[EMAIL PROTECTED] writes:
exten = 109/_9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
[...]
as you can see, I'm wanting 109 to dial out broadvoice_1, [...]
You've got the source and target reversed. :-)
You want the Caller ID of your local extension 109 to be 109, and
then you should
Windows messanger
On Thu, 2004-09-23 at 23:47, Florin Andrei wrote:
On Thu, 2004-09-23 at 01:59, Vladyslav wrote:
HI ALL.
Please help.
Problem: video calls drop after 15-20 seconds all the time.
Use * latest cvs.
What are the clients that you're using?
--
Best regards
Vlad
DEMAINE Benoit-Pierre wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
~ I would like to : set up a server on Linux on which my friends can
connect with msn or netmeeting, suporting at least sound conferance, and
optionally video, but I dont want asterisk server to lock up the sound
card; and
Hello,
I've been browsing through this archive and the wiki web
but I can't find any info on how to implement some dynamic
configuration like group joining/leaving from the phone,
or programmnig transfers for one extension from the phone attached
to it.
I'm trying to do things like:
dial *48 :
DEMAINE Benoit-Pierre a écrit :
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
~ I would like to : set up a server on Linux on which my friends can
connect with msn or netmeeting, suporting at least sound conferance, and
optionally video, but I dont want asterisk server to lock up the sound
card;
Hi Mark,
Just a couple of points/requests I'd like to add.
1. Change the Commedian mail to something more generic, like Voicemail
or Welcome to voicemail
2. The password prompt, just says password, would it not be better to be
a bit polite and have something like, please enter your password
Hi all,
There is both GSM and WAV files for the ones I've recorded already on my
website. They can be found here http://www.g7ltt.com/VoIP/vmfiles.html
As for polite files. I agree that they could be better. I also note
that the diction is very American particularly in the demo area.
I've also
Asterisk is a project that proves that OSS works. Through it is also proven
that a business model based on OSS works.
Thanks for the time and effort you put into this proect.
Yiannis Costopoulos.
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On Fri, 24 Sep 2004 00:14:42 -0400, Administrator [EMAIL PROTECTED] wrote:
I think today is a time to say Thank You Mark Spencer and thank you
Asterisk community, so this project is alive and project is booming and
growing like crazy.
And let's not forget to give credit to Jim Dixon, who
I would like to redirect a call from one IAX destination system to another
and I'm wondering if it is possible and if so, how.
Here is an explanation of what I mean:
Point A is our main office. It has the lines from the PSTN and it converts
them to VOIP traffic on an * box.
Point B is a
Adam Goryachev [EMAIL PROTECTED] wrote:
On Fri, 2004-09-24 at 01:12, Kevin Walsh wrote:
I seem to be hoarding patches, and sending them out on request. I
should set up a website to list and share them more easily.
I did, but nobody used it... http://www.websitemanagers.com.au/asterisk/
Hi,
Were going to build an IVR system with a
TE405P and 4 E1. Were sure that the 120 channels will be filled by 120
simultaneous calls during peak, so we want to have the good server to manage
this.
We wonder a lot of things and maybe you could help us.
- Are you ever build a
Hi,
Can, smb help with url, where I can find detailed how-to about Cisco
Phone application development.
thnx
--
Andrew
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On Fri, 24 Sep 2004 10:54:36 +0200, Andre FAURE [EMAIL PROTECTED] wrote:
How would you do this? Is the dial plan enough or should some
Usually when joining/leaving something you create/remove entries in the
asterisk database, and then use these bits of the database elsewhere. On
voip-info.org
If you want the patches in CVS the best place is to put them on the bug
tracker at http://bugs.digium.com then everyone can benefit from it.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Adam Goryachev
Sent: Thursday, September
i am experiencing errors with the rxfax application when receiving faxes
from a 'brother' fax device.
the rxfax application picks up the incoming fax but the subseqeuent
'negotiation' process seems to fail with the messages logged to the asterisk
console as below;
fast carrier up
coarse
apologies as i forget to mention to the receiving device connected to PSTN
is x100p fxo i/f
- Original Message -
From: Graham Turner [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 24, 2004 1:12 PM
Subject: latest cvs / spandsp
i am experiencing errors with the rxfax
Last but not least let's not forget to thank Anthony Minessale (anthm) on
IRC for all the work he has done on the project.
Bridge config
Valetparking
res_perl
res_sqlite
ChanSpy
ControlPlayback
AND many many many more patches that have been commited. I knew anthm and
worked with him on some
Cees de Groot wrote:
On Fri, 24 Sep 2004 10:54:36 +0200, Andre FAURE [EMAIL PROTECTED] wrote:
How would you do this? Is the dial plan enough or should some
Usually when joining/leaving something you create/remove entries in the
asterisk database, and then use these bits of the database
THIS GUY!!! Yes without him and Mark Asterisk wouldn't be what it is today.
GREAT JOB GUYS... we can only get better from here... Now who wants DS3 and
Hardware DSP cards to scale asterisk to the DS3 level? :)
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Hi,
How can I get the Fax Status of transmited document - complete,
error, etc.?
Regards,
Miro.
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Ya dude, I know this is a little late.. But if you took pics put em up
somewhere..
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
813.901.5182x107 Office
813.864.3164 Direct
813.817.9961 Cellular
813.881.9762
You need to make sure the path to the openh323 and pwlib libs are in
your ld.so.conf (or equivalent) file.
On Sep 19, 2004, at 4:12 PM, Trevor Morrison wrote:
HI,
I have the latest RC2 of Asterisk on a RH 9 non-modified-load box. I
have
an Avaya IP phone that uses h323, so I am trying to
Ryan Courtnage wrote:
Hi all,
We have an analog line from telco, on which 3-way calling is subscribed
to. This line is plugged into an FXO module on a tdm400p.
If an incoming call comes in on this line, can */zaptel send Flash to
telco via the FXO module? If it could, then an incoming call
Check that the sip address that kphone needs to register is
sip:247417@ipaddr or fqn of server
with password
xyz123
I also find that using ethereal *really* helps debug these things.
But the real reason I'm writing is to ask: what did you do to get
asterisk to log to /dev/log?
On Fri, 24 Sep 2004, Greg Boehnlein wrote:
On Thu, 23 Sep 2004, Gary Carr wrote:
The RPMs had errors for me
After installing RPMS and running modprobe zaptel I get
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol
register_chrdev_R07a6f6f0
Hi,
I would like to make a simple application with address book which
to dial the numbers and to transfer the call to the caller before the
called party is answered. How can I do that?
Regards,
Miro.
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On Fri, 2004-09-24 at 04:32, Scott Lykens wrote:
[snip]
Then I have a perl script that reads each file in and puts them into a
MySQL table.
[snap]
Would you mind sharing the perl script and the database schema?
TIA,
Patrick
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On Fri, 24 Sep 2004 14:21:17 +0200, Andre FAURE [EMAIL PROTECTED] wrote:
So if I understand correctly, anything beyond
basic actions has to be programmed through
the use of the database?
Well... 'has to be' is a bit strong - there are more ways to skin this
particular cat, like using AGI, the
I use Gnomemeeting directly with Asterisk.
h323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=ilbc
allow=gsm
allow=alaw
allow=ulaw
[arkadi]
type=user
host=62.85.6.146
context=from-h323
extensions.conf:
exten = 6000,1,Dial(H323/[EMAIL PROTECTED])
arkadi.
[EMAIL PROTECTED]
Hi,
Up until now the advice on installing asterisk seems to have been to
checkout zaptel, libpri, and asterisk from
:pserver:[EMAIL PROTECTED]:/usr/cvsroot and then do make install
in each directory.
Is this still the recommended method of installation? Or is it now get
the latest tarred
On Fri, 2004-09-24 at 06:14, Administrator wrote:
Folks,
Today was great day, Asterisk 1.0.0 was released.
Indeed.
[snip]
Once again, please say and post your comments for Mark and our Asterisk
Community. Congratulations to all of us !
Also we have Astericon, thanks to Steve and Olle
On Fri, 2004-09-24 at 14:13, Brian West wrote:
If you want the patches in CVS the best place is to put them on the bug
tracker at http://bugs.digium.com then everyone can benefit from it.
And don't forget to send a signed disclaimer to Digium.
Regards,
Patrick
Hi,
I'm a bit confused about how Asterisk decides in which order of
preference it should list the different codecs in its SDP message during
SIP call setup.
In my sip.conf [general] section I've got
disallow=all
allow=gsm
allow=ulaw
allow=alaw
But when Asterisk
Chris wrote:
Are you able to get MOH working by setting up an extension in your dialplan?
Yes,
[internal-services]
exten = 555,1,Answer
exten = 555,2,MusicOnHold
I use this in my production system to allow users to listen to hold
music while on break.
Thanks,
Chris
I've been asked to recommend a solution for a one-E1-port PSTN gateway
supporting SIP. I've never set up a Cisco 5300 or equivalent, but I know
they work. I use the Asterisk software in a couple of places and would like
to use the E100P. My question is whether anyone out there has any
The main speaker (that I believe is Mark), is looking at a list of post
1.0 wish features. Is this list online somewhere?
--
Andrew Thompson
http://aktzero.com/
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On 23/09/04 16:57 -0700, Dan Clark wrote:
I'm trying to run some inbound test to my Asterisk box using Telesthetic's
gateway in MI to my GNU/IAXtel account.
Am I missing something? I set up my user account on the GNUPhonne site,
configured Asterisk to talk to IAXTel. *
Asterisk works ok, but it have a lot of errors...
1st: It ever handle audio packet, and you cant do for exacmple only
SIGNALLING
2st: It cant handle more than 20 channels simultaneous ... I tested it.
3st: It does not have fully Radius support.-
-Mensaje original-
De: [EMAIL PROTECTED]
On Fri, 24 Sep 2004 09:38:04 -0400
Bill Hamlin [EMAIL PROTECTED] wrote:
I've been asked to recommend a solution for a one-E1-port PSTN gateway
supporting SIP. I've never set up a Cisco 5300 or equivalent, but I
know they work. I use the Asterisk software in a couple of places and
would
I think I've twigged what's going on but I don;t know how to fix it.
Everytime GV sends me a SIP Invite I send them back a 407 Proxy
Authentication Required challenge which they ignore. They offer me
another Invite which I challenge and so on until I dump the call.
Firstly, I'm not sure why I
It's close -- it still requires an FXO port, and is probably not
inexpensive itself. So between the FXO port and the device, you're
probably in for it at $200 or so. I can get away cheaper with a
cell-socket. I'd prefer a bluetooth dongle (1) because of cost, and (2)
because of the sheer
On 24/09/2004 00:27 Roger Schreiter said the following:
and where do I get a Zaptel-version matching
asterisk 1.0?
I only know CVS as source for the zaptel drivers.
this may have been asked before, but how does * release engineering work ?
1.0 has just been released, and would it be correct to
Hello everybody,
I would like to know if there is a support of IAX in
vicidial.
I want to make predictive dialing use vicidial using IAX
soft phones.
Thanks in advance
Lamine
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Hello everyone!
Anyone know if there is a way to use chan_skinny or chan_sccp to emulate a Cisco 7960
and talk to the CallManager. I realize it is intended to talk TO the phone, but I'm
looking for an SCCP softphone solution (OS X). IPBlue has a client for Windows, yes.
So, how much of a
Not yet, but this is the next protocol that we plan on adding(IAX clients)
After that we want to try adding IAX trunks. We should be able to add IAX
clients in a couple months when we get some IAXy's in here to play around
with.
Thanks,
MATT---
-Original Message-
From: Mamadou Lamine KA
1. Turn on CLI sip debug
That will tell you what SIP is doing and why things are not registering.
2. Have you setup the SIP phone properly?
Is the plugged in? ;)
Can you telnet to the phone?
Can you ping the phone?
3. Use CLI interface and try to call the phone
4.
Hi -
I think I might have seen this problem on the list before, so I'm sorry
if this is a duplicate, but I couldn't find it when searching through
the archive
I'm just setting up a new machine with asterisk. It's a RH9 box, and
I've tried the RC2 tarball, the 1.0 CVS and the 1.0 RPM's
Same problem here..
Redhat 9 2.4.20-31.9
John B
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein
Sent: Friday, September 24, 2004 8:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk
Asterisk works ok, but it have a lot of errors...
1st: It ever handle audio packet, and you cant do for exacmple only
SIGNALLING
OP is looking for a c5300 equivalent, afaik c5300 isn't able to do that.
2st: It cant handle more than 20 channels simultaneous ... I tested it.
Retest. Full E1
cell-socket. I'd prefer a bluetooth dongle (1) because of cost, and (2)
Wouldn't it be cheaper a cell-socket + a 30 u$s phone? (how much does a
bluetooth capable phone cost?)
Saludos,
HoraPe
---
Horacio J. Peña
[EMAIL PROTECTED]
[EMAIL PROTECTED]
Hi everybody,
I have still problem with setting-up asterisk.
I use asterisk with gnu gatekeeper and h323 phones.
I read lots of much documents, but there's no any reference to
setting-up how to put on hold an incomming call.
I mean:
1.) somebody call me from PSTN (via my ISDN BRI card in
On Fri, 24 Sep 2004 15:02:53 +0200, Patrick [EMAIL PROTECTED] wrote:
Would you mind sharing the perl script and the database schema?
Perl script and database layout are below. Its not pretty since I
never intended it for external consumption but it does get the job
done.
If you unzip the files
I'm in the process of turning up a PRI in one of my markets and have
run into a problem I have never seen before. I am unable to place a
local outgoing call. Long Distance over the same PRI works fine.
When I attempt to place a local call using the PRI I see Asterisk
attempt to dial, and am
Could someone please post the url for the conf? also mute
your mic so everyone can hear!!!
~ken
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.768 / Virus Database: 515 - Release Date: 9/22/2004
Hi everybody again,
I try to use music on hold.
I have no idea how to put current call on hold, so I try
to use musiconhold by force.
I define these:
extensions.conf =
exten = ${KLP_TEST2},1,Answer
exten = ${KLP_TEST2},2,SetMusicOnHold(default)
exten =
- Original Message -
From: Alex Zeffertt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Friday, September 24, 2004 9:09 AM
Subject: Re: [Asterisk-Users] Asterisk 1.0 released
| Also - forgive me if this is a silly question - are
Hi there,
Can I compile asterisk in AS400 over linux fedora core 1??
I don`t know what wrong in this, the compilation stop in this line
###
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat
ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6
Some switches are fussy about you getting the NPI and TON (sometimes
jointly known as the dial plan) right. That is usually the cause of the
problem you see.
Regards,
Steve
Paul Oster wrote:
I'm in the process of turning up a PRI in one of my markets and have
run into a problem I have never
On Fri, 24 Sep 2004 10:06:06 -0500, Paul Oster [EMAIL PROTECTED] wrote:
I'm in the process of turning up a PRI in one of my markets and have
run into a problem I have never seen before. I am unable to place a
local outgoing call. Long Distance over the same PRI works fine.
When I attempt
On Sep 24, 2004, at 1:07 AM, Tom Ivar Helbekkmo wrote:
[EMAIL PROTECTED] writes:
exten = 109/_9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
[...]
as you can see, I'm wanting 109 to dial out broadvoice_1, [...]
You've got the source and target reversed. :-)
You want the Caller ID of your local
Could someone please post the url for the conf? also mute your mic so
everyone can hear!!!
IAX2/[EMAIL PROTECTED]/4569
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Thank you very much.
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320 - ext 2010
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Lykens
Sent: Friday, September 24, 2004 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi :)
I have do same test with Nokia 3650 (bluetooth) and Motorola A835 (bluetooth
and USB)
I have do a log of widcomm software and I can setup a coll (is not only a a
ATDxx)
Now the problem is the voice
With bluetooth is possible to use voice-gateway function I'm not a good
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat
ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -fsigned-ch
ar-DASTERISK_VERSION=\1.0-RC1\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/et
c/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
I am in the process of ordering a support contract from Cisco for my new
7960 phone, but I would really like to get it up and running. At the risk of
being flamed off this list, could someone send me or point me in the
direction of the SIP image files I need to change the phone over?
Thanks,
~c
Got that, someone said there was a brainstorming site or something?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Williams
Sent: Friday, September 24, 2004 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Fri, 2004-09-24 at 17:03, Scott Lykens wrote:
[snip]
Perl script and database layout are below.
[snap]
Thanks!
Regards,
Patrick
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Hello, everyone
I am having problems with a TDM400 that has 3 fxs modules and 1 fxo. When plug a line from the telco to the fxo module it changes state from onhook to offhook, and of course I can not receive any calls. (When I tried to call from the outside to that line it shows as busy). Could
Interesting. I think either the phonelabs adapter or cellsocket might
be an interesting idea. We are moving to a biz mobile package I use
iax2 term to fwd to a nextel since it's free inbound but having a cell
on the asterisk box is probably a better fit. Besides on a biz plan w/
tmobile and
Cirelle did you delete the .version file in the src tree on your box?
I doubt cvs is 2 wks behind since I got cvs commit emails this
morning. I believe make update will remove the .verision for you too
which will fix that issue.
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On Fri, 24 Sep 2004 10:27:03 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I'm curious if there is a way I can do some kind of balancing if an
outgoing
connection is already being used?
I was thinking about using the System command with a python script to
keep
an inventory of what
Hi.
Did you check out the plug? Is it wired
correctly?
I once had this problem because there was
two lines on that same plug (red/green and yellow/black) and plugging it into a
fxo would short them out and make then off-hook.
Michel Belleau
De:
[EMAIL PROTECTED]
Just tried it with 7,10, and 11 digit dialing, and got the expected
error from the switch, the number you have dialed is not a long
distance number, there is no need to dial the digit one before the
number...
Good suggestion, but that doesn't appear to be the problem.
On Fri, 24 Sep 2004
Asterisk and Cisco 79XX series configuration:
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx
Christopher Jacob wrote:
I am in the process of ordering a support contract from Cisco for my new
7960 phone, but I would really like to get it up and running. At the risk of
being flamed off this
What are you sending for the CSID? Dialing LD goes through the CLEC and may
be excepting your call no matter what the CSID is. The local switch may be
rejecting you because the CSID you are sending is not what they are
expecting. I had a the same experience on a legacy phone system.
Hello.
I trying to use SER with Asterisk together. I have a question
regarding the RTP path. If i make a call from one of my endpoints
registered in SER Server, and that call in particular is forwarded to
Asterisk and then to a PSTN-GW, Does the media goes through Asterisk?? is
there a
- Original Message -
From: William Suffill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Friday, September 24, 2004 11:56 AM
Subject: Re: [Asterisk-Users] Asterisk 1.0 released
| Cirelle did you delete the .version file in the src
As for how BT transmits Audio:
www.bluetooth.org
www.bluez.org
How Linux utilizes Bluetooth:
http://www.google.com/search?hl=enie=UTF-8q=linux+bluetooth
www.bluez.org
For how to write a channel, I suppose a seasoned linux programmer would
know by looking at the sources for existing channels.
*IF* I didn't already have the phone...
CellSocket $100
FXO Port $80
Non-BT Phone $0 (after rebates for new service)
$180
BT Dongle $10
BT-Phone $75 (after rebates for new service)
$85
But aside from all that, many
Hi Rgis,
Were going to build an IVR system with a TE405P and 4 E1. Were sure
that the 120 channels will be filled by 120 simultaneous calls during
peak, so we want to have the good server to manage this.
We wonder a lot of things and maybe you could help us.
- Are you ever build a similar
Has anyone else experienced a problem with app_queue
where after a time, calls can still come into asterisk, but once they enter a
queue, they just get silence, any calls in the queue get frozen in it, and
never get sent to an agent, yet calls can be made in or out of the phone
system.
DISCLAIMER: This code is free (I am not charging you to use it), but
you might have to pay royalty fees to the G.729 patent holders for using
their algorithm.
I finished this last Saturday and have had it on an Asterisk machine for
5 days without a crash, so I'm hoping that means it's safe to
Whicch version of zaptel and Zapata should I use with 1.0?
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On Friday 24 September 2004 12:46 pm, Christian Victor wrote:
Hi Rgis,
Were going to build an IVR system with a TE405P and 4 E1. Were sure
that the 120 channels will be filled by 120 simultaneous calls during
peak, so we want to have the good server to manage this.
We wonder a lot of
On Friday 24 September 2004 01:53 pm, Anton Tinchev wrote:
Whicch version of zaptel and Zapata should I use with 1.0?
One should always try to use the same version. CVS will give you all the files
you need.
--
Steve Szmidt
They that would give up essential liberty for temporary safety
It has happened at two different locations with two different cables/plug. Also when I plug to a normal phone it works ok.
"Michel Belleau (malaiwah.com)" [EMAIL PROTECTED] wrote:
Hi.
Did you check out the plug? Is it wired correctly?
I once had this problem because there was two lines
Has anyone had any experience with connecting
asterisk and Verso's new SIP stack in their Class 5 Call Manager?
I am hearing theres incompatabilites, but I can not
get anything directly from Verso themselves.
Their Call Manager is supposed to support XTens
softphones, so I would think that
Has anybody been able to get in touch with anybody at digium today?
Regards
Greg Cirino
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Heres a patch for the app_valetparking not working with music on hold.
This patch was made against the version at
http://www.bkw.org/app_valetparking.c.
As you can see, the original author of app_valetparking simply forgot to
copy the chan-musicclass to the new masq'ed channel. I'm not
I had a similar problem but not exactly same: when telco lines are
plugged into the FXO ports, initially zap show channel 1 says it is
Onhook but I cannot make outgoing calls. Once I unplug the telco
line and re-plug it, or after there is an incoming call, zap show
channel 1 says Offhook but both
On Sep 24, 2004, at 10:12 AM, Cirelle Enterprises wrote:
Has anybody been able to get in touch with anybody at digium today?
I suspect that they're all at Astricon.
Scott
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| On Sep 24, 2004, at 10:12 AM, Cirelle Enterprises wrote:
|
| Has anybody been able to get in touch with anybody at digium today?
|
| I suspect that they're all at Astricon.
|
|
| Scott
Ah... didn't realize astricon was still going on.
Greg
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I am new to Asterisk and I am investigating setting up a very large
Asterisk server farm. I have found a lot of good information on this topic
on the Wiki pages. I am drinking from the fire hose and I thought that I
read somewhere on Wiki a caution about a potential problem with running
Back in the office post-astricon. 1.0.0 running in the lab.
YIIIHAA!
THIS GUY rocks. Thanks to Mark for *, Steve and Olle for the conference
and to ALL community members. Everyone using * is contributing in one way
or another.
See y'all next year
Jason Kawakami
CSID
is caller sending ID. This is what
number you are sending from the PBX to the local carrier.
-Original
Message-
From:
Paul Oster [mailto:[EMAIL PROTECTED]]
Sent:
Friday, September 24, 2004 12:02 PM
To:
Henry Devito
Subject:
Re: [Asterisk-Users] Local Outbound Calls on
Yep, they took over one of the conference rooms, and basicly everyone from
digium is there. They had planned on having calls routed to them there but
there were lots of problems with the internet connection at the hotel, so it
didn't work very well. Today is the developer day(last day of
I've installed a TDM04B and a TDM40B. I haven't plugged any lines
into them yet but I'm starting to see this in my logs...
[EMAIL PROTECTED] asterisk]# grep alarm /var/log/messages
Sep 20 09:13:22 webster kernel: Power alarm on module 1, resetting!
Sep 22 11:07:07 webster kernel: Power alarm on
If I recall correctly, the problem with fxo port 1 is a hardware
design issue with early TDM cards. Call digium support to confirm.
I had a similar problem but not exactly same: when telco lines are
plugged into the FXO ports, initially zap show channel 1 says it is
I just signed up for Broadvoice, and used a similar network configuration that
I have on stanaphone, voipjet, and others.
My asterisk box is behind a vanilla Linux masquerade (netfilter/ipchains)
firewall. The SIP and IAX services have been working fine in both directions
for the other SIP
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