similar to: Temporary loss of audio on all SIP channels

One of our customers has an Asterisk conference bridge connected to a SIP trunk from an ITSP. Yesterday, they had two inbound calls that didn't get hung up properly. From the tcpdump SIP trace that we have running continuously, I can see that no BYE was received by the bridge, and when some hours later the hangup was forced from the bridge end, the bridge sent a BYE to which it received a 481

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

Most of my experience until recently has been in Asterisk 1.2, and I am just starting to make use of Asterisk 11 for new systems. I have a question about using SIP on a multi-homed machine. I have a customer who wants an Asterisk box with two network interfaces: one on the public Internet (no NAT), and one on a private LAN. The box will not do any IP forwarding between interfaces. They want to

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 very specific, and I don't need right now to discuss the ins and outs of the above setup. What I am seeing is that when I have set up 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 =>

I have some Asterisk boxes in my SIP provider's colo, and the SIP connections are done without using any kind of authentication credentials. It's a secure network, so I guess that's OK. I now want different sets of calls from one box billed to different customers, by the providers billing system, and the provider says that the distinction can only be made by having the differnet sets

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

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'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)

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 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 have some more investigation to do on this, but I wanted to see if anyone here had any insight into the issue I've run into. The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one of several systems that have been running without issue since 2010/2011. They have all been running CentOS 4 32-bit with Zaptel 1.4.12.1 (with patch for gen 5 card), libpri 1.2.8 and

I have Asterisk running voip-only on my colo server, and have subscribed to the Voiptalk PSTN->SIP service in the UK. All has been working fine while I have had incoming calls going straight to an phone extension. I am now trying to put in a simple IVR Welcome script. I have found that I cannot receive DTMF keypresses from the incoming PSTN caller. Nothing registers on the Asterisk server, and

In article <CAHZ_z=w5DMg93gShtC93kuC+fnmraPgV46BS956U5BQXVgyhxg at mail.gmail.com>, Matt Fredrickson <creslin at digium.com> wrote: > On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield <tony at softins.co.uk> wrote: > > I have some more investigation to do on this, but I wanted to see if anyone > > here had any insight into the issue I've run into. > > >

I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks. The application relies on a DTMF digit string sent by the phone after the call has connected. This DTMF is detected by Asterisk under the control of WAIT FOR DIGIT commands send from an AGI processor over a FastAGI connection. Usually the DTMF is detected without error, but on a significant minority of calls, Asterisk is missing

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

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

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