similar to: how to read sip debug

I think you need to increase verbose output and search in /var/log/asterisk/full for any error message related to IAX2 registration or simil. 2017-06-05 17:12 GMT-03:00 <thelma at sys-concept.com>: > No, I don't think it is IP table issue, I've not upgraded dd-wrt for a > while and it was zoiper was working OK with my previous version of > asterisk. > > After upgrade

Hi, Oliver. Maybe something like this (add this script to your crontab): ------------------------8<-------------------------- #!/bin/bash # # File: asterisk-watchdog.sh # Date: 2015.05.26 # Build: v1.0 # Brief: Secuencia para monitorizar procesos. # # ${PATH}: Variable de entorno con las rutas a los ejecutables. PATH=/bin:/sbin:/usr/bin:/usr/sbin # ${DAEMON}:

Another nice sip packet is sngrep Shows realtime the sip flows But i think you have to chk the asterisk answer in the dialplan logic to chk what context its hitting etc. ?????? 6 ????? 2016 10:05 PM,? "Steve Edwards" <asterisk.org at sedwards.com> ???: > On Wed, 6 Jul 2016, Victor Villarreal wrote: > > If you experience problems with inbound calls from a SIP trunk or

> On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards <asterisk.org at sedwards.com> wrote: > Now that the g729 patents have expired, how do we use g729 in > Asterisk? > > Will Digium be releasing a g729 codec for 'free' use or do we > download the 'free' codec off the Internet now that we can use it > without moral or legal

True agree, problem is somehow the people purchased.... am supporting to overcome that. Trying level best... around 20 phones has been purchased.... ?? On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, <mefhigoseth at gmail.com> wrote: > With all the money you plan to invest in firmware, licenses, etc., you > have bought a Grandstream IP phone or Yealink... > -- >

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am

I was debugging a turnup with Global Crossing the other day and they presented me with a web page that displayed a 'ladder diagram' of a call including a ton of detail all neatly organized in tabs and links so you could drill down to any level of detail needed. The copyright notice says 'Copyright? 2008 Empirix.' Is there any free software available to analyze a pcap or

AFAIK g729 patent is expiring sometime in 2019-2020. Mitul Limbani On Feb 8, 2017 5:02 AM, "Victor Villarreal" <mefhigoseth at gmail.com> wrote: > Hi Steve, > > I understand your question and your point, but I use the g729 codec from > the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13 > without a single problem. > > So, sory but I

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hello, There is a built-in tool in Wireshark for this : menu Telephony => Voip Calls, the select your call and click on "Flow Sequence". Best regards Jean Aunis Le 17/01/2017 ? 12:27, Yves a ?crit : > Hi, > > I am using pcapsipdump for debugging sip calls. > > when I have to debug a call, pcapsipdump generates two files per > call... one for the sip dialog

On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

with the slip of a finger, i destroyed by extensions.conf (grep -i > extensions.conf) I have a backup that is dozens of hours of code old. is there a way i can use the asterisk cli (or some other asterisky method) to recreate that extensions.conf ?

We have a problem in that calls are dropped after 15 minutes (on both internal and out going calls, incoming calls do not seem to have that limit) How do we fix it? This is the version on that PBX Kernel Linux(x86_64)-2.6.18-371.1.2.el5 Elastix elastix-2.4.0-8 elastix-a2billing-1.9.4-5 elastix-addons-2.4.0-10 elastix-agenda-2.4.0-14 elastix-asterisk-sounds-1.2.3-1

Hi Steve, Thanks for the answer. Since that's what we already have configured, any idea why it wouldn't work? As I said, when "asterisk -rx 'logger reload'" is run it still rotates the log file. On Wed, 20 May 2020 at 18:37, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Wed, 20 May 2020, David Cunningham wrote: > > > We have an Asterisk

Now that the g729 patents have expired, how do we use g729 in Asterisk? Will Digium be releasing a g729 codec for 'free' use or do we download the 'free' codec off the Internet now that we can use it without moral or legal restrictions? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com

hi all, can anyone give me a guide on any asterisk admin solution / interface for config management, and monitoring? No database use is intended and I prefer open source. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

I googled about a bit without success, so... Is there a version matrix available? Something that would say: for kernel version w, you can run up to version x of Asterisk, DAHDI version y, and libpri version z? For example, I have a bunch of remote hosts running kernel 2.6.26, Asterisk 11.6.0, and DAHDI 2.7.0.1. We experience occasional Asterisk crashes, so I'd like to get as up to date

I have two problems. The first is that when I dial my number from a phone on the Internet or any phone outside my LAN, Asterisk does not respond in any way, which means somehow my system is not picking up the fact that there's an incoming call to it. The second problem is that I thought I'd try an internal phone to see if I could get the hello-world stuff working at the least. I

On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of