similar to: USB phone for Linux?

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

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,

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

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

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

I have a potential customer who has an existing PBX with analogue FXS ports connected to phones. He wants to allow a single remote worker to be connected to one of the analogue extension ports using VOIP. I know I could do it using Asterisk with an X100P card, but that seems a bit overkill. Does anyone know of an analogue->VOIP adapter that has an FXO port in it instead of just an FXS port?

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

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

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

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

worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -----Original Message----- From: tony@softins.clara.co.uk [mailto:tony@softins.clara.co.uk] Sent: Tuesday, April 20, 2004 12:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In

In article <301a2e78-d490-3805-e30f-41b668aac5c1 at sysnux.pf>, Jean-Denis Girard <jd.girard at sysnux.pf> wrote: > > Hi Tony, > > Le 20/07/2019 à 06:29, Tony Mountifield a écrit : > > Are there any other languages/libraries I should be considering? > > Same here, after years of AGI / AMI, I recently made my first project > using ARI on Asterisk-16. I love

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]:

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>

@tonymountifield Does this still hold true? https://superuser.com/a/1075837 On Sun, Sep 27, 2020 at 7:21 AM Tony Mountifield <tony at softins.co.uk> wrote: > 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

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

In article <55D20981.7030902 at centos.org>, Johnny Hughes <johnny at centos.org> wrote: > On 08/17/2015 10:57 AM, Tony Mountifield wrote: > > I recently applied updates to a CentOS 5 box running MySQL. I've discovered > > that the new version of openssl, 0.9.8e-36.0.1.el5_11, breaks MySQL SSL > > connections. > > > > If I rename

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

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,