Connect to the Asterisk console with verbose turned on and try to dial.
Post that output.
Curt Shaffer wrote:
This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone
line is
connected to the right port. No luck. Thanks.
-Original Message-
From: [EMAIL
Greetings-
Was running the Digium FreeBSD g.729 codec until recently when the latest
Asterisk bits were obtained via svn:
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
This produced:
Checked out revision 30652
This on FreeBSD 6.1-RELEASE
Attempting to start asterisk it
Hello Michiel,
* Michiel van Baak [EMAIL PROTECTED] [27-05-06 17:15]:
You have to do a couple of things:
1. Open your firewall so it allows the protocol you want to
use.
ok, that should be easy.
2. Configure asterisk to accept guest calls
3. Configure asterisk to ring some phones when
Dear all,
I have an Asterisk box running [EMAIL PROTECTED] 2.7 . and A2billing. when my
asterisk box dial using
dialcommand_param=|45|HL(%timeout%:61000:3) its working fine .
but when i use dialcommand_param=|45|L(%timeout%) call got drop
after 62 seconds.
i used this same setting into
On 13:19, Sun 28 May 06, Matthias Fechner wrote:
Hello Michiel,
2. Configure asterisk to accept guest calls
3. Configure asterisk to ring some phones when someone dials
your domain.
and how is this working?
Is the person who want call me dial [EMAIL PROTECTED]
How can ppl reach me if
Hi!
I know that is not SER discuss, but probably some of you faced with the same
problem:
to detect trunk status (ok/unreachable) in *, it's a must, to set qualify=yes
as * connecting to SER, it's not replying to qualify messages, so even i can
use it well without qualify, with qualify it's says
I think asterisk dropping you to s-BUSY, s-CONGESTED, s-UNREACHABLE
priority, better have a look there
(you can play a busy tone, or playback(called-party-is-busy))
A few employees have noticed some problem here and there when trying to
make outgoing phone calls. After it happens, they try
|20060528-110627|1148832387.1) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060528-110627|1148832387.1: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(SIP/103-a555, No recording needed
|20060528-110627|1148832387.1) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060528-110627|1148832387.1: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(SIP/103-a555, No recording needed
Kim Culhan wrote:
Was running the Digium FreeBSD g.729 codec until recently when the latest
Asterisk bits were obtained via svn:
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
MAYBE it is the same problem:
http://lists.digium.com/pipermail/asterisk-users/2006-April/147577.html
If someone here happens to have a mpg123 binary compiled for Centos 43
in a Pentium Dual Core, let me know.
Somehow mpg123 cant compile.
[EMAIL PROTECTED] mpg123-0.59r]# make linux
make CC=gcc LDFLAGS= \
OBJECTS='decode_i386.o dct64_i386.o decode_i586.o \
audio_oss.o
-- Executing GotoIf(SIP/103-a555, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/103-a555,
recordingcheck|20060528-110627|1148832387.1) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060528-110627
Using sip connections some peers are not able to transmit or recieve
audio. All peers are setup the same aside from the NAT settings. The
call will go through, called device will ring, but when it answers there
is no audio connection. From the callee, they will not here the rings,
only
You need to describe your NAT setup more.
One thing to try is to set qualify to yes or a short number.
Essentially a keepalive for any routers in the middle. If you have
multiple phones behind a remote NAT, make sure they are using different
ports.
Miles Scruggs wrote:
Using sip
I can't imagine why Gnu's assembler compiler wouldn't work for i586
instructions, but does mpg123 perhaps need NASM instead?
No guarantees, but I was toying with installing a CentOS 4.3 box today
anyway, maybe I'll see if I can get it to compile for me and let you
know if and how. I
Jerry Jones wrote:
Create a contact entry with their extension and enable buddy watch on it
It will then show up on an unused line key
On May 27, 2006, at 3:26 PM, Faris Raouf wrote:
I've somehow managed to battle may way through hinting issues with
type=peer type=friend and various other
-- Executing Macro(SIP/103-a555, record-enable|103|OUT) in new stack
-- Executing GotoIf(SIP/103-a555, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/103-a555,
recordingcheck|20060528-110627|1148832387.1) in new stack
-- Launched AGI Script /var/lib
Hi,
is there any way to increse the buffer or something to make SIP
connections sound better? When I make the calls with Asterisk as a SIP
client (through sip.voipbuster.com) the sound quality is poor -
constantly breaking (there are few occasional seconds when the sound is
OK)- but with
Hi,
I'm running * 1.2.7.1 on Red Hay 9.0 w/ a TDM 400 2 x FXS, 1 x FXO.
I'm using a VTech cordless that makes three short beeps when someone
another extension is picked up, presumably this lets you know if
someone is trying to listen in..
Everything works, except the VTech now makes the three
On Fri, 26 May 2006, Guido Hecken wrote:
We had the same problems with some cheap LevelOne Switches.
The Snoms rebooted during a call, calls dropped etc.
Replacing the switches was the solution.
A switch should NEVER cause ANY device to lockup, ever. Period.
If a phone locks up / reboots due
On Fri, 26 May 2006, Rich Adamson wrote:
Dr. Michael J. Chudobiak wrote:
Blaming the 3com switch is very likely to be the wrong root cause. High
probability the 3com was not configured properly for the phone.
Just curious - what configuration issues did you have in mind?
A partial list of
On Fri, 26 May 2006, Remco Barende wrote:
There is just no valid reason why the phone would need to lockup or reboot
even if the network connection would be problematic, no matter what. That is
just poor design, not a feature.
I agree 100%. No device should ever lockup or reboot due to a
Sure that's not the message waiting stuttering indicator?
Terrelle
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: Sunday, May 28, 2006 12:40 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Analogue phone w/ TDM400
Hi,
Hi,
to increase the security for remote extensions I would like to limit a
sip-peer to a specific MAC address. Is it possible to hook into the
authentication mechanism in asterisk and allow/deny incoming registrations?
cheers
urban
___
--Bandwidth
On 5/28/06, Matthias Fechner [EMAIL PROTECTED] wrote:
If there is a site or howto etc. available it would be a pleasure formy to get something to read :)
Go to www.voip-info.org you'll find all you need to know and more.
-- Lacy MooreAspendora, Inc.
Hi
List,
I'm looking for a
Asterisk radius module ... Anybody has one ?
Thanks,
Oliver
Oliver
VermeulenWorld Venture Group
Telecom
Corporate
Address:Str Avionului
Nr 35/bl16J/3Bucharest, 014333 RomaniaOffice: +(40)21-569-4700Office2:
+(40)31-860-0030Fax:
+(40)31-860-0031USA
I'd change s,104 to something along the lines of a playback for debugging purposes just to be sure, but it looks as though all of your channels are busy. The way I am reading that is that you have 4 voice channels on your PRI, is that correct? Could you already have 4 simultaneous calls going on
You could try pri debug span 1 and watch for anything that looks strange.
Lacy Moore - Aspendora wrote:
I'd change s,104 to something along the lines of a playback for
debugging purposes just to be sure, but it looks as though all of your
channels are busy. The way I am reading that is that
Hi,
Has there been any improvements to this
patch?, what is its state now?. Has anybody tested this?. Any results?
I tried the link, seems the site is not up. Where can I download the patch from?
-Vij
On 3/3/06, Juan Salas [EMAIL PROTECTED] wrote:
I will
try to test your adaptation.
Hello,I work for a company that is experimenting with the implementation of Asterisk. We have a VoIP provider that is giving us a demo account with 200 minutes on it. We can register with their service but cannot complete calls with Asterisk. We can use a Grandstream GXP-2000 with the supplied
We need your dialplan and output from the console to help.
Thanks,
Steve Totaro
Steven Haldeman wrote:
Hello,
I work for a company that is experimenting with the implementation of
Asterisk. We have a VoIP provider that is giving us a demo account
with 200 minutes on it. We can register
How about this one?
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
--VoIP StreetDID origination serviceswith support you can count
on!http://www.VoIPstreet.com
- Original Message -
From:
Oliver
Vermeulen
To: asterisk-users@lists.digium.com
Cc:
hi,
I want to complete asterisk configuration from database(MYSQL),now I come
across some doubts:
1.
http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+databasediff2=3
says that Dynamic 'friends' (Asterisk v1.0.*) and the number of options
supported by this
hello,,
Yes, asterisk can use realtime mode
On 5/29/06, 吴应芳 [EMAIL PROTECTED] wrote:
hi,
I want to complete asterisk configuration from database(MYSQL),now I come
across some doubts:
1.
http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+databasediff2=3
says
3.Is there any other way to complete asterisk configuration from database?
Have a look at this : http://www.voip-info.org/wiki-Asterisk+RealTime
hth
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Asterisk-Users mailing list
To UNSUBSCRIBE
Hello,
Does anyone had tried to configure asterisk server as sip client to
connect to go2call service.? If it works can you share your sip.conf
and extension.conf configurations.
Thanks,
Leo
___
--Bandwidth and Colocation provided by Easynews.com --
to increase the security for remote extensions I would like to limit a
sip-peer to a specific MAC address. Is it possible to hook into the
authentication mechanism in asterisk and allow/deny incoming
registrations?
This would be only mildly useful on the same subnet and completely useless
The sip debugging info is here http://pastebin.com/744005, the sip.conf, extensions.conf and consoleoutput are here http://pastebin.com/744065.Thatnk you, StevenSteve Totaro [EMAIL PROTECTED] wrote: We need your dialplan and output from the console to help.Thanks,Steve TotaroSteven Haldeman
Got 1 issue I can't seem to knock out of this particular box.
The IVR works fine on the zap channels and the incoming SIP routes. But
coming in via the IAX2 route leaves me with a silent phone.
The prompts all work still letting me navigate the menu. But just can't
hear anything.
This is
Henry J. Cobb wrote:
to increase the security for remote extensions I would like to limit a
sip-peer to a specific MAC address. Is it possible to hook into the
authentication mechanism in asterisk and allow/deny incoming
registrations?
This would be only mildly useful on the same subnet
Never used it, but I knew some sort of Asterisk
radius thing existed so I searched the Wiki for it and replied. Sorry it doesn't
work for you, good luck with your search.
--VoIP StreetDID origination serviceswith support you can count
on!http://www.VoIPstreet.com
- Original Message
The asterisk host is connected directly to the internet, the phones I am
having issues with are behind NAT, but I'm only having issues with some
of them. Most specifically the phones on my linksys PAP2 adapter. NAT
at the remote location is provided via a standard out of the box config
of a
another questions！
According asterisk realtime sip webpage，I had done following steps：
(1) Make, make install asterisk-addons then copy res_mysql.conf.sample to
res_mysql.conf and edit the res_mysql.conf with my databases parameter
(2) Edit extconfig.conf ---add
sip.conf =
On Sun, 2006-05-28 at 23:41 -0400, Steve Totaro wrote:
Henry J. Cobb wrote:
to increase the security for remote extensions I would like to limit a
sip-peer to a specific MAC address. Is it possible to hook into the
authentication mechanism in asterisk and allow/deny incoming
Hello:
Do you need install Mysql-devel.
Best Regards
- Original Message -
From: 吴应芳 [EMAIL PROTECTED]
To: Asterisk Users Mailing List - No asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 12:04 AM
Subject: Re: Re: [Asterisk-Users] doubts about asterisk configuration
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