similar to: Can't delete voicemail messages

Hey folks, I stumbled across the Mastershaper project ( http://www.mastershaper.org/ ) but I have a little problem: I wanted to shape the traffic coming from the router itself aswell as coming from the LAN behind the router, for that task I need IMQ, but with IMQ iptables-(layer7)-matching is not possible. Now I''ve talked with the programmer and he said the following: >The problem is

Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

Hello I have two questions ! 1. What is the best speech recognition engine for asterisk? I have searched and asked on forums and found that lumen vox is best for asterisk bala bla bla 2. For TTS (text to speech) which TTS engine will be better to use ? I have tested Flite , cepstral (i have not buyed lisence for it trial only) but still thinking may be i have a good option ? -- Best Regards

Hello list. The VoiceMailMain()-application has an advanced menu. Can I get a Voicemail-application that has less functionality ? I only want the user to listen to new voicemail-messages (and delete them), not the functionality with the folders and redirecting messages to other mailboxes... I've looked at the code in /usr/src/asterisk-1.4.30/apps/app_voicemail.c but it seems complicated

Hello all I need your support in a big decision in front of two alternatives related with *. I must buy an E1, in order to manage 30 channels, given this big price; or I could opt for a 15 BRIs without cost to replace the same number of channels, and the question that inmediatly emerge is : ? can asterisk manage 15 BRIs ? If yes to the latter, could posible somo guide, for instance, wich Digium

hello list, I subscribe to Sixtel.net for a DID just to see how it worked. they say DID active inmediatly , but after 4 days , I have no DID , I tried to call tech support office and it seems they all ways are in hollidays, a support ticket no good luck, I call back service also not good luck. so anyone here has experience with them? are they a SCAM? fortunally only put there $10 bucks but

Which version? Similar issues parsing callback number in 1.2.12 > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Kristian Kielhofner > Sent: Thursday, September 28, 2006 10:27 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail callback

This is the first time i wrote you, i would like to know if you have any idea about my problem: ? I install and conigure the last version of openssh and it is working like a root user ? $ ssh root at server ? and it?s just fine, even i can use CVS software, but if a try to use it with other user it doesn`t work ? $ ssh any at server ? ask me for a password and it wrote me, ?have a lot of fun? and

The problem description below is relevant for those who use samba + LDAP. We installed four Intel Xeon servers with standard SuSE 8.2, samba + ldap. The W2K client complained about very, very, very slow reponse from the server. Below is we descripe the reasons and the solution. We have big problems with openldap version 2.1.12 (standard suse 8.2 rpm) and the name server cache daemon versiom

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

Greetings all. Some small question regarding schedule poll operation hypercall. 1. struct sched_poll poll.timeout is measured in what unit of time? Secs, ms? ns? 2. After issuing the hypercall_sched_op(SCHEDOP_poll, &poll); if no timeout is used in poll struct how long will I yield the CPU? 3. If I issue the hypercall and the event never comes is it possible to to yield the CPU for ever?

Following up with some more tests... I just left a message in a voicemail box, and got the same behaviour (i.e. it says its starting to play ('First Message') then it goes to the VM ending menu (i.e. press 7 to delete) without playing the message. On this particular test.. I went ahead & left a 2nd VM in the inbox, then when I went to check them, they would both play properly... The

Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video attachment while using any video phone. This does work in my 1.8.23.1 installation. The file is skipped with the ast_streamfile error (and moved to OLD), and the prompts following that message display the ast_best_codec error. [Apr 7 16:05:50] WARNING[17497][C-00006fdd]: file.c:1017 ast_streamfile: Unable to open

Well, I've installed Wine Doors, and I have problem. First of all, I'm trying to emule GUITAR PRO 5. The instalation runs right, and all seems to work, but once opening GP5, it pups a window saying that: Violation d'?cces ? l'adresse 39840950 dans le module "gdiplus.dll". Lecture de l'?dresse 00000048. Well, after that I've installed timidity and executed it, but

Hi, I want to use G.723.1 on *, I read it is supported in Pass Through mode, but I don't understand whats the meaning of that. I have a GW 5300 and an ATA 186 and I want to place calls to PSTN. I setup this config: [general] port = 5060 bindaddr = xx.xx.xx.xx context = sip tos=throughput maxexpirey=360 defaultexpirey=120 [gw5300] type=friend insecure=yes host=xx.xx.xx.xx

Is there a way to distinguish, in the answer of the Dial command, when a channel is not available (for example, an unregistered but valid SIP user) v/s when the dialed channel is inexistent, even when it matches an extension? For example, I've the following simple dial plan: exten => _XX,1,Dial(SIP/${EXTEN},10,) exten => _XX,2,GotoIf(DIALSTATUS = CHANUNAVAIL?4:3) exten =>

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?

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

is the hint 99,hint,ZAP/1 supposed to work or how do I get the lights on the phones to display channels in use in addition to extensions in use?

Hola a todos, Mi nombre es Juanjo. Llevo un tiempo intentado hacer funcionar un programa que necesita de la instalación de RPROJECT y REXCEL. He mirado varias guías e instalado los programas pero obtengo varios errores. El principal es que tras instalar RProject, Rexcel y statcon y al intentar correr esta macro desde Visual Basic… Sub Start() RInterface.StartRServer