Hi,
I have the following two tries. While the first calls out just fine, the
second call does not work.
My asterisk box is behind a NATed fli4l fw. Port 1-2, 5060, 5004
are all forwarded to the asterisk box.
What throws me off, is the fact, that the first call worked fine, and
the
A comment and a question about the latest version:
First, the question: What is the organizing principle of the Asterisk
Minutes chart, with respect to the ordering of the various days? It
seems random. . . I also assume the last column is the average number of
minutes per call?
As to the
I hate to answer myself, but another thing:
The first call worked fine.
This is not true. The moment the other side answers the call, Asterisk
-- Attempting native bridge of SIP/25-e2f9 and SIP/sipgate-e2ee
This obviously can't work. How can I make sure asterisk never tries to
do this?!
I had a similar issue when installing my G.729 licences. I contacted Digium support
and an engineer logged into my system and performed some hocus pocus and got it
working for me ...
-Original Message-
From: Derek Samford [mailto:[EMAIL PROTECTED]
Sent: 25 March 2004 18:29
To: [EMAIL
Currently the asterisk port is blocked due to vulnerabilities in pwlib.
Chris
- Original Message -
From: Joe Lewis [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 25, 2004 9:30 PM
Subject: [Asterisk-Users] FreeBSD Segmentation Fault on start up
To all;
I've got two
Hi,
Having a small problem here and wondering if anyone else has seen it..
My Asterisk box is behind NAT so I need to register with the external
IAX Asterisk boxes for calls to be received..
Up till yesterday I only needed to register with a single external IAX
server and all was working
Well the warning is:
Mar 18 16:46:47 WARNING[737296]: Huh? An ilbc frame that isn't a multiple of 50 bytes
long from RTP (38)?
And the sound is cripple (really broken).
Any solution?
--
Alejandro Escanero Blanco
Administrador Sistemas CEC.
___
Hi List,
Two boxes
A has a PRI
B terminates SIP devices
A --IAX-- B
Both on the same switch, same IP network.
Call from PSTN to A gets pushed via IAX to B - Sip device
with no problems.
Call from Sip device - B via IAX - A - PSTN
will drop exactly 5 seconds after the call
John Todd wrote:
I've finally uploaded the newest (LARGE) list of sound clips in .gsm
format to the bugtracker.
Great thanks a lot :)
Assume they'll make it into CVS tarball sometime soonish.
Please see http://bugs.digium.com/bug_view_page.php?bug_id=985 for
details and a full sound file
Works like a charm for me.
I have both VoicePulse and NuPhone registered in IAX.
Depending upon the phone nr dialed, I send a call via NP or
VP.
And yes, my [*] box is behind a NAT.
Include the relevant lines of your iax.conf so we can take a
look.
Cheers, Willy
- Original Message Follows
this patch however worked for me, all calls through the patched
chan_h323 are ok, hold, transfer, etc works perfectly. except that there
is no music on hold, while in fact asterisk shows that it is playing,
yet there is no audio heard on the callmanager side.
so i had test on both oh323 0.5.10
[EMAIL PROTECTED] wrote:
Works like a charm for me.
I have both VoicePulse and NuPhone registered in IAX.
Depending upon the phone nr dialed, I send a call via NP or
VP.
And yes, my [*] box is behind a NAT.
Include the relevant lines of your iax.conf so we can take a
look.
Cheers, Willy
There is
Gus,
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED]
register = user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
nesesary to
Hi,
I am a newbie and am currently writing an
application for making outbound calls for a reminder service.I am using AGI for
that
Problem isfor SIP calls. As soon as the call
goes through (even ringing), Asterisk says that the call is answered.
Checking CHANNEL STATUS gives me 6 even
Rich Adamson wrote:
Gus,
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED]
register = user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
Hello friends, Just yesterday i joined asterisk mailing list. We want to use it in our company. Can someone tell me which version or type of Linux would best work for it and also what should be the configuration of machine (hardware configuration) to install Asterisk? That would help me a
Eric Wieling wrote:
On Thu, 2004-03-25 at 09:33, Steve Underwood wrote:
exten = 5678,1,txfax(/tmp/testfax.tif|caller)
There are a zillion fax and tiff formats. I'm trying to figure out what
output format I should tell GhostScript to use. Any suggestions on
which format to try?
These are
On Thu, 25 Mar 2004 23:29:52 -0600, Christian Hoffmeyer wrote:
- Original Message -
From: Clif Jones [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 25, 2004 8:06 PM
Subject: Re: [Asterisk-Users] Semi OT: WiSIP and WEP
Are you saying that the clicking only occurs when
Hi Reynaldo,
The is the only report of seg faults I have had with recent versions of
spandsp. There was an older version (spandsp-0.0.1b I think) which had a
silly bug that caused seg faults.
Some people have had older versions of libtiff installed, which seem to
cause seg faults. If you are
There is a calling card AGI script that you can probably find in the
archives by searching for calling card agi. Also I posted a good
little script for unsupervised call transfer of incoming calls some time
ago. I'm sure people have improved it since then. The only thing you
need is a centrex
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED]
register = user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
nesesary to register
Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I
have asterisk on public side and phones on the private side. I am able to
get the phones to register and make outbound calls but the inbound calls are
intermittent. I have NAT enable in asterisk and on the Cisco 7960.
Any
Rich Adamson wrote:
Thats the problem I have a dynamic IP on my side which is why I need the
register line in the iax.conf..
iax2 debug shows that it is registering the first line but not the second..
I assume you've tried the easy stuff... separate the two statements with
some additional
Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I
have asterisk on public side and phones on the private side. I am able to
get the phones to register and make outbound calls but the inbound calls are
intermittent. I have NAT enable in asterisk and on the Cisco 7960.
I
That sounds about right.
I sometimes use my PDA with xlite installed on it, and I get clicks when I
am using 128 Bit wep.
I now have 2 access points. One uses 256 Bit WEP and is only for my .g data
network, and then I have a Orinoco rg-1000 in access point mode and the
provides access for my pda.
Yash,
Just yesterday i joined asterisk mailing list. We want to use it in our
company. Can someone tell me which version or type of Linux would best
work for it and also what should be the configuration of machine (hardware
configuration) to install Asterisk?
I did it on Debian 3.0 with
Ugh,
I didn't have a chance to try it last night. I guess I'll have to pull out
my old Apple 802.11b AP and use that for the phone and/or non-secure
connections from friend's laptops, etc.
Good real world experience, is there a way to feed this back to Jeff,
maybe for a WiSIP v2?
-Original
Hi all,
Sorry to bother you all with this, but I'm sure I've done something stupid.
I've followed the instructions at:
http://www.voip-info.org/wiki-Asterisk+phone+sjphone (substituting 100 with
377 - the extention I was for this softphone) but when I try to register the
softphone client I get
In you iax.conf file, are you using type=friend? I seem to remember a
discussion about problems if using type=friend instead of type=user for
multiple registered servers with the same userid. I may have the details
messed up on this though!
I have the same setup your describing and it works
Jakob Strebel wrote:
Yash,
Just yesterday i joined asterisk mailing list. We want to use it in
our company. Can someone tell me which version or type of Linux would
best work for it and also what should be the configuration of machine
(hardware configuration) to install Asterisk?
I did it
Hi,
I have just solved it, I deleted the iax.conf and created a new one
(exactly the same) and now its working.. I guess there was a gremlin in
the text file somewhere the was twisting things up..
Thanks to everyone for you help and suggestions..
later..
Ed Rubright wrote:
In you iax.conf
I have an 8 channel isdn30e coming in from BT. Can
anyone point me to sample zap*.conf that will work. Thanks JC
- Original Message -
From: Adams, Gavin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, March 26, 2004 9:01 AM
Subject: RE: [Asterisk-Users] Semi OT: WiSIP and WEP
Good real world experience, is there a way to feed this back to Jeff,
maybe for a WiSIP v2?
I do not believe that
All,
I figured it out Typo in the secret field... How embarrassing...
Matt
-Original Message-
From: Matt Bridges [mailto:[EMAIL PROTECTED]
Sent: 26 March 2004 15:04
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie Softphone Problem
Hi all,
Sorry to bother you all with
On Thursday 25 March 2004 15:30, Joe Lewis wrote:
I've got two installations of asterisk. The last one (installed a
few days ago) is from the FreeBSD ports, and many thanks, because
it compiled BEAUTIFULLY! However, I can't run it. Everytime I
start asterisk, I get a segmentation fault.
Our cisco router have these dial peers:
dial-peer voice 900 pots
application session
destination-pattern 5000
port 1/0/0
!
dial-peer voice 800 pots
application session
destination-pattern 9
port 1/1/1
!
dial-peer voice 701 pots
application session
destination-pattern 3003
port 1/0/1
!
dial-peer
Fran Boon [EMAIL PROTECTED] wrote:
I've finally uploaded the newest (LARGE) list of sound clips in .gsm
format to the bugtracker.
Great thanks a lot :)
Assume they'll make it into CVS tarball sometime soonish.
Out of interest, how much did it cost to get all those clips recorded?
--
Hello all,
Reading the varios sources of documentation I can only find reference
to one FXS USB adapter being supported by asterisk: the Digium S100U. Is
there other alternatives? I´ve found that http://www.tjnet.com/ makes a voip
chipset that has been used on a lot of internet phones
Solved my problem, but... you may not like the way I did it.
I dived into the yacc code for the humble ast_expr, and found a few
nits, and cleared them up, and submitted a bug report with a patch:
http://bugs.digium.com/bug_view_page.php?bug_id=0001292
Now, here's my complaints about the $[ ]
Title: Message
OK...I've got an *
box with a T100P in it. For the most part incoming calls are going through
just fine. Outgoing calls, however, I'm having some more trouble
with. Whenever I make an outgoing call, the call begins, however after the
dialing process all I hear is dead air.
yes, the 7960 is sending the right digits, because in message log from
asterisk I can see each dtmf. A brief message log is below:
Mar 25 19:28:33 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0
(SIP/2010-9164)
Mar 25 19:28:33 DEBUG[1200826048]: Bridge stops bridging channels
Mark Messmore, Technical Support, University Telcom Inc. wrote:
-- Executing Dial(SIP/mark-2d08, Zap/g3/2550559) in new stack
-- Called g3/2550559
This looks like its using group 3 (g3)
zapata.conf
group=5
And this looks like the only group definition in zapata.conf
Where is group 3
Yeah...sorry about posting the wrong group...I've been doing a lot of
testing with different groups and settings just trying to get something
to work...here's group 5.
context=conference
signalling=em_w
switchtype=5ess
group=3
callgroup=3
pickupgroup=3
channel = 6
busydetect=yes
callprogress=yes
Hello asterisk experts,
I have a running installation with a Cisco 7960 and an ATA186.
Attended and unattended transfer of an incoming PSTN call from 7960 to ATA works
as expected.
From ATA to 7960 users can press the flash button, dial the 7960, talk to the
other ext. and should then be able
Can any one help me with an error im getting with
asterisk? I have VoicePulse Connect and Nufone and when i try to make a call out
on VoicePulse i get the follow error:
Mar 26 14:34:24 NOTICE[196624]: app_dial.c:545
dial_exec: Unable to create channel of type 'IAX2' == Everyone is busy
at
There is a USB handset based on the TigerJet 560/560A chip that is being
marketed as a Rocket Phone... VID 06e6 PID 0210... that is the same unit
as the Digium S100U. The wsfxs driver even recognizes it as the digium
hardware.
- Original Message -
From: Gelson Dias Santos [EMAIL
Dear Scott
i have made every possible combination from timing source to cvs version is
possible but nothing i speack a lot in chat with nice guy about it but
nothing... the cable is not the problem im sure of it because i use Trend
Aurora Sonata Tester (..but im not good telecom
Hello,
I'm pleased to announce that we are finally up to version 1.0.0 with the
astguiclient suite.
The biggest changes to the suite have been to make it easier to install and
the addition of a complete from-scratch installation instruction document.
you can see screenshots of the gui clients
Hello,
http://astguiclient.sourceforge.net/scratch_install.html
That is the address of our new in-depth step-by-step instructions for
installing Asterisk on a blank machine all the way through the installation
of Linux, MySQL, Apache/PHP, Asterisk and the astguiclient suite. All with
complete
Hi,
I want to add a new command to the astman interface which would allow one to
call any AGI application/command using astman. As I'm new to the Asterisk
source code, I would like to know if there's any caveats that I should be
aware of before I start coding it. Such addition will help to
After the last FreeBSD-hostile response from someone on the list, I was
wondering about something someone else said a few weeks ago: that something
of the magnitude of a *real* Asterisk under FreeBSD project would probably
require its own mailing list (and perhaps its own project website). Has
Anyway, in my P.S. yesterday (the main post was on Codec problems), I
described a situation where any IAX softphone was registering
successfully, and then having zero sounds heard on either side of the
call. Here is an iax2 debug output from a DIAX call to a local *
server, dialing the
I'm having trouble determining which ISDN4Linux devices are usable in
the US. I want to integrate ISDN into my Asterisk PBX. My circuit
provider is Qwest.
Does anyone have a working ISDN BRI interface in the US? Does the fax work?
Thanks,
--
Michael Welter
Introspect Consulting, Inc.
Hi Dimitri-
I'm not sure, but it looks like you have too many clock sources.
Assuming that the Sonata can serve as a clock source:
Try this in zaptel.conf, (zapata.conf info is shown too)
BOX 1: (with the TE410P)
Span 1 (connected to the sonata): span=1,0,0,ccs,hdb3,crc4,yellow (and
Michael,
I won't be much help because I am just a couple of steps ahead of you,
but I will try. It looks like there are not many people in the U.S.
using BRI. Because it is so unpopular here, there are not many cards
available that work. The key to getting a card that might work here,
is if
But I thought asterisk was just a console application, with no need for
XWindows at all. Or is it on hold due to the GUI controlling
mechanisms?
Joe
Chris Stenton wrote:
Currently the asterisk port is blocked due to vulnerabilities in pwlib.
Chris
- Original Message -
From:
- Original Message -
From: Derek Bruce [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, March 26, 2004 2:43 PM
Subject: Re: [Asterisk-Users] Supported USB adapters ?
The wsfxs driver even recognizes it as the digium
hardware.
wcusb ?
Christian Hoffmeyer
YottaDot Solutions
Joe Lewis [EMAIL PROTECTED] wrote:
(Article converted from unnecessary HTML to nice plain text.)
But I thought asterisk was just a console application, with no need for
XWindows at all. Or is it on hold due to the GUI controlling mechanisms?
You don't have to run Asterisk from the console.
It was in fact the problem.
When I wanted to test the MeetMe feature, I have installed ZTDummy but did
not recompile Asterisk program after!!!
Thanks for your help.
Franck
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Olle E.
hi all,
I'm looking for a config to bind asterisk and hylafax.
Asterisk just have to pass the line to hylafax in the dial plan.
I don't have any digium hardware, but this is certainly possible.
I got it with capi4hylafax, but this config is not acceptable in a large
system, the call is not
I recently purchased a tigerjet usb phone with the TIGER560B chip($40). I
needed a portable phone that avoided my crappy laptop soundcard. Their
salesman said the phone is supported on linux and asterisk support would be
coming... When installing the linux drivers, the make crapped out. After
After months of delay, I am releasing a second Beta of IAX Phone. This
version looks just like the prototype I have had on display for several
months. It includes a number of new features and shows signs of others that
will be coming soon.
New Features:
- Speaker phone
- Muting
Hi,
-Original Message-
I have a running installation with a Cisco 7960 and an ATA186.
Attended and unattended transfer of an incoming PSTN call
from 7960 to ATA works as expected.
From ATA to 7960 users can press the flash button, dial the
7960, talk to the other ext. and
Hi. I have posted a fix for announce so that it does not stop the music on hold until
after playing the
announcement file. If you can, please test it out for me.
Thanks,
Kevin
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
After some sleep, google gave me some additional information:
Wade said before that MWI is done by FSK and voltage-type MWI is not
supported:
http://lists.digium.com/pipermail/asterisk-users/2003-August/018426.html
For the existing MWI to work, voltage on the line needs to drop for a
fraction
William C. Ray [EMAIL PROTECTED] wrote:
Can any one help me with an error im getting with asterisk? I have
VoicePulse Connect and Nufone and when i try to make a call out on
VoicePulse i get the follow error:
Mar 26 14:34:24 NOTICE[196624]: app_dial.c:545 dial_exec: Unable to
create
I just received my first Cisco 7960 today and was looking forward to
playing with it this weekend, however I can't seem to get it working via
skinny (can't find any information via the wiki regarding what needs to
be on the tftp server for skinny). I would like to get my hands on the
SIP images
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