I used the packages that were mentioned and have a link on the main website and
www.asterisk.org/download
I thought these were ok for production/use...
I just compiled 1.4.1 (./configure and make, no make install) and copied the
chan_zap.so module into /usr/lib/asterisk/modules, restarted
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still load the driver as modprobe qozap ports=12 as I've always
done. But now it only sees 2 ports.
Цитат на писмо от Doug Shubert [EMAIL PROTECTED]:
Hello,
I'm trying to use MySQL for Dialplans and have followed
the
Asterisk RealTime Extensions setup.
The MySQL table is called extensions and I have entered
two records..
ext 1000 and 2000.
I also added
switch = Realtime/[EMAIL
On Fri, Apr 06, 2007 Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still load the driver as modprobe qozap ports=12 as I've
I haven't updated for a while and when I looked on the web site how to
do a SVN update, I cannot find it anymore.
CLI show version
Asterisk SVN-branch-1.2-r42600M built by root @ asterisk on a x86_64
running Linux on 2006-09-10 22:52:42 UTC
1. Where is the description for the SVN update now?
Michael Boers wrote:
I have recently moved an asterisk system to a new location. This location
is experiencing terrible echo. I installed the HPEC from Digium but that
has caused a new problem.
When HPEC is enabled and more that 16 taps are used, the audio from the
outside caller gets
No need, fixed.
It was the dialplan. I had commented out some stuff, for other
troubleshooting and forgot about it.
Once I posted and got some replies and reviewed the basics, there it
was.
Thanks, all, for the push.
joe a.
Gustavo Cordeiro [EMAIL PROTECTED] Wrote: 4/5/2007 5:47
PM:
There seem to have been many discussions about this, so sorry if this is boring.
Can one connect a standard fax machine (or fax modem) to an analog port on a
TDM400p (as if it were an analog phone, say) and expect it to work reliably?
For sending, that is. Detecting and routing the call is
Is your carrier delivering service via a TDM circuit?
It has been our experience that you will get far more reliable fax
performance via the method you describe (analog device terminated to
a port on a FXS line card) than attempting to use an ATA on the LAN.
However, if your carrier is a
Where are you located?
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906
-Original Message-
From:
Same issue here as well.
Running Centos 4.4 (x86_64) with all updates installed on a Pentium D 2.8
GHz, 1.5GB RAM and TDM800P with 1 x Quad FXS and 2 x FXO ports.
I installed the new 64 bit HPEC module earlier in the week when it was
released and have been unable to get it to work. I have
I am having an odd problem with a linksys pap2 ata and asterisk...
Asterisk won't detect digits from it until I issue a 'sip debug'. As
soon as I turn on sip debugging, everything works perfectly (classic
heisenbug)!
Asterisk is latest Debian 'etch' packaged 1.2.13. sip.conf looks like:
Salvatore Giudice wrote:
Where are you located?
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
Is it me, or do people not know the value of whois information or
actually looking at the
AFAIK, the FAX targets are normal FAX machines, on the PSTN.
What happens is, there appears to be a dial out, and a FAX negotiation, but it
always fails.
joe a.
Bryan M. Johns [EMAIL PROTECTED] Wrote: 4/6/2007 8:49 AM:
Is your carrier delivering service via a TDM circuit?
It has been our
J. Oquendo wrote:
Is it me, or do people not know the value of whois information or
actually looking at the domain name then the contact information. For
someone in the Security training field especially, you should have
been able to find this out easily. Did a majority of people forget
On Fri, 6 Apr 2007, Joe Acquisto wrote:
AFAIK, the FAX targets are normal FAX machines, on the PSTN.
What happens is, there appears to be a dial out, and a FAX negotiation,
but it always fails.
I think what Bryan is asking: Where is your FAX source? If the sending
system is coming in on
J. Oquendo wrote:
Salvatore Giudice wrote:
Where are you located?
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
Is it me, or do people not know the value of whois information or
Hi all,
I am wondering: Is it possible to setup on demand conference recording
with meetme?
I have found out that one-touch recording ala automonitor does not work
here.
Any suggestions?
Thanks,
Ondrej
The information contained in this e-mail and in any attachments is confidential
and is
While pondering several issues, poor quality PSTN POTS lines, potential cost
savings with multiple cell numbers, the FAX problems over TDM400p, etc, I
wondered about:
Cell phone Base stations to replace POTS lines. Devices to cradle cell
phones and connect to TDM400p, for instance, to mimic
hi all,
for a user who has connected a asterisk syetem of his own to my asterisk
system, i use the following dial command
Dial(SIP/[EMAIL PROTECTED],30,Tt)
for others who are using sipura i use the following dial command
Dial(SIP/user,,Tt)
what im trying to do is, pass extension to the user
Hi all,
can anyone tell me what are the fields in the SIP header and URI for dialing
a peer?
--
Regards
Rizwan Hisham
Software Engineer
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Philipp Kempgen wrote:
How can people know that your asterisk system is where whois
says you are?
Regards,
Philipp
A start would be to get the contact information and actually CONTACT
the person about it. Come on now.
--
J. Oquendo
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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To
J. Oquendo wrote:
Philipp Kempgen wrote:
How can people know that your asterisk system is where whois
says you are?
Regards,
Philipp
A start would be to get the contact information and actually CONTACT
the person about it. Come on now.
Most people probably don't even know whois.
Hello,
From dialplan perspective, it seems you can't tailor your voicemail
behaviour to specific needs (dial 1 for old message listening, ...).
Can anyone recommend a way to do it ?
Does it make sense to write your own IVR and store audio files somewhere ?
Best regards
Rizwan Hisham wrote:
Hi all,
can anyone tell me what are the fields in the SIP header and URI for
dialing a peer?
--
Regards
Rizwan Hisham
Software Engineer
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On 6 Apr 2007, at 00:59, Jason Wolfe wrote:
I need to decide on the best way to add a voip SIP or IAX client to
a website. I'm thinking that I'd like it to be inline, like an
aplet, on the page. I've got some asterisk servers running to
connect up to, so the real challenge is finding an
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/6/2007 9:51 AM:
On Fri, 6 Apr 2007, Joe Acquisto wrote:
AFAIK, the FAX targets are normal FAX machines, on the PSTN.
What happens is, there appears to be a dial out, and a FAX negotiation,
but it always fails.
I think what Bryan is asking:
A start would be to get the contact information and actually CONTACT
the person about it. Come on now.
Maybe I'm confused... Isn't that what Dovid did when he replied to Tim's post?
Regards,
David
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On Fri, 2007-04-06 at 09:30 -0600, David Thomas wrote:
A start would be to get the contact information and actually CONTACT
the person about it. Come on now.
Maybe I'm confused... Isn't that what Dovid did when he replied to Tim's post?
Regards,
David
Both of the zaptel (1.2 and 1.4) drivers I tried were the latest. Are
the older drivers working better? Does anyone have a configuration
close to mine that is working?
--
Michael Boers
On 4/6/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Michael Boers wrote:
I have recently moved an
Hi:
Greg Siemon wrote:
Same issue here as well.
Running Centos 4.4 (x86_64) with all updates installed on a Pentium D
2.8 GHz, 1.5GB RAM and TDM800P with 1 x Quad FXS and 2 x FXO ports.
I installed the new 64 bit HPEC module earlier in the week when it was
released and have been unable
J. Oquendo wrote:
Rizwan Hisham wrote:
Hi all,
can anyone tell me what are the fields in the SIP header and URI for
dialing a peer?
--
Regards
Rizwan Hisham
Software Engineer
Sean Bright wrote:
Since when is Canada part of the rest of the world? I thought it was
a US National Park? ;-)
Only in July -- the rest of the time it's where the heating fuel comes
from :P
-Stephen-
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hi lee.
I see your problem with trunk iax, probably i don't have the solution but i
don't knew if you can help me to solve mine.
i want to connecte two asterisk server: server A and server B. i want make
possible calls betwen all asterisk users.: users in server A with sip number
022100 can phone
Joe Acquisto wrote:
AFAIK, the FAX targets are normal FAX machines, on the PSTN.
What happens is, there appears to be a dial out, and a FAX negotiation, but it
always fails.
What does zttest say about your zap card configuration/installation? If
it's not always 99.98% or better then it's
Here are the relevant parts from extensions.conf file. It works only for local
extensions whose number id 806xx. Note one thing: When you use the H extension
the generated CDR is wrong - the destination extension is H and not the
original number. I've done some small code change in Asterisk and
Hi all,
Has anyone had any experience getting Polycom phones repaired? The screen on
one of our IP600s got smashed, and I'm wondering if it's worth the effort to
get it repaired, or if it'd just be cheaper to buy a new phone.
Thanks,
--
James Andrewartha
Systems Administrator
Data Analysis
In my case, the rx and tx gains are 0. I will try the 1.4 version for
fxotune to see if that helps. Thanks for the suggestions!
--
Michael Boers
On 4/6/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Hi:
Greg Siemon wrote:
Same issue here as well.
Running Centos 4.4 (x86_64) with all
Dear List,
I'm having a blinking MWI light on the snom 320 even when there's no message
waiting in Asterisk.
We've managed to make the voicemail button work using
fromdomain=192.168.0.1 in sip.conf
vmexten=2500 (our VoicemailMain application extension in extensions.conf).
We also added
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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To
Lee Howard [EMAIL PROTECTED] Wrote: 4/6/2007 2:15 PM:
Joe Acquisto wrote:
AFAIK, the FAX targets are normal FAX machines, on the PSTN.
What happens is, there appears to be a dial out, and a FAX negotiation, but
What it always fails.
What does zttest say about your zap card
You can find zttest.c in the zaptel source package. Download it from the
asterisk.org.
Sds,
Gustavo
From: Joe Acquisto [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Joe Acquisto wrote:
Lee Howard [EMAIL PROTECTED] Wrote: 4/6/2007 2:15 PM:
Joe Acquisto wrote:
AFAIK, the FAX targets are normal FAX machines, on the PSTN.
What happens is, there appears to be a dial out, and a FAX negotiation, but
What it always fails.
What does zttest say
In an article on voip-info.org (
http://www.voip-info.org/wiki/view/Asterisk+setup+research+lab ) , it is
suggested that one thing you can try to reduce echo is to insert a 500 ohm
resister on the ring line of the POTS line. Has anyone successfully tried
this? What was your experience. Also,
On Fri, 2007-04-06 at 12:00 -0700,
[EMAIL PROTECTED] wrote:
Date: Fri, 6 Apr 2007 16:13:29 +0100
From: Tim Panton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Open Source VoIP client (on a webpage)
To: Jason Wolfe [EMAIL PROTECTED], Asterisk Users Mailing
List - Non-Commercial
On Tue, 2007-04-03 at 05:30 -0700, Jason Kim wrote:
Is it exists?
If not, how could they have done this:
http://opensourcepbx.tmcnet.com/topics/applications/articles/5450-industry-forum-hails-successful-voip-over-asterisk-ipv6.htm
(But i'll guess it's not mainstream code yet...)
hw
--
Michael Boers wrote:
In an article on voip-info.org http://voip-info.org (
http://www.voip-info.org/wiki/view/Asterisk+setup+research+lab ) , it
is suggested that one thing you can try to reduce echo is to insert a
500 ohm resister on the ring line of the POTS line. Has anyone
successfully
Hi Matthew,
Tim is based in the UK so is probably offline by now so I thought I'd answer
your question for you (I'm in New York).
Mexuar is licensed Per Server (it's tied to the external IP address of your
server when you order the license), apart from this however there are no
I find the source. IIRC, the problem was there was some dependency
(terminology ?), or package that was missing and I got complaints when trying
to compile. I can't recall what it was, but it is something that is included
in most distro's but not SUSE by default.
Is there a way to compile
Ariel Monaco wrote:
Dear List,
I'm having a blinking MWI light on the snom 320 even when there's no
message waiting in Asterisk.
We've managed to make the voicemail button work using
fromdomain=192.168.0.1 in sip.conf
vmexten=2500 (our VoicemailMain application extension in
I am having an odd problem with a linksys pap2 ata and asterisk...
Asterisk won't detect digits from it until I issue a 'sip debug'. As
soon as I turn on sip debugging, everything works perfectly (classic
heisenbug)!
Instead of SIP debug, try capturing the traffic with tcpdump etc. on
the
May be slightly off topic, but I was wondering what everyone thinks of this
latest ruling against Vonage? Does anyone really know what Verizon hold
patents for, and could those patents possible affect anything in Asterisk?
Who knows who Verizon will go after next.
Brent
http://www.theregister.co.uk/2007/04/06/vonage_new_customers_ban/
this is bad bad bad, wondering how a court can go after the customer but
not also the USA vendor of the hardware.
Cheers,
Dean
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
It begins to look like government-backed racketeering and gangsterism.
What is happening to free enterprise these days?
Dean Collins wrote:
http://www.theregister.co.uk/2007/04/06/vonage_new_customers_ban/
this is bad bad bad, wondering how a court can go after the customer
but not also
On Fri, 06 Apr 2007, Paul wrote:
It begins to look like government-backed racketeering and gangsterism.
What is happening to free enterprise these days?
Dean Collins wrote:
Stepping back into reality for a moment, emotions aside, I side with
the law if Vonage infringed on VZ's patent.
J. Oquendo wrote:
On Fri, 06 Apr 2007, Paul wrote:
It begins to look like government-backed racketeering and gangsterism.
What is happening to free enterprise these days?
Dean Collins wrote:
Stepping back into reality for a moment, emotions aside, I side with
the law if Vonage
I've seen in the wiki that it is possible to use a celldock device to
use a cell phone as a PSTN line to Asterisk, but I haven't seen any
comments as to how well this actually works. I was thinking about
hooking a celldock to a FXO input of my Digium TDM400P card and use it
to connect via
On Fri, 06 Apr 2007, Paul wrote:
I just think an order against taking on new customers is going too far.
It is like Verizon is saying they will never grant Vonage a license at a
reasonable enough price to remain in business. Besides, I saw some news
that Vonage is exploring arrangements with
Hello
Approximately every 15 minutes I'm getting the following error (on all
channels):
[Apr 7 01:34:27] WARNING[6530]: chan_zap.c:6628 handle_init_event:
Detected alarm on channel 31: Yellow Alarm
[Apr 7 01:34:27] WARNING[6529]: chan_zap.c:2388 pri_find_dchan: No
D-channels available! Using
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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Steve Prior wrote:
I've seen in the wiki that it is possible to use a celldock device to
use a cell phone as a PSTN line to Asterisk, but I haven't seen any
comments as to how well this actually works. I was thinking about
hooking a celldock to a FXO input of my Digium TDM400P card and use it
Thanks for the helps Stephen. I was running non standard gains but setting
regain and txgain to zero (then reloading chan_zap.so) does not help. I
still get the broken audio, in fact sometimes I don't get any audio at all.
In testing the server just froze a number of times and had to be rebooted
Warning - novice question ahead!
Dear List,
I have installed Asterisk 1.4.2 on an AMD dual core x86-64 box running
CentOS 4.4. Compilation and installation were straightforward.
The box only supports IAX connections so I have no zap hardware.
My question is this - where do I set the txgain
When I listen to voicemail from my Google Talk buddy, the envelope says, from
an unknown caller. But the voicemail correctly records the caller ID of calls
that arrive via Zapata into the same context that receives Google Talk calls.
How can I configure the voicemail to include the caller's
On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote:
zttest does not exist on this system, Suse 10 based. IIRC, I never
found the file(s) needed to compile it.
Do you actually have a timing source?
head -c 0 /dev/zap/pseudo
Do you get input from there in a resonable time?
On Fri, Apr 06, 2007 at 12:37:47PM -0700, Lee Howard wrote:
It's usually built and left in the zaptel source directory where you
extracted and built zaptel. If it doesn't get built for you from
zttest.c then check the Makefile that it has zttest in BINS like this
from mine:
BINS=ztcfg
On Fri, Apr 06, 2007 at 04:11:45PM -0400, Michael Boers wrote:
In an article on voip-info.org (
http://www.voip-info.org/wiki/view/Asterisk+setup+research+lab ) , it is
suggested that one thing you can try to reduce echo is to insert a 500 ohm
resister on the ring line of the POTS line. Has
On 4/5/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
tried x+102 ?
NEVER do that. The call can fail for other reasons besides the
carrier. It can and will create conditions where your carrier properly
connects the call and then the call is re-dialed via another provider
or line.
On 4/6/07, Steve Prior [EMAIL PROTECTED] wrote:
I just found out that the celldock I'm talking about is also called the
Dock-N-Talk.
Works just fine. There is a delay, actually a LONG delay from the
time you dial the number and the cellphone connects the call. Or, at
least with my Motorola
We should have a welcome back to work party for fb.
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On 4/6/07, Joe Acquisto [EMAIL PROTECTED] wrote:
While pondering several issues, poor quality PSTN POTS lines, potential cost
savings with multiple cell numbers, the FAX problems over TDM400p, etc, I
wondered about:
Poor quality POTS lines and fax problems do seem to be related. The
added
On 4/6/07, Joe Acquisto [EMAIL PROTECTED] wrote:
I find the source. IIRC, the problem was there was some dependency
(terminology ?), or package that was missing and I got complaints when trying
to compile. I can't recall what it was, but it is something that is included
in most distro's
Thanks for the suggestion. I am using a tdm400p with 4 fxo channels. I am
in the US, so opermode should not be ok at default settings. I just
recently got fxotune working on my system. The version that comes with
zaptel 1.2.16 would simply hang. I am using the 1.4 fxotune now with the
1.2.16
Oops, meant to say that opermode should BE ok at default settings.
On 4/6/07, Michael Boers [EMAIL PROTECTED] wrote:
Thanks for the suggestion. I am using a tdm400p with 4 fxo channels. I
am in the US, so opermode should not be ok at default settings. I just
recently got fxotune working on
I am having an odd problem with a linksys pap2 ata and asterisk...
Asterisk won't detect digits from it until I issue a 'sip debug'. As
soon as I turn on sip debugging, everything works perfectly (classic
heisenbug)!
Instead of SIP debug, try capturing the traffic with tcpdump etc. on
From: Bob Smither [EMAIL PROTECTED]
Date: Fri, 06 Apr 2007 20:22:34 -0500
Warning - novice question ahead!
Dear List,
I have installed Asterisk 1.4.2 on an AMD dual core x86-64 box running
CentOS 4.4. Compilation and installation were straightforward.
The box only supports IAX connections so
From: hind habaoui [EMAIL PROTECTED]
Date: Fri, 6 Apr 2007 18:01:11 +
hi lee.
I see your problem with trunk iax, probably i don't have the solution but i
don't knew if you can help me to solve mine.
Can't seem to see what the problem you have? Errors? Incorrect result?
(What is expected
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