similar to: MeetMe conference delay increasing

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

Just a trivial question: was the term "Meetme" invented for Asterisk as something like a brand name for its conferencing? Or was it an existing generic term for dial-in conferencing? Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org

Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should

Hi, could anyone explain the intent and usage of the new e and E flags in the MeetMe app? The Wiki doesn't mention them yet, and I have not been able to find any other documentation of them. Thanks Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org

Sorry Newbie asking everyones option. I am setting up a couple of small asterisk phone systems for my work, I started using some snom 190 and bt102 sip phones (the bt102 works really well with iLBC), but the complaint from my workmates is there is no way to see if other people are on there phone or not, or what lines are being used. The snom 190 only has 5 function keys, the snom 220 seems a bit

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

This question is directed towards those who are familiar with the inner workings of the Asterisk code. I'm quite at home hacking on the source code, and have become familiar with certain parts of Asterisk's operation. I'm looking for some advice on the most fruitful avenues to explore in order to achieve a particular application I need: either in the source code or in AGI (with which

I have a customer with an Asterisk 1.4 system (r144238 - between 1.4.22-rc5 and 1.4.22 released). It uses SIP to connect to the PSTN via a provider who is on the same LAN as the box (it is co-located at the provider). They also have about 20 SIP phones as extensions that connect to the box over the internet. "sip show peers" indicates that most phones have a latency of 90ms-100ms. The

I wonder if anyone can suggest ways to diagnose an infuriating problem being experienced by customers of a company I did a large Asterisk project for. First some background: The system is a conferencing system using a modified MeetMe. There are seven Asterisk boxes (we call them bridges) each with four T1 PRIs into a TE405P. No VoIP is involved. A conference is always local to a single bridge.

In article <mfbt6f$9rt$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that > is behind a network device to which I don't have ready access, which is > performing NAT with possibly some kind of SIP ALG, and an Asterisk 11 > system on a public IP. > > My question is

I know that I can record the contents of a call by calling Monitor() or MixMonitor() from the dialplan just before invoking Dial(). I have a potential customer who wants only the first minute of each call recorded (for identification purposes, without the storage overhead of keeping the complete call). Can anyone here think of the easiest way to do this? The only possibilities I can think of

In article <n009u2$85v$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > Apologies, this is slightly off-topic being to do with an EPEL package, > although it's running on CentOS6, so I thought others here might have come > across this issue. > > I have five CentOS 6 systems running fail2ban from EPEL, and this > package was updated

> -----Original Message----- > From: Tzafrir Cohen [mailto:tzafrir.cohen@xorcom.com] > Sent: Wednesday, October 25, 2006 10:18 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Re: Meetme... No channel type registered > for'zap' > > > On Wed, Oct 25, 2006 at 10:06:02AM -0600, Douglas Garstang wrote: > > > -----Original

All, Has anybody else experienced garbled voice between a phone using alaw/ulaw and one using iLBC? I have a Nokia E series phone with a preference to use iLBC and this works fine in Asterisk 1.2. However, since moving to 1.4 - I get garbled voice on Inbound (g711->iLBC). Outbound voice seems fine (iLBC->g711) though. It's not a 20/30ms framing issue as the phone uses 30ms

Le vendredi 10 juin 2005 ? 21:27 -0400, SteveK a ?crit : > I'm not an expert either, but I see people choosing iLBC over speex > all the time with asterisk; partly it's because they have more > market share in hardphones, and partly it's because of marketing and > such. (another reason is that iLBC source is included in asterisk, > and speex is only compiled in

In article <CAG2kNCyjsQZ2qW_8BBLp8BH_20=JgxoEYpn9BSwZhXg7_rHBbg at mail.gmail.com>, Gianluca Cecchi <gianluca.cecchi at gmail.com> wrote: > On Mon, Nov 21, 2016 at 12:49 PM, Tony Mountifield <tony at softins.co.uk> > wrote: > > > I am just applying the latest C6 updates to a couple of KVM Linodes. > > It appears that the latest update of util-linux-ng has

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

Ok, that sounds a little more elegant. Does that delete switch delete those files after download, or does it stop it from downloading at all? On Mon, Jan 29, 2018 at 10:48 AM, Tony Mountifield <tony at softins.co.uk> wrote: > In article <CANZsmmM6C_F+NuPdjd+mGDEXaJVcfc1bdhpWdESSbC2CR7Dz3 > g at mail.gmail.com>, > Felipe Westfields <felipe.westfields at gmail.com>