i have problem in compiling asterisk-addons 1.0.1
-
[EMAIL PROTECTED] asterisk-addons-1.0.1]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/local/mysql/include -c -o
cdr_addon_mysql.o cdr_addon_mysql.c
../asterisk: Not a directory
On Sat, 2005-01-29 at 00:11 -0800, Kamran Ahmad wrote:
i have problem in compiling asterisk-addons 1.0.1
-
[EMAIL PROTECTED] asterisk-addons-1.0.1]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/local/mysql/include -c -o
Have a look at http://www.sirrix.de/content/pages/pci4s0.htm
Currently they don't have an english website but in short this card
offers 4 Ports (8 Channels) and you are able to define the mode for each
channel (PtmP, PtP, TE, NT) on your own. They have asterisk-drivers
available (for Kernel
On fre, 2005-01-28 at 22:29 -0500, Jeff Lists wrote:
ISDN BRIs are delivered differently in different parts of the world. We
need more information to help you.
This is in the United States
The only Asterisk supported ISDN BRI hardware, that works in the states
are CAPI based active ISDN
Hi,
I'm trying to install asterisk with h323 support on rh9 box. I want to
find a working combination between asterisk,asterisk-oh323,pwlib and
openh323.
Thank you!
Ginel Tudorache
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Thanks for the link. will have to check it monday. Respect
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Hi!
Does anyone know of a speech recognition module (like say yes or no, or
numbers) I guess due to the complexity of speech recognition it might
just be found in commercial applications or am I wrong like always?
Search for sphinx.
Cheers, Philipp
Ginel Tudorache schrieb:
Hi,
I'm trying to install asterisk with h323 support on rh9 box. I want to
find a working combination between asterisk,asterisk-oh323,pwlib and
openh323.
Hi,
I'm using SuSE, not Redhat, but imho you'll succeed, if
you strictly follow the version hints mentioned in
Any suggestions? Anyone got something similar to work?
Mondial Software Limited
020 7043 2795
www.mondialsoftware.com
Click here to view our presentation of Cash Controller showing its
forecasting and automated bank reconciliation features
-Original Message-
From: [EMAIL PROTECTED]
Hi!
Is it possible to disable a reinvite on a specific call in the dial
plan? Any help would be greatly appreciatedJ
Put t or T into your dial statement, or set a codec that forces Asterisk
to do the transcoding using SetVar before Dial().
From the Wiki:
${SIP_CODEC}: Used to set the SIP
Hello,
This is like a repost as there was confusion
over type of incoming phone lines.
Over here incoming voice lines from telcos
are called trunk lines. This are basically
incoming voice lines that get connected to analog phones.
These are not T1 or PRI lines.
And lines that get distributed
Are there any VOIP lobbyist groups in Canada?
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brandon
Patterson
Sent: Monday, January 17, 2005 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Canadian
now it is giving another error
-
[EMAIL PROTECTED] asterisk-addons-1.0.1]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/local/mysql/include -c -o
app_addon_sql_mysql.o app_addon_sql_mysql.c
app_addon_sql_mysql.c:31:25:
Hi Michael.
Any work to support some USB Phones!? The ability to dial using the phones
keypad!?
Thanks.
Denis.
Em Sáb 29 Jan 2005 01:11, Michael Van Donselaar escreveu:
iaxComm is an Open Source softphone for the Asterisk PBX.
iaxComm compiles and runs on Win32, Linux and Mac OS X
On Jan 29, 2005, at 13:19, Kamran Ahmad wrote:
now it is giving another error
-
[EMAIL PROTECTED] asterisk-addons-1.0.1]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/local/mysql/include -c -o
app_addon_sql_mysql.o
Don't forget Howard, that Caller-ID presentation is an extra chargeable
service.
has it been turned on on these lines and confirmed ??
(its handy to carry a caller-id in your kit for checking:-)
On Sat, 29 Jan 2005 07:30:07 +1100, Howard Lowndes wrote:
On Fri, 2005-01-28 at 19:02, Simon Brown
Hi,
Currently I only have 1 PRI which I am using for dial-in
customers. The line is connected to a Portmaster3. I have never used more than
10 concurrent channels. The calls can be both analog or ISDN. It would be a
waste to order another PRI for my Asterisk box. Is there any way of
On Sat, 29 Jan 2005, David Norton wrote:
Currently I only have 1 PRI which I am using for dial-in customers. The line
is connected to a Portmaster3. I have never used more than 10 concurrent
channels. The calls can be both analog or ISDN. It would be a waste to order
another PRI for my
Hi,
I'm running Asterisk with HFC-S card connected to HW PBX in my office.
When I make a call from home using iaxComm connected to Office Asterisk,
the outgoing latency is about 0.25 sec, which is quite OK. But to incoming
latency begins on 0.5 sec and in a minute it's about 5 seconds (!) and
On Sat, 29 Jan 2005 15:07:48 +0900, Kuniyoshi Murata [EMAIL PROTECTED]
wrote:
Hi,
Is MacOSX version yet to come?
I don't have any hardware to compile, so I've been depending on people to send
me binaries.
I have made a request to the builder of the previous binary. As soon as I get
something
I'm not sure searching for Sphinx will do him much good. There's
really nothing concrete that I've seen.
On Sat, 29 Jan 2005 11:40:32 +0100, Philipp von Klitzing
Search for sphinx.
Cheers, Philipp
--
Is it something someone said, was it something someone said?
Hello List,
Ive set up asterisk and install astcc
application, everything was well installed, but im having problem using
astcc with SIP. I dont have any Trunk card or any other analogic VoIP
card connected to my asterisk box. Im using SIP and asterisk-oh323 to
connect to my VoIP
I has Asterisk up and running on my IP address.
I put a Linksys router in front of it and forward the following ports
22 TCP
5060 UDP
1-2 UDP
80 Both
None of my x ten phones work. They register but I get an message
Authentication Required
If you rearch the xten
[EMAIL PROTECTED] wrote:
Are there any VOIP lobbyist groups in Canada?
Well, he's not exactly *in* Canada, but there's Jeff Pulver:
http://www.crtc.gc.ca/ENG/transcripts/2004/tt0922.htm
He seems to be about all we've got, thus far.
-Original Message-
From: [EMAIL PROTECTED]
On Sat, 29 Jan 2005 10:46:21 -0200, Denis Galvão - iSolve [EMAIL PROTECTED]
wrote:
Hi Michael.
Any work to support some USB Phones!? The ability to dial using the phones
keypad!?
Not yet, but I'll probably add suport for the TigerJet phone eventually.
Thanks.
Denis.
Em Sáb 29 Jan 2005
[EMAIL PROTECTED] wrote:
I am looking at setting up the following configuration and any
help/input/comments before signing the PRI contracts will be greatly
appreciated.
PRI Tampa PRI
Sarasota PRI
-- Nortel BCM--Asterisk--Asterisk---
On January 28, 2005 09:16 pm, Ryan Cavanaugh wrote:
I would like to link the Nortel BCM to * using the a digital trunk card.
The BCM will continue to service the Tampa location, the * box would
simply be used to pass extensions over the PRI to another * server in
Sarasota and for a few SIP
Thank you so much for your help.
I was obviously missing the second ':' after the username
This will help me now in that I only need to do an update before reading
rather than downloading the entire document again.
Having said that I have another question for you. I've downloaded the
file to
Pedro wrote:
understood - I use the # sign as well, but some users are not used to
using the # sign so decreasing the timer helps those that may forget
to use the # key.
Properly setting up the dialplan on the SIPura eliminates most of these
issues. Here's my dialplan on my SPA-2k:
On Jan 29, 2005, at 16:15, Daniel Eboa wrote:
image.tiff
image002.jpg
Hi Daniel,
Would it be possible for you to turn off attaching two image files as
signature replacements to each of your email and maybe use a text
signature instead?
Thanks!
jens
Matthew Simpson wrote:
Hello, I have a dial plan that tries to place a call over several
different outbound gateways, like this:
exten = _1X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,2,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,3,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,4,Dial(SIP/[EMAIL
[EMAIL PROTECTED] wrote:
Hi community,
does anyone out there made some experience with Varion
(www.govarion.com) based E1/T1 cards ?
I have no hands on experience with them, but they are based on the
open-source Tormenta card:
www.zapatatelephony.org
With Asterisk, I think any hardware
I am looking at setting up redundant servers for my client and I know the first
step in creating an auto fall back would be to have the FQDN set up in such a
way that it points to both of the server IP addresses, but, how can I make it
so that the phones will (with minimal delay) authenticate
We have an old CAC and we're trying to get groundstart working on it, we
think it may be a dip switch setting. Does anyone have the config
settings and/or the manual to config this thing? Any help is greatly
appreciated :-)
Matt
___
[EMAIL PROTECTED] wrote:
does anyone know when current HEAD is scheduled to stabilise? Is
there a plan, or is it still some time in the future?
I believe I saw an announcement recently that it will start
stabilizing in February, with the goal of releasing 1.1 on the
six-month anniversary of
Stefan Gofferje wrote:
Hi folks,
I have a rather nasty problem. I have set up an asterisk test system
with a Cisco phone, a X-Lite client and so on and did some testing.
To the developers: great work! Hell of great!
However, it seems to me like asterisk puts the Caller ID Number into the
SIP
Currently I have a toll free line set up on my PBX at home, and it is working
great, BUT, when an inbound call comes in, the CID is reporting toll free call.
Does anyone know how I can get around this problem?
Thanks,
Dan
___
Asterisk-Users mailing
Jay Wilton wrote:
I just purchased a Cisco 7960 IP phone and I couldn't be
more happier.
It supports 6 lines. You can usually find a good one off
eBay.
I thought asterisk could only register once per ip? guess
i'll go read the wiki for a while longer. :)
No. Asterisk only supports one
Dave Morrow wrote:
Hi all, I have asterisk installed and working just fine with a couple
of Cisco IP Phones. I am now ready to pilot connectivity to PSTN and
am wondering what hardware would be recommended to make minimum
connectivity to the public telephone network.
Minimum would be any
You can get a support login at CAC for free, You can download all the
manuals in PDF from their support site. www.carrieraccess.com
-Matt
On Jan 29, 2005, at 11:11 AM, Matt Schulte wrote:
We have an old CAC and we're trying to get groundstart working on it,
we
think it may be a dip switch
On 16:11, Sat 29 Jan 05, [EMAIL PROTECTED] wrote:
I am looking at setting up redundant servers for my client and I know the
first step in creating an auto fall back would be to have the FQDN set up in
such a way that it points to both of the server IP addresses, but, how can I
make it so
On Sat, 29 Jan 2005 11:10:25 -0500, dean collins [EMAIL PROTECTED] wrote:
What should I be using to read this so it doesn't show the XML?
This would have been more appropriate to post to the asterisk-docs
mailing list, but either way.
The documentation is written in DocBook. Whatever is checked
Eric Wieling aka ManxPower wrote:
The issue happens when you want to use the same username/password for
more than one line. Can't do that (unless the phone has special support
for only registering once for all lines, i.e. the Polycom). Also you
can't have more than one device register with
I have asterisk connected to an adit 600 with fxo ports.
When I place a call with asterisk, asterisk will try to dial
out on the first line even if the first line is already being
used by someone else. Any ideas on what I'm doing
wrong?
Thanks,
Jon.
Hi All,
After several months using a TDM400p to access my 4 POTS lines I'm
thinking that there has to be a better way. I could order up a BRI from
SBC and ditch the POTS lines altogether.Does anyone have ISDN working
in the US? What are the issues involved with that interface?
Thanks,
Michael
On January 29, 2005 11:11 am, Matt Schulte wrote:
We have an old CAC and we're trying to get groundstart working on it, we
think it may be a dip switch setting. Does anyone have the config
settings and/or the manual to config this thing? Any help is greatly
appreciated :-)
Carrier Access is
Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
[snip]
Your diagram is a bit confusing to me. Still, the use of PRI in the BCM
is a good plan. You've avoided using the highly unstable VoIP functions
in the BCM.
Sorry, it got a little jumbled
PRI support on the BCM is based on the
Ok no probs, I thought the whole reason for CVS was so that it saved
downloading an entire copy of the book each time I wanted to update to
the latest version.
Thereby making for easier burden for you guys hosting the docs.
I'm happy to download a pdf each time (unlimited bandwidth available
Does anyone know where I can find the Asterisk Python interpreter?
The one referenced here:
http://www.sineapps.com/news.php?rssid=173
It seems like the site
http://vox.groovy.net/moin/PyAsterisk
is down
___
Asterisk-Users
On Sat, 2005-01-29 at 15:48 +0100, Zdik Kudrle wrote:
I'm running Asterisk with HFC-S card connected to HW PBX in my office.
When I make a call from home using iaxComm connected to Office Asterisk,
the outgoing latency is about 0.25 sec, which is quite OK. But to incoming
latency begins on
Hi, not sure if it is against the rules
to sell second hand equipment in here but havent seen anything against
it so here it is.
Im upgrading to 2 lines so I have
some spare equipment for sale here. This is an ideal starter pack and will get
you going with 1 line and 1 extension.
1
David Norton wrote:
Hi,
Currently I only have 1 PRI which I am using for dial-in customers.
The line is connected to a Portmaster3. I have never used more than
10 concurrent channels. The calls can be both analog or ISDN. It
would be a waste to order another PRI for my Asterisk box. Is
On Sat, 29 Jan 2005 15:07:48 +0900, Kuniyoshi Murata [EMAIL PROTECTED]
wrote:
Hi,
Is MacOSX version yet to come?
Thanks to Andreas Wrede for the binary!
http://iaxclient.sourceforge.net/iaxclient/iaxcomm-mac-1.0rc1.zip
___
Asterisk-Users mailing
On Sat, 2005-01-29 at 11:45 -0500, Jim Van Meggelen wrote:
David Norton wrote:
Hi,
Currently I only have 1 PRI which I am using for dial-in customers.
The line is connected to a Portmaster3. I have never used more than
10 concurrent channels. The calls can be both analog or ISDN. It
On Sat, 2005-01-29 at 10:36 -0600, Jon Gabrielson wrote:
I have asterisk connected to an adit 600 with fxo ports.
When I place a call with asterisk, asterisk will try to dial
out on the first line even if the first line is already being
used by someone else. Any ideas on what I'm doing
I would recommend using the local trunk and then you just need a context
that will dial out in your extensions.conf. Just put the context name
into the Peer/Trunk field on the trunks page. Currently there is not
support in astcc for oh-323. It would be trivial to add but
Darren Wiebe
Hi,
probably I won't be much of help, but I'm also looking for speech
recognition solution. But we're actually looking at two problems:
- one would be so called voice dialing (similar to celular phones) - one
records its own spoken names and speaks them after to call certain person -
this problem
i did evrything you mentioned, i thing is for my eyebeam version, mine is 3002s
what`s yours?
On Fri, 28 Jan 2005 23:10:40 -0500 (EST), Francois Meehan
[EMAIL PROTECTED] wrote:
Thanks Wessel,
You really have to know about that little switch on button, I had 2
eyebeam connected with their
I got this error when i try to dial:
-- Executing Answer(SIP/8000104-71a3, ) in new stack
-- Executing DeadAGI(SIP/8000104-71a3, astcc.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File
astcc-tone
Hello,
I'm looking at building a couple new PRI Gateway boxes using
TE405P cards, and was wondering if anyone has had any experiences (good or
bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics
builds some really nice (and cost effective) 1U servers based on the
Which phones do you have? We are using Cisco 7940G phones and I have
been able to do this by modifying the dialplan.xml for the phone to
rewrite numbers as they are dialed to include the 9 in front of
whatever is dialed from the phone. Now you can use the received calls
menus without having
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, January 28, 2005 10:22 PM
Subject: [Asterisk-Users] Integrating with existing 1BRI,6 POTS Panasonic
PBX ?
Hi,
at the
Yah tried that, they won't let me register. They want 50$ for the manual
:(
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 29, 2005 10:35 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] CAC Access Bank
On January 29,
We are a voip terminating company, we're using Channelbank with FXS
modules, Rhino, CAC, etc.. What we're wondering is, is how to would you
echo cancel a channelbank. Of course we're realizing that cancel'ing on
the T1 (on Ast) does no good (we think?) because the analog conversion
is at the
Bill Seddon a écrit :
Any suggestions? Anyone got something similar to work?
Yes it's working fine for me. Try by deactivate/reactivate IAX on your
account. I use the same config that Gonzalo send.
Mondial Software Limited
020 7043 2795
www.mondialsoftware.com
Click here to view our
The current astcc Makefile puts the sound files into the wrong directory.
It uses /usr/share/asterisk/sounds but it should be
/var/lib/asterisk/sounds.
Karl Putz
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa
Sent: Saturday, January 29,
According to the CAC manual for a Access Bank 1, there is no such option.
The FXS channels adapt according to the signaling bits sent to it over the
T1. ABC should be off for each channel.
Lyle
- Original Message -
From: Matt Schulte [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
[EMAIL PROTECTED] wrote:
does anyone know when current HEAD is scheduled to stabilise? Is
there a plan, or is it still some time in the future?
I believe I saw an announcement recently that it will start
stabilizing in February, with the goal of releasing 1.1 on the
six-month anniversary of the
Hi
I can't find out how to add more links to the Navigation box in
voip-info.org. Can someone help me out, please?
roy
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To
No, not really, what information do you need?
I have no idea whether it is an asterisk problem,
and adit 600 problem, etc... I was hoping someone
would be able to give me an idea of which config
file to look into, etc... so I wouldn't have to attach
20 config files to my email.
Jon.
On
[EMAIL PROTECTED] wrote:
Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
[snip]
Your diagram is a bit confusing to me. Still, the use of PRI in the
BCM is a good plan. You've avoided using the highly unstable VoIP
functions in the BCM.
Sorry, it got a little jumbled
No worries. I
On Sat, 29 Jan 2005, Matt Schulte wrote:
We are a voip terminating company, we're using Channelbank with FXS
modules, Rhino, CAC, etc.. What we're wondering is, is how to would you
echo cancel a channelbank. Of course we're realizing that cancel'ing on
the T1 (on Ast) does no good (we think?)
Folks,
Many thanks to Howard Lowndes who helped solve this problem; I ended
up using SetCallerID() instead of SetCIDName() as Howard suggested. Although
SetCIDName() changed the value correctly, desk set CID displays displayed
either unavailable or out of area on incoming calls from my
Agh, what I meant was the echo is heard from the PSTN side. It seems
echo canceling on the T1 (going to channelbank) does nothing, I'm
assuming because the T1 is digital and the channelbank is the
traversal from digital to analog.
-Original Message-
From: Peter Svensson [mailto:[EMAIL
On Sat, 29 Jan 2005, Matt Schulte wrote:
Agh, what I meant was the echo is heard from the PSTN side. It seems
echo canceling on the T1 (going to channelbank) does nothing, I'm
assuming because the T1 is digital and the channelbank is the
traversal from digital to analog.
Still, echo
A T1 is a two way transmission media. There is sound going both ways over a
'4 wire' interface. 4-wire means that Xmitt is seperate from Recv. In the
channel bank there is a 4 wire to 2 wire conversion. It's this junction
that introduces echo as some of the 4 wire recv gets feed back into the
On Sat, 2005-01-29 at 15:48 +0100, Zdik Kudrle wrote:
I'm running Asterisk with HFC-S card connected to HW PBX in my office.
When I make a call from home using iaxComm connected to Office
Asterisk, the outgoing latency is about 0.25 sec, which is quite OK.
But to incoming latency
Hello,
I'm looking for a
Cisco/Lucent/Asterisk VoIP SIP guru in the bay area. Full or part
time/consulting.
Please contact me if
you know someone or are interested yourself.
Thanks!
Todd
Houser
(916)
670.1873
___
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On Sat, 2005-01-29 at 00:38 -0700, Joseph wrote:
When I try to reload configuration *-1.0.5 can not find my sip.conf.
I don't see anything like:
== Parsing '/etc/asterisk/sip.conf': Found
WARNING[10301]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for
On Sat, 29 Jan 2005 11:56:56 -0500, dean collins [EMAIL PROTECTED] wrote:
Hi, not sure if it is against the rules to sell second hand equipment in
here but haven't seen anything against it so here it is.
From the Digium.com website under mailing lists:
The [EMAIL PROTECTED] list is for
On January 29, 2005 12:31 pm, Matt Schulte wrote:
We are a voip terminating company, we're using Channelbank with FXS
modules, Rhino, CAC, etc.. What we're wondering is, is how to would you
echo cancel a channelbank. Of course we're realizing that cancel'ing on
the T1 (on Ast) does no good (we
Hello Guys a pay in Mexico city a couple of days ago opened a E1 for testings.
wonder what was the concussions?
Did you get the stress test?
did the calls for the callers whent fine?
any comments about R2 in Mexico?
Does enyone have tested teh R2 support of asterisk with a R2 Card in
Option
On January 29, 2005 12:33 pm, Matt Schulte wrote:
Yah tried that, they won't let me register. They want 50$ for the manual
What? I could download it for free along with all their other manuals less
than a week ago. Are you perhaps asking for a hardcopy rather than just
downloading the .pdf?
I met Sangoma guys this Tuesday, and got a AFT102 for evaluation.
Right now I am in progress to develop a * channel driver for AFT10*
devices.
In that case you will have much more flexibility and to use all their
API.
Steven Critchfield wrote:
On Sat, 2005-01-29 at 11:45 -0500, Jim Van
When I try to call out to FWD over IAX2 I get:
Call rejected by 65.39.205.121: Unable to negotiate codec
I'm using asterisk-1.0.5 (the same settings works fine with *0.9)
I've standard settings in iax.conf
[general]
bindport=4569
register = x:[EMAIL PROTECTED]
[iaxfwd]
type=user
Has anyone written any drivers for ISA bus Dialogic Proline 2v or Dialogic
4 cards? I have a couple of them here gathering dust, I was wondering if I
could use them for Asterisk
Pat
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Hello,
I am new here. I am also somewhat new to telephony and IT, however, I am
technically adept.
I work for a small but growing non-profit in Colorado and I wear many hats.
Because I am the only one who has a clue about computers, I am the default
IT person.
Our communications needs are
Hi,
I was woredering if you could help me to put into practice this solution.
The idea: Create a IVR-Voicemail
The scene:
PSTN--/6--PBX/12- Internos
|
/4 ports
|
Hi,
After I boot of the CD ( version 0.4 ) I get following error message:
CD not found
The centos release 3.3 DC was not found in any of your cdroms drives.
Please insert the CentOS release 3.3 cd and press OK to retry.
Well I just boot of that cd so what is going on?
I have done some searches
On Sat, 29 Jan 2005 12:53:11 -0600
-Original Message-
From: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: Q: Can I over-ride the value of
${CALLERIDNAME} ?
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Folks,
I think you mean
http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-1.0rc1.zip
:-)
Andrew
On 30/01/2005, at 4:02 AM, Michael Van Donselaar wrote:
On Sat, 29 Jan 2005 15:07:48 +0900, Kuniyoshi Murata
[EMAIL PROTECTED]
wrote:
Hi,
Is MacOSX version yet to come?
Thanks to Andreas Wrede for the
Hello all, further to Rob's message, could I ask does zaptel.conf or
Zapata.conf need anything further (for caller ID) to allow the setting of
caller ID, as my below E1 pri debug shows that Asterisk seems to be
correctly sending the necessary Q931 instructions to the carrier (Colt) to
set the
there is a timing error:
The creation time for the file is:
Jan 29 15:30 18-20050129-152954-in.wav
The messages log is showing an error:
Jan 29 15:29:23 WARNING[3015]: Read error on sound device: Resource
temporarily unavailable
It would seems the file is still in use by soxmix when rm -f is being
Eric Wieling aka ManxPower wrote:
That's really up to your carrier. Many carriers to not allow customers
to set the Caller*ID number. I don't know of ANY that let you set the
Caller*ID name.
Well, pretty much all of them will let you send it over a PRI... it's
just another facility message
Conflict with with codec.
bandwidth=low should be disabled as ulaw is not a low bandwidth codec.
#Joseph
On Sat, 2005-01-29 at 13:13 -0700, Joseph wrote:
When I try to call out to FWD over IAX2 I get:
Call rejected by 65.39.205.121: Unable to negotiate codec
I'm using asterisk-1.0.5 (the
[EMAIL PROTECTED] wrote:
Folks,
Many thanks to Howard Lowndes who helped solve this problem; I ended
up using SetCallerID() instead of SetCIDName() as Howard suggested. Although
SetCIDName() changed the value correctly, desk set CID displays displayed
either unavailable or out of area on
Does anyone have a quad span 410P on a production Dell 1850? Digium
states that there are known incompatibilities with the 7x0 series,
probably related to interrupts.
Also I would like to find out if anyone attempted connecting more than
one of these cards to a dual Xeon system. I am strictly
Hi!
The transfer is always successful, but sometimes (lets say bout 50% of
the times), the MusicOnHold does not de-attach itself from the call,
Maybe a good start for an investigation is to try a different softphone
for the operator and see if it makes a difference. Then put Asterisk into
I am curious about Eyebeam so I went to xten's site and read up on it.
I still do not get a clear understanding as to what Eyebeam does. Help
me to understand it:
Am I correct?
It installs on a windows computer?
It connects/registers to my * box?
With a camera attached to my Windows computer I
Asterisk experts,
I have been pulling my hair out troubleshooting what appears to be
a Zap channel disconnect on the TDM side. I am trying to have an
inbound call on a T1 zap channel on the TE405P from our Harris
switch automatically dial a SIP channel. Any help appreciated
very appreciated.
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