similar to: Hangup after voicemail not detected

Hi there, Sorry for the late reply. I had too many emails in my mailbox to clean up. Anyway, I found out the problem is the sendmail in Linux did not work and the voicemail.conf in our asterisk is ok. There is another issue for email notification: some email servers rejects the email from asterisk. My engineer added Asterisk IP into the DNS to slove the email rejection issue. Thanks for

Hi On my FC3 box with asterisk 1.0.9....MusicOnHold is not working. It starts and stops immediately... An unknow option mono comes...from where it is originating.?? As there is nothing written in .conf file. Console output is below: I am using mpg123 version 0.59r. Although I am able to play music with mpg123 but why it is on No-cooperation movement against asterisk ? Need help..any

Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten => 2,1,Playback(/media/asterisk/answerphone-en) exten => 2,n,VoiceMail(2000,s) exten =>

Hi All, I am a newbie and trying to install Asterisk from instructions given in http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have Centos 3.3 so I downloaded rpm's from ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/ and tried installing one by one but I get the following errors error: Failed dependencies: asterisk = v1.0.9 is needed by

http://dovecot.org/releases/1.0/dovecot-1.0.10.tar.gz http://dovecot.org/releases/1.0/dovecot-1.0.10.tar.gz.sig v1.0.8 and v1.0.9 were a bit bad releases. Hopefully one day I've managed to have written a proper test suite which can be run before doing any releases.. * Security hole with LDAP+auth cache: If base setting contained %variables they weren't included in auth cache key,

http://dovecot.org/releases/1.0/dovecot-1.0.10.tar.gz http://dovecot.org/releases/1.0/dovecot-1.0.10.tar.gz.sig v1.0.8 and v1.0.9 were a bit bad releases. Hopefully one day I've managed to have written a proper test suite which can be run before doing any releases.. * Security hole with LDAP+auth cache: If base setting contained %variables they weren't included in auth cache key,

Hello, Let say Alice and Bob both have a sip phone connected to the same asterisk 11 box. Alice has T.38 enabled softphone. When Alice sends a fax to Bob extension, the following happens on my system: - Bob phone starts to ring - Bob answers - asterisk sends the incoming call to appropriate fax extension - Bob is hearing nothing at all: no tone, no sound at all. I want to play an audio file

Hi All- I am running into a small problem trying to implement voicemail Directory(). I'm sure it is a simple thing, but I can't figure out where the problem lies. I can get into the directory without a problem and can look up users by their last names, however I hit a snag when asterisk says "if this is the person you are looking for press 1 now". When I hit 1, the attendant

Hi, whenever I include a "Ringing" Line in some Voicemail Extension I get an error when a call from the outside (via ISDN) comes in, but it works when an internal (SIP-phone) calls the extension. Here is my configuration for testing: ------------extensions.conf------------ [isdnext] ; strep external "101", our number and leave only extension exten =>

mkumar@mantragroup.com wrote: > Hi All, > > I am a newbie and trying to install Asterisk from instructions given > in http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have > Centos 3.3 so > I downloaded rpm's from > ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/ and > tried installing one by one but I get the following errors

Hi all Asterisk 1.8.11.0 on Centos 6.5 My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom, South Africa). Unlicensed G729 codec version on server. 75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording. The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. I noted

Hi, voicemail plays after hitting "#" as final file "auth-thankyou". Is there any possibility to change this behaviour? Custom soundfile or disable it perhaps? Thanks for your answer(s)! -Thorsten-

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing

I'm happy to be able to announce that Xapian 1.3.4 is now available. Please note that 1.3.x releases are development releases - they are made to encourage earlier and wider use and testing of new and changed code. Our record with 1.1.x was very good - all the bugs I am aware of were either in new features, or were also present in the corresponding 1.0.x release. But if you main concern is

On Fri, 01 Jan 2016 at 18:19, Olly Betts wrote: >I'm happy to be able to announce that Xapian 1.3.4 is now available. > >Please note that 1.3.x releases are development releases - they are made >to encourage earlier and wider use and testing of new and changed code. > >Our record with 1.1.x was very good - all the bugs I am aware of were >either in new features, or were

Hello, I finished an Asterisk installation this weekend, and I'm experiencing a problem when a user hangs up on a line after leaving a voicemail message. I found two similar issues when reading through the archives, and have not been able to resolve my issue from their answers. http://lists.digium.com/pipermail/asterisk-users/2004-April/042453.html

dear list if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain -- AGI Script Executing Application: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205 __adsi_transmit_messages: Un able to send CAS May 21 04:25:10

Hi list, I have been looking around for a while now, and cant seem to get to the bottom of my problem. My setup is that I have a separate SIP server that servers my SIP subscribers, and I want to use Asterisk purely for voicemail for now. So I set up a common SIP extension at my SIP server, and made Asterisk register it, so that normal users can forward calls to that common extension, and

Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make

Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make