similar to: Voicemail not working properly

Hopefully someone here can save my sanity. I have been trying to solve this problem for days now, but just cant put my finger on it. Im new to * so I have probably done something stupid! I have a TDM400P with one FXO module and a FXS module. The main problem I have is not being able to get the extension attached to the FXS module to ring or be able to make calls. It gets a dialtone fine but I

hi, i am trying to do to autoattendant. here is my extension.conf part [tumpak] exten=>s,1,Dial,Zap/4|10 exten=>s,2,Voicemail,u9999 exten=>s,102,Voicemail,b9999 exten=>t,1,hangup so when a caller dials the extension 2 suppose, it enters to the above context.. everything is fine. the problem is when the caller hangs up the asterisk does not. after caller hangs up and tries again he

I'm having a few issues with the MySQL realtime configuration in CVS-HEAD. I tested it initially with realtime extensions (realtime_ext => mysql,asterisk,extensions) and a realtime switch in extensions.conf and that works fine, So I though I'd go back and test a static configuration mapping. I used the table structure from the asterisk guru postgres howto to create something

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

Paste your extensions.conf section that is relevant. -Matthew ----- Original Message ----- From: "Greg - Cirelle Enterprises" <gcirino@cirelle.com> To: <asterisk-dev@lists.digium.com> Sent: Monday, December 27, 2004 4:32 PM Subject: [Asterisk-Dev] realtime voicemail > Let me clarify my last message. > > If I put in the wrong password I get polled > again for

I am currently in the process of testing out realtime support for sip.conf. I have followed all of the directions that are listed in the Wiki, but for some reason this does not work. When utilizing a flat file, I am able to register endpoints without any problems, and calls can proceed. One interesting side effect that I have noticed is that when I am using realtime for sip, I am unable to see

Madhur Amilkanthwar via llvm-dev <llvm-dev at lists.llvm.org>於 2020年8月9日 週日，上午1:53寫道： > Hi Paul, > I hope you have gone through > https://llvm.org/docs/Contributing.html#how-to-submit-a-patch. > > Generally, I would do 'git add' on the new file. 'git diff' should show me > the newly added file. Further, I'd just do 'arc diff' and this should

Hi, I don't if was yet an issue. It really would be nice if each user is able to active/deactivate the mail forwarding of his voicemail via the VoiceMailMenu. Regard

Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network

I am ready to submit my first real submission for review with Phabricator. Please forgive my meager knowledge of Git. I did a 'git diff' to generate the diff file. The contents look good. However, there is one new file, a TableGen test file. How do I get that file included in the diff, or otherwise included in the submission?

Pentium(R) D CPU 2.80Ghz 3 GB of Ram 250 Gb harddrive i not sure if i'm downloading the right one here as my mechine runs on 1386 scale could someone help us ouit to see if i am downloading the right one Mike CentOS-5.3-i386-bin-DVD.iso -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Sir, I have implemented Samba PDC. Its working fine. But o do Highly Available, I have been trying to make it in 2 node cluster. Everything is running fine. But facing a problem, which I want to share. When I shift PDC to another cluster node. Everything is shifting fine. But my existing user can not log in. The can logged in again if I rejoined that mechine again to domain. I am explaining

I've added an introduction article about the ARA on my web site http://www.voip-forum.com/ The same text is now also added to CVS head as README.realtime. On the same site, you will also find the news item about how we used Asterisk for a call from an airline jet above Greenland to Stockholm, Sweden. The world is getting smaller and more connected every day! /Olle

Hi, all.... Honestly I'm a newbie in Linux. For a couple days, I've been trying to build my own PDC server. I've got enough manual resource for guidance, but somehow, untill now I can't log my Win2000 Profesional into the PDC server. I'm using Debian sid kernel 2.6.5, Samba 3.0.2a and a Win2000 box as client. I think there's nothing wrong with my smb.conf ( I guess...

I have a CISCO 7912 phone, the LED on the phone does not glow when there is new voicemail, can we configure Asterisk to have the LED glow on new Voicemail. Regards, Sanjay Rajdev

Hi all How can I make * ring one phone then if no answer Go to a different extension ?? Any help always appreciated Regards Mick

hi all. i really need help. i need a working example on shaping the ingress per user using tcng without IMQon a mechine which has two interfaces, and acts like a firewall, and NAT for intrenet connection sharing: eth0 is the external facing the Internet. eth1 is the internal towards my LAN/office network. Please i dont want other than tcng code. iptables code i read on some pages seems

Hello My asterisk is stoping. i am using asterisk with ser on same mechine here is the asterisk trace ------------------------------------------------ -- Setting call duration limit to 3000 seconds. Sep 2 15:58:12 WARNING[10334]: rtp.c:852 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Sep 2 15:58:12 WARNING[10334]: chan_sip.c:2313 sip_alloc: Unable to create RTP

Hello everyone, I’ve met a strange problem with 4.8.3. I firstly built a DC using samba 4.8.3 in one linux, and I can create a domain ‘euler.huawei.com’ successfully. Then I try to install samba 4.8.3 in another linux mechine, and add it to the existed domain. It failed when I used the command ‘samba-tool domain join euler.huawei.com DC ‘ to join it to the domain. Here’s the log:

Just wondering if anyone has tried MAC OS X and panther. I will like to do SIP to H323, not sure if this will be possible on the MAC because of the Libraries PWlib and OPenh32 for Linux.. Just curious.. Anyway, anyone has an easy guide (step by step) to setup oh323 with asterisk. I saw a guide but i am not very savy on linux. thanks, Francisco ----- Original Message ----- From: