Displaying 20 results from an estimated 10000 matches similar to: "asterisk log - "=======" extension not found?"
2011 Apr 20
2
Call files or AMI originate for mass outbound call
Hello Guys,
In the case of a multiserver environment for outbound
automatic calls, can you share you experience and preference between call
files and ami originate ?
thanks
--
*Adolphe CHER-AIME
Network / VoIP Engineer
CCNA, CCNA VOICE, Global VSAT Forum Certified
(509) 3449-4280*
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2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2012 Sep 26
1
Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
Hello,
I'm having issues connecting throu PRI with the following error "Requested
transfer capability: 0x00 - SPEECH"
Below are the logs:
== Using SIP RTP CoS mark 5
-- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003",
"CALLERID(num)=xxxxxxxxx") in new stack
-- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",
2011 May 19
1
v1.8.4: Extension Not found in Context?
Hello All,
This is probably another one of those completely silly questions that I'm
going to hit myself later on, but I have the simplest issue right now but I
can't figure out why it's happening:
I have a trunk from one * box (box a) to another * box (box b)
the call comes in from box a with an extension 2222 which acc. to the peer
config in sip.conf is set to use context
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
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2012 Jun 11
1
Differences between PBX and SBC
Hello,
I would like to know the difference between encrypt the rtp and signaling
between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm
trying to understand whether an SBC could fit an Asterisk deployment
Thanks
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2011 Jan 19
2
Asterisk extension not found problem...
Hi All,
I am using Asterisk for one of my projects in OpenBTS. I am having the age
old problem of "extension not found" when try to make
a call from one registered SIP phone to other registered SIP phone (two
mobile phones connected to Asterisk via OpenBTS).
The exact error thrown on Asterisk CLI is
*"chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to
2015 Jan 28
2
queue show <queue-name> vs queue log for calculating average hold time
Hi
We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for
queues.
For a particular customer, when I run queue show <queue_name> I get the
following numbers:
<queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s
holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s
So from that data we look at
17s holdtime
And assume that is the
2010 Oct 23
2
Just Take dCAP at Astricon?
Since it is Saturday evening (7PM EST) I am asking this on the list in case
someone who knows sees it and can answer.
Astricon is in my back yard for the first time, and I could hit you with a
rock. I would always like to attend, and spoke at the 2007 Astricon in
Phoenix but don't have the idle cycles.
Question: Can I just go to Astricon and take the dCAP exam only? In and
out? Cost?
I
2010 Oct 21
10
Asterisk 1.80-rc5
Just done a clean install of rc5 on a totally new machine and found the
following:-
/etc/init.d/asterisk start
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
Dave Cotton
2012 May 03
1
AMI disconnects
Hi all.
I've got a perl script that connects to Asterisk's management interface using Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to asterisk restarts and sip reloads.
However, my script gets disconnected quite frequently, causing false alarms in my monitoring.
Here's what the code looks like:
2011 Apr 08
4
IAX2/0.0.29.199
Where this revers IP comes from ?
== Using SIP RTP CoS mark 5
-- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack
-- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack
-- Hungup 'IAX2/0.0.29.199:4569-5255'
-- Executing [s at
2010 Jul 24
4
getting some segmentation faults with 1.8
I downloaded the latest 1.8 (27922) but got some segmentation faults.
The first one was when it loaded cdr_odb, and so I changed menuselect
not to compile that one, but the second one was when it tried to load
chan_agent and so I stopped there to see if anyone else was seeing
this. The agents.conf is all commented out except for [general] .
Anyone know what is happening?
Thanks.
P.S. I deleted
2010 Oct 22
5
dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook?
2010 Oct 14
6
Audiocodes firmware
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body text="#000000" bgcolor="#ffffff">
<font size="+1">Does anyone have links to the most recent audiocodes
2012 Jul 28
3
Asterisk on Dynamic IP to a SIP extension
Verizon has put another good third party DSL supplier out of the DSL business. Their mindset is to kill the competition and then kill DSL and copper althogether in FIOS areas.
So am soon losing my static IP and I need to prepare for the change. I currently have Asterisk running using, besides local extensions, a remote SIP extension in another state. In the new configuration both Asterisk and
2011 Jan 07
3
Definations of READ/WRITE parameters of manager.conf contexts?
Hi Everyone,
I want to know each and every parameter's detail that can be included in
the
read=
write=
in manager.conf
Where can I find this?
Thanks
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2010 Dec 20
3
Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
Hi All,
I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend.
My dialplan:
exten => _XXXX,1,Dial(SIP/${EXTEN},60,rt)
Now, when I Dial extension 1050, and there is no 1050 peer registered I got:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to 0.0.4.26:5060 returned -1: Invalid argument
In 1.6 there was no problem, I have got Channel is
2010 Nov 12
3
Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Hi All,
I'm having an issue where Asterisk continuously sends out a GAZILLION "SIP NOTIFY" messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine.
I'd like to confirm the layout of the
2011 Mar 25
3
reload command not availeble asterisk 1.8.x
Hey Guys!
I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ?
satish-desktop*CLI> core show version
Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC
satish-desktop*CLI> re <tab><tab>
realtime reload
shirley*CLI> core show version
Asterisk