answer at this time

Hello, In my sip.conf I have: ;Register and forward FWD numbers to internal extensions register => FWDNUMBER:PASSWORD@fwd.pulver.com/9500 Which should register Asterisk at FWD and then when any calls are made to FWDNUMBER those calls should be forwarded to extension 9500 as specified in the extensions.conf. What I am getting is it is trying to dial the 9500 (IAX Firefly) client twice when

Hello all, I've installed AAH 1.1 on my VIA C3 powered mini PC. I've made the necessary changes to the * makefile, so the compilation went well. The first thing I did was configuring two extensions from AMP, namely 200 and 201. Then I installed X-lite on two PC's and configured them with one of the extensions: System settings - SIP proxy - Default: Username: 200 Authorisation user:

Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten => *97,1,Answer exten => *97,2,VoicemailMain(${CALLERIDNUM}@default) exten => *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing

I am using the following macro to dial a ZAP channel. When I dial in, * answers and I go to voicemail. I never hear any ringing, though. It doesn't work with the Ringing command before or after the Dial command. [macro-zapdial] ; ; call a ZAP extension for ${ARG2} seconds, and then voice mail ; ${ARG1} - Extension ; ${ARG2} - Time to ring exten => s,1,Dial(ZAP/${ARG1},${ARG2}) exten

Hello, I'm facing problem where as whenever there are incoming call from pstn, there will be no audio coming in. User at the other end also could not hear my voice. This happens few days back. Im using asterisk 1.6.1.2 with dahdi tool 2.2.0. I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and asterisk 1.6.2.5. However, it does not help at all. My current config as follows :-

Hi there, im having some troubles with my asterisk service, sometimes when im trying to make an outbound call, to any of the phones configured on the asterisk box, it enters inmediatly to voicemail and then hungs up. After that its necessary to stop the service and putting up again manually. Here is a piece of my log file when a call is trying to incoming: "Jun 9 06:30:16

Hello, all. I am attempting to use IMAP voice mail storage in Asterisk 1.6.1.1 on CentOS 5.3 using Zimbra 5.1.6. I will not be using it as it has proved terribly unstable - Asterisk segfaults on every voice mail message although the message is successfully deliver to my email inbox - but I thought I should report it. Here are the errors from the Asterisk console: -- Executing [210 at

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 -----+|

I am attempting to write in incoming context for calls. 1. If the caller id is given and it is not black listed it will Playback a greeting and then right the phone or go to voicemail under busy or unavailable conditions 2. If no caller id is given, then Privacy Manager will ask for the number. I am testing 6145551212 to see if the black list will work 3. If a caller id is given, and it is

All, I am setting up my first * box and am trying to configure SIP Softphone to SIP softphone dialing. When I dial ext. 2001 it rings once (Very short) then immediately goes to voice mail for ext. 2001. I keep seeing the "Got SIP response 482 "Loop Detected" back from 192.168.1.252", immediately followed by "No one is available to answer at this time". I've

What would cause this? It happened out of the blue: -- Executing VoiceMail("Zap/3-1", "u326@default") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/6' (language 'en') -- Playing

I have the following in extentions.conf: exten => 2111,1,Dial(SIP/2111 at gw1.langley) exten => 2111,2,Voicemail(u2111) exten => 2111,3,Hangup exten => 2111,100,Voicemail(b2111) exten => 2111,101,Hangup I have the following in sip.conf: ; Cisco 1750 [gw1.langley] type=friend host=172.16.17.1 context=default canreinvite=no Like the ATA, lots of stuff doesn't work on the 1750

Hi all! This is my VoIP network scheme H323EndPoint ----- --- GW H323/SIP-IN -- -- SIP Phone | | (Sipquest) | | | | | |

Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a