Displaying 20 results from an estimated 70000 matches similar to: "failing to start asterisk on centos7"
2005 Jun 15
1
app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP -> SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give
2004 Dec 17
6
Realtime and PostgreSQL
Has anyone had any luck with PostgreSQL and Realtime? The realtime
instructions on voip-info seem pretty straight forward... just not woking for
me. I've included all of my config files below, and my console output.
Entire console bootup output:
[root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing
2019 Jul 05
2
Asterisk and Linphone
I have no speex translation
ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24
slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw
ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000 17000 17000 15000 15000 17250 15000
alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000
2020 Jun 13
5
Voice "broken" during calls
Am 13.06.2020 um 13:47 schrieb Michael Keuter:
Hi
> Try "sip show peer <peername>" for a phone.
So:
mobile phone:
bpi*CLI> sip show peer 0049177xxxxxxx
* Name : 0049177xxxxxxx
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Record On feature : automon
2006 Apr 29
2
Codec G729 no longer works.
I upgraded my server from Fedora Core 4 to Fedora Core 5.
I was wondering if anybody else has run into the problem and know's the
fix?
I recompiled asterisk and if I don't have
the /usr/lib/asterisk/modules/codec_g729a.so
file in place it works.
I use or used to use the licensed G729 Codec from Digium.
This is the error message from asterisk -vvg:
[app_playback.so] => (Sound File
2004 Aug 06
2
Asterisk not starting
Hello!
Asterisk "CVS-HEAD-08/06/04-14:55:13" won't start on two of three different
Gentoo machines. This is the output of gdb:
ultra asterisk # gdb /usr/sbin/asterisk
GNU gdb 6.0
Copyright 2003 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
2009 Dec 30
2
Skype for Asterisk
Hi Sir,
We have integrated Skype with Asterisk (skype user id:-
rexesbposolutions). Each call which is coming to skype account is
getting transfered to Asterisk Queue. It has following two cases:
case 1: When we call from normal skype account to skype account
(rexesbposolutions), everything is working fine.
case 2: This skype account (rexesbposolutions) has been assigned with a
online virtual
2015 Sep 30
3
Change Asterisk MulticastRTP codec
Greetings everyone,
I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream.
In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information.
I have noticed that when I do a MULTICAST page and send data
2013 Jan 10
1
Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2
After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault.
[root at localhost asterisk-11.1.2]# asterisk -vvvvvvc
Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components
2006 Apr 20
3
Asterisk Won't start after SVN Trunk Update
Hi:
I deleted old modules in /usr/lib/asterisk/modules
before make install. I built zaptel and libpri before
asterisk. Modprobe zaptel and modprobe -v wctdm
executed witiout complaint. Starting asterisk
produced the output below with several warnings and a
failure. Can someone help, please. I double-spaced
the warnings in the text below. The first warning is
about music on hold because it
2014 Sep 23
1
Change codec when dial from SIP to DAHDI
Hi:
I am useing asterisk 11.12.
I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI
use alaw. G722 is great when ip-phone talks to each other. but when
ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to
transcode to alaw.
so I try to change the codec when dial from SIP to DAHDI. I tried
to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP
2019 Oct 03
2
Asterisk not using common codec between (SIP) endpoints
Hello people,
I've ran into two problem that I can't seem to be able to solve on my own.
Here's my scenario (running Asterisk 13.28.1):
In short: - Asterisk behaves unexpectedly (at least to me) when
negotiating between endpoints
that have a different but intersecting set of codecs
(preventing direct media flow).
- Also, when an endpoint sends RTP with an
2004 Jan 17
4
Asterisk Indications
Hi,
Just wondering if someone could better explain how the indications.conf file
actually affects Asterisk?
I am using a Cisco 7940 from my Asterisk system, and have set in
indications.conf "country=au" thinking that this would make the
dialtones/call progress sound like the familiar Australian tones?
However when I call another extension on my system, it still sounds like
2019 Jul 08
3
opus codec
Hi All,
I am trying to get the opus codec working with linphone.
I followed the instructions... This shows me its loaded
core show translation paths opus
--- Translation paths SRC Codec "opus" sample rate 48000 ---
opus:48000 To g723:8000 : No Translation Path
opus:48000 To ulaw:8000 : (opus at 48000)->(slin at 48000
)->(slin at
2007 Jun 05
1
g729
I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call.
Any ideas?
ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies
Jun 5
2005 Feb 19
3
Still asterisk startup crash plz help
Hi,
First i would like to thank the kind people of the list who have
answered my previuos mail, but i am still stuck as asterisk still
crashes upon startup, i have read the install article at
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
and i have search the asterisk archives, but i still cant get asterisk
to work, i have tried reinstalling asterisk but it still complains and
2005 Jul 07
1
Asterisk Crashes after update
After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from
CVS, Asterisk crashes on startup with an apparent MySQL
(res_config_register) error:
# asterisk -vvvgc > asterisk_startup_error1.log
asterisk: symbol lookup error:
/usr/lib/asterisk/modules/res_config_mysql.so: un
defined symbol: ast_cust_config_register
The log is shown below. I've seen the posts
2006 Dec 15
2
call from h323 to SIP
Hi
i am trying to do the same thing:
receive a call from a cisco callmanager and forward it to a SIP user.
Asterisk is compiled with h323 support, and is configured as a gateway
in the cisco callmanager.
h323.conf:
[general]
port = 1720
bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP
address for this machine
allow=all
extension.conf:
exten = 3298,1,Answer
exten =
2005 Aug 23
1
Can't get G729 working after buying a license.
List,
I purchased 2 g729 licenses but I can't get it to answer a g729 call
from a cisco router with a vwic card. In the debug output below you
will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263)
when it should support g729 according to the config also listed below.
The real odd thing is I can place g729 calls to the router, just not
from the router to *. Anyone have any
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723