similar to: SIP audio delay after call transfer?

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've just noticed a strange behaviour with a MeetMe conference. I have a pair of phones (GS BT102) on my desk, and dialled both of them into a conference on speakerphone. If I spoke or made a sound, I heard it replayed from both speakers together a split second later, as expected. I went away for about 15 minutes, leaving the conference running. When I came back any sound I made came back

Hi, I'm using OH323, mostly with success, to interface Asterisk to a provider's switch (World Telecom INX). I have noticed a particular effect, and I wonder whether anyone else has seen the same? The effect is audio flutter (almost like the flutter one gets on MF or HF radio sometimes) which only happens intermittently. Audio coming into Asterisk is unaffected, as proved by using the

I'm attempting to get chan_h323 working on Asterisk CVS-STABLE. I've compiled it ok using the Janus release of pwlib/openh323, by editing the makefile as per the comments. Call setup and cleardown seems to work fine, but no audio is being passed in either direction. Doing an "h.323 trace 9", I noticed the following sequence at the end of the call setup: h323.cxx(1685)

Hi, I'm trying to diagnose a particularly elusive problem, and am wondering if anyone else here has seen anything similar and can offer any ideas. I have a conference bridge running Asterisk 1.2.32 (with slight mods), in a colo talking via a LAN to an ITSP using SIP/RTP. It is dedicated to a single customer. On several occasions over the last few months, the customer has reported instances

I have a Logitech USB headset which I have used very successfully on Windows for a long time. The other day I thought I would try it out on my CentOS 5 server, which doesn't have its own sound hardware (HP ML110). On plugging it in, all the required modules magically got loaded, and the "play" command would play sound files as expected. However, at all times when playing a sound

I'm just experimenting with dynamic spans using TDMoE, which looks ideal for clustering Asterisk boxes with low latency (i.e. 1ms for TDMoE compared with 20ms for IAX). When using normal E1s and T1s, I've always used ISDN signalling: either EuroISDN or NI2. So naturally I tried that over the TDMoE. I've found that I get a lot of HDLC errors and the D-channel is up and down like a

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

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 system running CentOS 5.9 and DAHDI 2.6.2 with a 2-port E1 card using the wct4xxp driver (also using Asterisk 11.5.0, but that isn't relevant to the question). With DAHDI and Asterisk started, the system appears to run normally, as far as I can tell from limited testing. I am monitoring User, System and Nice CPU usage using SNMP and MRTG, and I have noticed that when I have started

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