similar to: Is "Meetme" a generic term?

Is there anything to choose, in performance, between a TE405P and a TE410P? I understand the difference between the PCI bus voltages, and certainly don't intend to try Andrew's hacksaw operation :-). But if I choose the card first, and a compatible mobo second, does it make any difference which? Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play:

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, I have a project coming up which will need to interface Asterisk to E1 trunks in the UK. I have a couple of questions which I hope someone can answer, or give me some pointers: 1. If I want two E1 trunks, is there anything to choose, performance-wise, between using two ports on a single TE405P, and using two E100P cards? 2. How can I test the E1 operation in the lab, which doesn't

Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:

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>

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

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

Thanks for the suggestion Tony, I installed each codec for MoH, core sounds, and extra sound packages. Unfortunately the tests produce the same results. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 ( continuously for a while followed by a [Sep 1 20:36:46] WARNING[7761][C-0000770d]:

I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do "asterisk -rvvvvv" on a normal login, either via the

In article <E02FA554-9D6D-4E7D-8A78-5FBDE1DE939D at kicp.uchicago.edu>, Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: > > > > On Sep 26, 2020, at 8:05 AM, Jerry Geis <jerry.geis at gmail.com> wrote: > > > > I have a disk that is flagging errors, attempting to rescue the data. > > > > I tried dd first - if gets about 117G of 320G disk

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>

In article <ml1jnh$afr$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > When I started using CentOS 6 instead of CentOS 5, I discovered that > "less" no longer understood \< and \>, which I had been used to using > since almost forever. > > Eventually research revealed that in the Fedora version on which > RHEL 6 was

On 06/09/2015 12:48 AM, Nicolas Thierry-Mieg wrote: > On 06/09/2015 12:33 AM, tony at softins.co.uk (Tony Mountifield) wrote: >> In article <ml1jnh$afr$1 at softins.softins.co.uk>, >> Tony Mountifield <tony at softins.co.uk> wrote: >>> When I started using CentOS 6 instead of CentOS 5, I discovered that >>> "less" no longer understood \<

I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and

Does anyone know how a program, script or shell user can best determine whether the machine is running on bare metal or is a VMware guest? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org

I want to use fail2ban on CentOS 6 to monitor Apache with the standard default logfile format ("combined"). Has anyone here succeeded in doing so? The format has the IP at the start of the line, followed by two dashes (if no authentication) and THEN the timestamp. What I've read on the fail2ban wiki seems to say that the timestamp must ALWAYS be at the start of the line, followed by

Has anyone here used Asterisk inside a WatchGuard firewall, talking via the WatchGuard SIP Application Layer Gateway to an outside SIP service? I have a customer doing just that, and I am 100% convinced there is a bug in the ALG regarding the media port number it inserts into the SDP when it rewrites it. However, either they or WatchGuard will not accept there is a bug, despite my very detailed

Hi, despite some searching I haven't found an answer to this question: Is there a way I can see in the log, or find out in the dialplan, what codec has been negotiated for a SIP channel? If possible, I'd like to do this in both Asterisk 11 and in an old 1.2 system. What I'm specifically trying to do is to determine historically the usage of the G.729 licences installed in a system,

In article <dafd6c5f-9676-c4da-8761-619d6f003525 at microlinux.fr>, Nicolas Kovacs <info at microlinux.fr> wrote: > Le 01/10/2019 ?? 09:05, Liam O'Toole a ??crit??: > > [...] > > > > FWIW, the current release has squidguard 1.6 > > https://packages.debian.org/source/stable/squidguard > > And this all begs the question: who's flying this plane?