similar to: [patch] allow the transfer keys from app_dial's't' and 'T' and hangup key 'H' to be configured via features.conf

Can anyone tell me how I can implement the features added in the following link for call transfer? The authors seem to feel they are finished but it doesn't appear to have been integrated into what everyone can download. It is referred to as a patch but I don't understand how it could be applied. Here is the link: http://bugs.digium.com/bug_view_page.php?bug_id=0002010 I guess I just

Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section

I wonder if anyone else's Asterisk box drops the connection to voicemail after 30 secs even when the maxmessage parameter is set to 180 (3 mins). Here is the general section of my voicemail: [general] ; format=wav49 maxmessage=180 attach=yes Even if it only gave the caller 30 sec to leave a message it would be nice to tell the caller that they have run out of time before ending the

Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware. http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch= -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

I hate to bring this thread back to life, but... > it may be possible to get it supported, do you think the price >point is remotely competitive with Digium hardware? Also as I am not >about to divulge my information to them to look in the downloads >section, what is the licensing of their SDK? What is the licensing of >the driver? > >Steven >On Tue, 2002-11-26 at 14:52,

I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone.

I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? -------------- next part

Why does this person have my e-mail address ? ----- Original Message ----- From: <markogift@astriholics.org> To: <hackerwacker@cybermesa.com> Sent: Tuesday, November 23, 2004 1:13 PM Subject: Gift for Mark Spencer > Hello everyone! > > We have been thinking about something that we could do for Mark > Spencer as a holiday gift. We have decided to try to orgranize a

hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am running ntpd on the same machine as asterisk in order for the grandstream phones to display the time. After a while the time display fails until the phone is re-booted. Has anyone run into this problem before? Is it simply a bug in the GS firmware? Sean

Friends, Romans, Countrymen, lend me your ears! Or, in this case, your eyes. It has become a problem I've noticed over and over again, that we tend to think the mailing list is some sort of forum, used to post whatever your hearts desire. This list, not unlike any other mailing list, has a purpose. A focus if you will. Hopefully this 'document' will point out some key things we

Hello World, I am a newcomer to this technology. I am very pleased to have found it and VERY interested in using it, as a better-cost alternative for schools where I teach open source & Windows technology. I travel the world, and help set up schools in underprivileged countries where it is very expensive to acquire the Windows desktop. Wine seems like a VERY good solution, much better than

I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called me@NuFone/1515480XXXX -- Call accepted by

On Sun, Sep 20, 2020 at 09:34:50AM -0700, Kevin Brott wrote: > On 9/19/20 11:02 PM, Damien Miller wrote: > > OpenSSH 8.4p1 is almost ready for release, so we would appreciate testing > > on as many platforms and systems as possible. This is a bugfix release. > > Debian GNU/Linux 10 (buster) > gcc version 8.3.0 (Debian 8.3.0-6) > OpenSSL 1.1.1d? 10 Sep 2019 > >

Hello guys, i have been on this mailing list for some weeks now, and i was wondering if someone here has installed linux on the XBOX and use it as a dedicated server. Its a 200 USD computer and could make it perfect to asterisk, its little and doesnt really take much space. My question is could this make it for a stable server??? here are some links i found for linux on XBOX

Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland

On Sun, Apr 15, 2018 at 06:38:37PM +1000, Damien Miller wrote: > On Sat, 14 Apr 2018, Colin Watson wrote: > > This isn't an OpenSSH bug, but just in case anyone else is ambitious > > enough to automatically run interoperability tests against other SSH > > implementations, here's a heads-up: > > > > https://twistedmatrix.com/trac/ticket/9422 > >

This isn't an OpenSSH bug, but just in case anyone else is ambitious enough to automatically run interoperability tests against other SSH implementations, here's a heads-up: https://twistedmatrix.com/trac/ticket/9422 TL;DR: commit 7c856857607112a3dfe6414696bf4c7ab7fb0cb3 caused there to be an SSH_MSG_DEBUG message sent where there wasn't one before, which tripped over an

Hi All, Can we stop this thread pl. This lady has no intentions to learn asterisk. She is just a troll and wasting our time. With her corporate attitude, what she expects is support that available with paid commercial products. Her company has enough money to buy commercial products, let she go there. Hey lady, whoever u are, dont waste our time. this is not for u. Lets move on to something

Hi, I am trying to migrate a domain (testDomain) from node1 to node2. Here is some information about the problem: node1#: xm migrate testDomain node2 After the command is executed, there is some communication between the node1 and node2 Xen hosts. xfrd daemon (which is forked by xend) is primarily responsible for vm migration and there is error in its log (/var/log/xfrd.log) The exact