similar to: Voicemail RFC

Running on branch/1.4 I have been watching some the queries from Asterisk and I think I have a place where some efficiency can come, but I am at a lost as to what is calling it... It appears that every 10-15 seconds, every mailbox INBOX and Old gets queried for the number of voice mail files. I have exposed SIP to verify that it wasn't the phones requesting. It is not much of a problem in

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 anyone other than me having trouble dialing out via IAXTEL? I havn't changed my config files in weeks but seems that IAXTel calls (800 and FWD) stopped working in the past week sometime. Robert

Hi, The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all my machines, no error, no log. Although, the CVS version works great on Linux. Is it anybody who knows how to compile it with mingw32 ? Or better, could anyone, who already has mingw32 installed, make a binary snapshot ? Thanks in advance, Jean-Christophe -------------- next part -------------- An HTML attachment was

Hi all, I am using to Xlite to save video voice mail. when i retreive it, then only video show , no voice is here out. Plz tell me where ,i am wrong , and how i can able to see video plus here audio in voice mail box. I did following configuration In Sip.conf videosupport=yes [phone1] type=friend host=dynamic context= employees mailbox=101 at default

Hello List - I've done some reading on voip-info regarding hardware requirements for an Asterisk server; but I haven't been able to find anyone doing what we plan to, so I am hoping you can assist. We are looking to provide a voice mail only Asterisk solution for approx. 100 homeless people, a customer of ours is planning to provide the service. The Asterisk service will reside in our

If my extension is 22, and voice mail access number is 909, then with exten => 909,1,voicemailmain(s22) I can access voice mail 22, without number and password prompt. But, I want that every extension can access its voice mail without number and password. So, when I put exent => 909,1,voicemailmain(${calleridnum}) voicemail want only password. I want to eliminate password too, so when I

Thanks for all your help Martin, Guys, This is a good find and hopefully could help someone else. I've been having a problem with forwarding voicemail from one mailbox to another. I ran down the sendmail and soundcard path and came up goose eggs. With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9 Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it

Hello All, I have my own AGI script running and I am trying to push the call to voice mail when Busy, Unavailable and Not Answered. Everything is working fine but the only problem is voice mail greetings for Busy and Unavailable is not played. By default only "Temp Greetings" voice mail greetings is played. I am passing the correct parameters for Busy => 'b', Unavailable

I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup

Does anyone know how to send a message to a Cisco 7940/7960 phone running SIP images 6.3 telling it to light up one of its LED's when new voice mail arrives? I found alot of web based solutions http://www.voip-info.org/wiki-Asterisk+GUI and easy ways of getting email or getting paged of a new voice mail - but nothing where you can just look at the phone and see a blinking light or

-- Im not sure if you recived this message, so I hope it is okay that i resend it. Hello everyone, I have experienced that my audio indication for waiting voicemail don't work after i upgraded my system from 1.0.9 (Gentoo) to 1.2.1(FreeBSD6). I get the following message when i got a context in sip.conf with mailbox=something: Got SIP response 481 "Call/Transaction Does Not

can anyone give me a reference to the retrieval of voicemail from the Asterisk PBX using a cisco 7940 phine running sip image. i have configured a single voicemail box using the script, the corresponding entry in voicemail.conf and configured the extension to use the voicemail box . i can see from the asterisk console the message being passed to the voice mailbox, and correspondingly the sip

Try this subject line if you will. On 11/14/06, joe a. <joea@j4computers.com> wrote: > > Did not know how to make up a subject line for this. > > I have a dial plan that allows a caller can try my cell phone. And that's > fine. If the call cannot be made, it sends caller back to voice menu. > > However, I'd like a way for the caller to elect to go back to the

I have really tried to do my best googling and wiki-reading before asking this question. I couldn't find the answers there so I throw myself at the mercy of the list... I get excellent quality for SIP -> PSTN and PSTN -> SIP calls, however when I or anyone else calls from PSTN -> * the voice menus are oftentimes very choppy. Sometimes they are absolutely perfect and I cannot tell

The orginal did not make it to the list... Spam filter issue??? No repeat of the lockup yet. Lyle -------- Original Message -------- Subject: voicemail locked up Asterisk 1.4.13 Date: Thu, 01 Nov 2007 20:57:27 -0500 From: Lyle Giese <lyle at lcrcomputer.net> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> I am running Asterisk

Hi, I didn't get any response to this posting so thought I'd post again in the hope that anyone who missed the posting the first time may be able to offer some ideas. This problem isn't specific to the particular model of phone - I can see from sip debug that the local extension itself is not being sent any Messages-Waiting headers. Any alternative strategy would also be welcomed

Good day all I got my voicemail message working,thanks but now,keep in mind I'm using SIP We have,for example 4 people in our admin department.Each user has its own voicemail so that when their extension is dialed directly and not answered it gos to voicemail. But there is also a option to dial 3 for admin with will dial all 4 number in sequence.This I got working 100% but now I want a

At last weeks' LPC, there was some interest in my patches for Auto/Lazy Migration to improve locality and possibly performance of unpinned guest VMs on a NUMA platform. As a result of these conversations I have reposted the patches [4 series, ~40 patches] as RFCs to the linux-numa list. Links to threads given below. I have rebased the patches atop 3Nov10 mmotm series [2.6.36 + 3nov mmotm].

At last weeks' LPC, there was some interest in my patches for Auto/Lazy Migration to improve locality and possibly performance of unpinned guest VMs on a NUMA platform. As a result of these conversations I have reposted the patches [4 series, ~40 patches] as RFCs to the linux-numa list. Links to threads given below. I have rebased the patches atop 3Nov10 mmotm series [2.6.36 + 3nov mmotm].