On Wednesday 19 January 2005 23:15, Eric Bishop wrote:
Well guys this is truly bizarre. I managed to get a DL360 G3 to show
interrupts with FC2 but not FC3. Exact same config and setup
proceedure. Ofcourse neither FC2 or FC3 show interrupts with the DL360
G4. I think TE410P is just a
Hello everbody,
I am having problems is Database version and Real time version of Asterisk.
Users are connecting with no problem,
they gets authenticate and its working fine,
but
after 2-3 minutes, registration with the same user comes and it gets
failed to authenticate. dial tone gone, users
Hi
All,
I've managed to
compile make and make install asterisk on Mandrake 9.2.
However on startup I
get the following message:
[cdr_tds.so]Jan 20
11:13:54 WARNING[20999]: loader.c:258 ast_load_resource: libtds.so.3: cannot
open shared object file: No such file or directoryJan 20
Sorry all,
Did that and its going good now.
Rgds
Nic
From: Nic le Roux [mailto:[EMAIL PROTECTED]
Sent: 20 January 2005 11:22 AMTo:
'asterisk-users@lists.digium.com'Subject: Asterisk 1.0.3
startup
Hi
All,
I've managed to
compile make and make install asterisk
Hi All,
Does anyone know of a way to dial two different outbound numbers and bridge them together using the Asterisk API?
Cheers,
Taff.
ALL-NEW
Yahoo! Messenger
- all new features - even more fun!
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I would also suggest that while it is possible to do something, it is
not always wise :) See the significant volumes of reports in the
archives regarding multiple zaptel cards in one system.
I must be lucky: I have 2 X100P and a TDM400 with zero IRQ or other
issues. And double NAT for the voIP
On Thu, 20 Jan 2005, Wilson Pickett wrote:
I would also suggest that while it is possible to do something, it is
not always wise :) See the significant volumes of reports in the
archives regarding multiple zaptel cards in one system.
I must be lucky: I have 2 X100P and a TDM400 with
Hi,
Were using Asterisk with Digium TE110P card for the
PSTN E1 interface. Our PRI is enabled with detecting the
connected-party-number feature. When an OUTBOUND call is made to
a phone, the PRI will send back an ISDN messages containing the
connected-number and we can use that
On Thu, 20 Jan 2005, taf taffey wrote:
Does anyone know of a way to dial two different outbound numbers and
bridge them together using the Asterisk API?
Which api do you mean? There are at least two ways:
- Using a call file in the spool directory
- Using the originate command in the
Hello,
I want to build a PBX with the following
specs :
1. we have two trunk lines
2. we need upto 8 extensions - all analog phones maybe
one voip phone
What is hardware that I need and where to find it ?
Thanks
Varun
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Hi,
sip.conf has a paramter regexten using which we can assign an extension
to a registered SIP client and can use the same number to call that client.
Is there any such parameter for realtime sip table sip_buddies. Why was this
missed out in this table ?
Thanks,
~Vamsi
Hello,
Asterisk provides its own Asterisk gatekeeper is there
other wise it supprots gnugk
please tell me
Thank u
Sailatha[EMAIL PROTECTED]
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I've been struggling with connection Asterisk to ISDN BRI lines for a
while.
I have it working with the latest bristuff and compatible Asterisk
version:
Asterisk 1.0.3-BRIstuffed-0.2.0-RC3a
I am using a cheap Centronics ISDN card and the zaphfc drivers.
It works but users complain that the
Can anybody help me find this patch?
So nobody knows of a pre-built web-interface that can accomplish these
goals? Ohh well, time to work with a developer to custom build one. Anybody
else interested in these features? Should I post the source/code once I have
it?
-Original Message-
I have this problem for 2 days and i dont understand
I am behind a nat
my sip.conf is:
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip
disallow = all
allow= gsm
allow= ilbc
allow= ulaw
Dear Steve and *.* e1r2 developers and users,
now MFCR2 is successfully installed! many thanks for
your help.
I'm living in Argelia. I have configure my MFCR2
according argentina R2 settigs. (look at the end of
the message)
the testcall run perfectly (only warnings and I think
that is just
HiA followup on my previous problem (no sound) description: I compiled zaptel and ztdummy, and loaded themThen i recompiled asterisk and configured sip clients and a conference. When i load the ztdummy module into the kernel, and run asterisk, the conference room seems to work, but i cannot hear
Hello.
I want to change the domain in the from url when making a call. I can
change de user ID with SetCallerId but asterisk adds @192.168.1.2
How can I define what to add to the CallerID in extensions.conf?
Thank you.
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I'd say you'd need at least a Quad Pentium 4 Xeon 3.2ghz server. 4gb of RAM
and a Raid 5 array with 5 120gb 10,000 rpm SCSI drives. Check ibm.com for
it.
:-)
Just Kidding. Seriously though, 8 extensions isn't much. When you say trunk
lines though, do you mean 2 Voice T1/PRI's? Each with 23 phone
Are you using PC or mac?
Isamar
On Thu, 20 Jan 2005, Wilson Pickett wrote:
I would also suggest that while it is possible to do something, it is
not always wise :) See the significant volumes of reports in the
archives regarding multiple zaptel cards in one system.
I must be lucky: I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ben Merrills wrote:
| There's a few (open source/free) ones in development. I myself am
| developing one of them.
|
| Ben
|
Hi.
Why not join all the project in just one ?
Actually which queue log analyzers projects are beeing developed ?
Check the mail
I've not released the source yet, I asked last week on the mailing list for
people to send me over some example queue_logs, because so far I've only been
able to test the software against my own.
I have however made a lot of changes to it since last I posted about it.
Template engine has been
Good day all
We have 2 asterisk servers connected to each other via IAX2 using ilbc.
Each call we make goes up to 25kbit and each one there after 25kbit as
well
Is there a way to bring it down?
Pleas Help
Altus
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Hi
Is it possible to somehow monitor/log packet loss and/or jitter in RTP?
I want to know how things look if someone complains about audio.
Best regards
roy
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It's called asternic, www.asternic.org .. The client is based on flash which
connects to a perl daemon on the server. It uses the manager (manager.conf)
interface to determine extension status. Pretty neat :-)
Matt
-Original Message-
From: David John Walsh [mailto:[EMAIL
Hi!
I'd like to be able to park/pickup a call with h323.
Parking the call is easy, using ParkAndAnnounce.
But ParkAndAnnounce does not return the parkinglot in a variable.
So, I can't retrieve the call later.
To keep it simple, here is a simple scenario of what I want:
The call is established
Am Mittwoch, 19. Januar 2005 19:18 schrieb Asterisk List:
The current CVS HEAD version already has ## transfer built-in. See
the included configs/features.conf.sample file. You can define your
own transfer key sequence. There is also an attended transfer
feature.
What is an attended
I would like to create a rock solid asterisk server that comes up no
matter what.
Ofcourse I can consider hardware raid1 but for the cost of a hardware
raid controller I can also buy a 2GB compact flash card that doesn't
produce any heat or noise and is friendly to the electricity bill and
Hi again folks! ;)
As before, I will transform one E1 30 Channel PRI into 30 FXS channels using
Adit 600.
Now I'm into choosing server platform. And the two opponents are:
* Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1)
* FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1)
As I've seen
So if you think the server can handle 5 TDM400P cards let me know.
I've done an installation with 5 TDM400P cards - 4 PSTN lines and 12
analog phones.
There are no outstanding issues that havent been solved by tweaking a
particular config option (e.g echo, callprogress issues etc...).
Good day,
I just downloaded the latest CVS and it will not compile. This is the error
I receive:
pbx_dundi.c:54:18: zlib.h: No such file or directory
pbx_dundi.c: In function `update_key':
pbx_dundi.c:1313: warning: implicit declaration of function `crc32'
pbx_dundi.c: In function
Let me restate my problem. I have a group of users behind a constrained
pipe to the public network. There are a few mobile users that will
mostly be working from their home offices. I *really* want to avoid
having a call from a mobile user to a public number cause double the
traffic on
On January 20, 2005 11:42 am, Begumisa Gerald M wrote:
I've done an installation with 5 TDM400P cards - 4 PSTN lines and 12
analog phones.
I'm curious -- what is the motherboard you're doing this on? CPU?
That's a lot of interrupt load!
-A.
___
Hello,
I am using chan_capi 0.3.5 and Asterisk CVS-v1-0-12/29/04-15:32:48 on a SuSE
Linux 9.0 with Kernel 2.4.21-99-default
In the system is a AVM C4 with one port connected to PSTN at PTP BRI and
another one to an ISDN PBX with an PMP BRI.
The system is running fine, but I have regualary this
Hello list ,
I´d like to report a success case with a modem based
on chipset : Motorola 62802-51.
It works fine , and zaptel identifies as a X100P
( not clone ) .
Red Alarms can be identified . :) This doesn´t
occurred on MD3200 ambient chipsets.
can you send us more info?
driver,versions,logs,
Title: Message
Hi,
Is
there a free toll for SIP stress testing that supports RTP?
Can
SIPp be used for such purposes (to send audio)?
Regards,
Stojan
Sljivic
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Kaws:
Are you using unicall-0.0.1d or earlier?
If yes, please switch to 0.0.2pre4 (or pre3 if you have sound problems with
pre4) and test again. 0.0.1 versions had a lot of problems, mostly in
outgoing calls
Guillermo
From: kaws elchamal [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List -
Good day all
We have 2 asterisk servers connected to each other via IAX2 using ilbc.
Each call we make goes up to 25kbit and each one there after 25kbit as
well
bit rate is 1bps, giving 1667 bytes/sec
packetization is 20ms, giving 34 bytes per packet
IAX header is 4 bytes
UDP header is 8 bytes
Then if let say instead of buying TDM400P cards I get this : Clipcomm
CG-410 Quad FXO Gateway
is it any good? They also sell Quad FXS Gateway.
Clipcomm seems to sell the cheapest Quad FXO/FXS Gateways arround so I'm
wondering if it's working fine with asterisk.
I found this one too but at a
There is already one, you can find it here :
ftp://ftp.asterisk.org/pub/asterisk/webmin
But I never managed to make it work, maybe it should be updated
Anybody wanna take the challenge ? :)
BTW, I've done some web pages that show you your configuration, and
let you edit the text files in your
hi,
i have 1 PSTN line and ip or analog phone
i need get call(with phone ip or analog) from PSTN and transfer it(i.e. to
sales) to the asterisk on corporate network
pstn - gw - asterisk
|
phone
can you recommend me some hardware (cheapest than PC+fxo card+asterisk)?
Last concern about making my channels in a group and add that group in
my dial plan. How can I make sure it will start with channel 4 and not
pick a random one between the 3 channels as I'm pretty sure if I put in
my dial plan a group having channel 2, 3 and 4 it might do the opposite
and
In extensions.conf the order you list the channels for a given dial plan
does not matter, the priority you set for the channel is the order that the
system utilizes.
Can't help you with the other questions. I use Digium T1 cards to a channel
banks.
John Dunham
-Original Message-
From:
Hi,
Case1:
-
-- extensions.conf
exten = 1023,1,Voicemail(101)
exten = 1023/101,1,MeetMe(200)
Case2:
-
- extensions table (using realtime extensions)
++-+--++--+-+
| id | context | exten|priority| app | appdata |
Use lists.digium.com for list browsing
On Thu, 20 Jan 2005 06:47:53 -0600, Matt Schulte [EMAIL PROTECTED] wrote:
It's called asternic, www.asternic.org .. The client is based on flash which
connects to a perl daemon on the server. It uses the manager (manager.conf)
interface to determine
pstn - gw - asterisk
|
phone
can you recommend me some hardware (cheapest than PC+fxo card+asterisk)?
That's the kind of stuff the sipura 3k really shines at.
It offers one FXS, one FXO, 2 VoIP channels and decent routing
capabilities for about $100.
I've never managed to get echo
SIPp has no facility to originate audio/media, it can just send back the
media it receives on its RTP port, more like an RTP proxy.
~Vamsi
On Thu, 20 Jan 2005 14:55:20 +0100, Stojan Sljivic - Pamet
[EMAIL PROTECTED] wrote:
Hi,
Is there a free toll for SIP stress testing that supports RTP?
On Thu, Jan 20, 2005 at 01:16:42PM +1100, Adam Goryachev said:
On Wed, 2005-01-19 at 10:43 -0500, C F wrote:
On Wed, 19 Jan 2005 15:44:44 +1100, Adam Goryachev
[EMAIL PROTECTED] wrote:
On Tue, 2005-01-18 at 20:03 -0600, Eric Rees wrote:
Has anyone been able to find a way to disable
Hi,
Is there any other free tool for SIP testing that has facility to originate
audio/media?
Regards,
Stojan Sljivic
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Vamsi Pottangi
Sent: Thursday, January 20, 2005 15:43
To: Asterisk Users
Roy Sigurd Karlsbakk wrote:
Good day all
We have 2 asterisk servers connected to each other via IAX2 using ilbc.
Each call we make goes up to 25kbit and each one there after 25kbit as
well
bit rate is 1bps, giving 1667 bytes/sec
packetization is 20ms, giving 34 bytes per packet
Actually, iLBC
Andrew Kohlsmith wrote:
On January 19, 2005 12:23 pm, Paul Fielding wrote:
I think you might want to clarify that Best audio quality is in relation to
other highly compressed codecs. Certainly my (albeit limited) experience
is that g711 is much more clear than g729. Compared against gsm, for
I couldn't find this option, I'm running the latest stable there is an
unstable version, is it in that one?
-Original Message-
From: Nicolás Gudiño [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 19, 2005 9:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Thursday, 20 January, 2005 14:42 : Felix Deierlein
[EMAIL PROTECTED] wrote:
Jan 18 16:00:09 WARNING[2919]: Avoided initial deadlock for
'CAPI[contr1/1429092]/128', 10 retries!
2.) Patch to chan_capi
I did not tried it. The patch should solute that problems and enable
faxing? Has anybody
Joe Greco wrote:
911 Testing is a very complicated issue. For a clec it typically
involves scheduling with them so they will expect your call. Also we
frequently use false addresses (that are MSAG resolvable) and some very
sophisticated PSAPs even have fake addresses that MSAG resolve to a
Searching through wiki and google.
http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html
but there are also other products on the
market.
---
Wondering if its possible to connect as
follows:
Extension - Asterisk -
ZyxelAnalogTelephoneAdapter - GSM gateway.
The best way would be to
You *could* write a script to check the voicemail partition and slap
this in a cronjob. If it finds a problem, then have it switch out your
extensions.conf with one that is the exact same except it plays a
voicemail is currently unavailable-type message when you either try to
check your vmail or
Roy Sigurd Karlsbakk wrote:
Hi
Is it possible to somehow monitor/log packet loss and/or jitter in
RTP? I want to know how things look if someone complains about audio.
ethereal can do some of this for rtp, I think. At the very least, if the
endpoint supports RTCP (most do, except for asterisk),
http://bugs.digium.com/bug_view_page.php?bug_id=0003252
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Just for future reference, I think that the ldd command might have helped you
figure out where files are that are being looked for. For example, on my system:
aslan:/home/dana# ldd /usr/sbin/asterisk
libdl.so.2 = /lib/libdl.so.2 (0x40017000)
libpthread.so.0 = /lib/libpthread.so.0
On Thu, 2005-01-20 at 09:18, [EMAIL PROTECTED] wrote:
Last concern about making my channels in a group and add that group in
my dial plan. How can I make sure it will start with channel 4 and not
pick a random one between the 3 channels as I'm pretty sure if I put in
my dial plan a group
Sometimes I have problems and changing to another of their servers makes
it start working again. There probably is a way to make * deal with this
properly. I am using the broadvoice account for test purposes at this
time so I just edit sip.conf and restart * when this happens. What I
have
Hi all.
Somebody knows if AMP will work with the newest version of asterisk(1.0.3)!?
Somehting that I need to know before update!? How is the best way to get my
system updated!?
Thanks.
Denis.
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Hi!
But the only server they gave for sip registration is sip.broadvoice.com I
have several for outbound proxy proxy.chi.broadvoice.com and etc...
Do you have any other for sip?
Best regards,
Helder
- Original Message -
From: Paul [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
I will try out your pages..
Thanks, David
PS I would love to work on your Asterisk Webmin pages, but I don't know
how.
On Thu, 2005-01-20 at 06:01, [EMAIL PROTECTED] wrote:
There is already one, you can find it here :
ftp://ftp.asterisk.org/pub/asterisk/webmin
But I never managed to make
Hi all,
I really hope that you guys can help, because I've been tearing my hair
out for the past 5 hours on this one.
I have a Zaphfc (BRI) card in TE mode connected to the S-Bus of a Nortel
Meridian phone system. Phone calls from the Nortel to say MSN 510 are
correctly being sent to the right
Yes, the Shoreline IP100 is just a rebranded Polycom Soundpoint IP500, and
uses the same software. By default it uses a MGCP image, but it can be
changed to run SIP. See
http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones for more info.
B. J.
-Original Message-
From:
Has anyone testing the maximum limitation for people in a Meetme
conference?
-Brian
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As a Firefighter I would call the 911 Dispatch Center first using there
office number (XXX-). Tell them what you would like to do, what
time, address and phone number. Then give them you cell. If something
happens with the test they will call you back on your cell. Also some
Dispatch centers
Attended transfer, also called supervised transfer, works like this:
While on conversation with another party, you dial ** the transfer
key sequence. Asterisk says Transfer then gives you a dial tone,
while put the other party on hold music. You dial the transferee
number and talk with the
Hi all.
Somebody knows if AMP will work with the newest version of asterisk(1.0.3)!?
Somehting that I need to know before update!? How is the best way to get my
system updated!?
Thanks.
Denis.
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Having seen all the discussions about troubles with faxing I am thinking
of another solution, keeping it all digital.
I want to put two ISDN BRI cards in a faxserver (not running Linux). The
two BRI ports will be connected to a QUAD BRI card in an Asterisk box.
The asterisk box will have a
I believe that there is a Sip software version 1.3.4
available for that phone. The freedomphone.com has
only 1.3.1.
Any idea where to get the latest?
Thanks a lot for your help.
robert
--- B. J. Bomar [EMAIL PROTECTED] wrote:
Yes, the Shoreline IP100 is just a rebranded Polycom
Soundpoint
Robert Augustyn wrote:
Hi,
Is it the same as IP500?
Does it run the same software or do I need to flash
it?
Is so whare do I get it?
Thanks a lot.
robert
It is a Polycom IP500 running MGCP image if you're using ShoreTel. We
just finished a major ShoreTel installation at my work place.
_Update_
Strangely, I've just found that I can dial local (6 digit) numbers using
the '9' prefix - Zap/g2/9742xxx. I'm probably doing something really
daft, but for the life of me, I can't see how I managed to create this
strange scenario.
John
John McEleney wrote:
Hi all,
I really hope that
Sorry about the repost. I got an error in the first one.
Denis.
Em Qui 20 Jan 2005 14:48, Denis Galvão - iSolve escreveu:
Hi all.
Somebody knows if AMP will work with the newest version of
asterisk(1.0.3)!?
Somehting that I need to know before update!? How is the best way to get
my system
Does someone know a free SIP softphone which can be used from a web page
and with Asterisk?
Thanks in advance
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does sipura support analog fax machine (14400 bps) or analog modems?
the cisco ata-186 does support fax machine 9600bps
anyone with a linksys pap2?
thx
Sergio
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Sorry if I missed the beginning of this thread, but I've never heard of
the ** transfer key sequence, nor have I found a way to make it work.
Would you mind, please explaining this further or pointing me to somewhere
where it's documented? (I checked Wiki and Google but no joy.)
Thanks
Bruce
Okay, I'm going to preface this by saying I'm sure I've overlooked something
really basic here. I just need someone to hit me with a clue stick and
point out what I'm missing.
I've got a TDM card with four FXO modules. I've plugged one of them into a
PSTN line. I'm working through the examples
HI
I.m working on echo on my asterisk and I'm wonder if is it possible to use
Zaphfc (BRI) exactly like i4l? How to attach msn number to such card?
Thanks in advance for any help.
Regards,
Corvin
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features.conf
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bruce Komito
Sent: Thursday, January 20, 2005 11:05 AM
To: Asterisk List
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] # Transfers.
Sorry if I
I justed edited the Wiki Asterisk config file features.conf for this
attended transfer features. Please check Wiki again for details.
Best regards,
--JJL44
On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED]
wrote:
Sorry if I missed the beginning of this thread, but
We will put a graphical asterisk load tester online next week.
( i know i said this before, but now its really there :)
zoa.
signature.asc
Description: OpenPGP digital signature
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Does this work with app_queue/chan_agent?
Cheers,
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
List
Sent: 20 January 2005 17:28
To: Bruce Komito
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] # Transfers.
I justed
We will put a graphical asterisk load tester online next week.
( i know i said this before, but now its really there :)
zoa.
signature.asc
Description: PGP signature
signature.asc
Description: OpenPGP digital signature
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In article [EMAIL PROTECTED],
taf taffey [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
Hi All,
Does anyone know of a way to dial two different outbound numbers and bridge
them together using the Asterisk API?
I answered exactly that question on this list within the last two days.
Should
Joseph,
How did the insatllation go?
Any problems?
How do you power this units?
Thanks.
robert
--- Joseph Finley [EMAIL PROTECTED] wrote:
Robert Augustyn wrote:
Hi,
Is it the same as IP500?
Does it run the same software or do I need to
flash
it?
Is so whare do I get it?
Thanks a
I finally got around to signing up with Iaxtel and Free World Dialup...
the price was right, as far as that goes!
I've gotten and placed calls via FWD just fine. But I can't seem to get
registered, or stay registered, with Iaxtel.
My logs show the story; at startup I see:
Jan 20 07:14:44
I have no idea if atxfer works with app_queue/chan_agent. Can anyone try it?
Best regards,
--JJL44
On Thu, 20 Jan 2005 17:38:25 -, Ben Merrills [EMAIL PROTECTED] wrote:
Does this work with app_queue/chan_agent?
Cheers,
Ben
-Original Message-
From: [EMAIL PROTECTED]
Are there any cards
that work with * that do the VoIP-to-TDM processing on the cards, with multiple
interfaces?
The QuickNet
Internet LineJack meets the description I believe, but it only has a single FXS
or FXO. Are there any cards that have more than one FXS?
Thanks.
__
Dana
Olson
Michael, could you provide me with contact information for your versatel
account manager or dutch versatel PRI tech person i could contact?
Joachim.
Michael Devenijn wrote:
Problem solved :
The reason was quite simple ... but annoying :
Interrupts !!! damned !!!
Thank you
Sorry. I don't know
what I'm smoking today.
We need T1
interfaces... :P
So let me rephrase
the question:
Are there any cards
that work with * that do the VoIP-to-TDM processing on the cards, with
multiple
T1interfaces?
The QuickNet
Internet LineJackseems to meet the
description I
First, TURN OFF HTML.
On Thu, 2005-01-20 at 12:53 -0500, Olson, Dana wrote:
Are there any cards that work with * that do the VoIP-to-TDM
processing on the cards, with multiple interfaces?
Hmm, haven't seen any cards with an ethernet and a TDM interface. You
must mean a card that does the TDM
Olson, Dana wrote:
Are there any cards that work with * that do the VoIP-to-TDM processing
on the cards, with multiple interfaces?
The QuickNet Internet LineJack meets the description I believe, but it
only has a single FXS or FXO. Are there any cards that have more than
one FXS?
It's been
Mine works well with a Multimodem ZPX at 14.4kbps. I'm using a
SPA-2000, the linksys should pretty much be the same deal. As I
previously noted on the list, the Sipura fax settings seem to break
faxing so I leave them disabled.
On Thu, 20 Jan 2005 18:13:29 +0100, Sergio [EMAIL PROTECTED]
Thank you everyone. Makes a lot of sense...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 20, 2005 7:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] E911 Testing !
Joe
I'm going to be testing the new realtime stuff further in the next few
days, and just wanted some clarification on a couple of things before I
start on it.
I believe I can store any config file in a external config such as
mgcp.conf for example, by adding it to extconfig.conf with the below
Other then the standard sip debug is there any other
sip debug bugs like for errors, events, etc.
Kurt
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Thanks, Brett, for the info! I actually /like/ the long winded descriptions.
FYI - In some places, the 911 dispatchers are the same people who answer
the Sheriff's, Local PD and Fire phone numbers. So, simply calling the
Sheriff's Dept. and saying that you just installed a new phone system
and
I did look there. If you read my follow up, I screwed up the original question.
What I want is a card with multiple T1 ports that do the processing on the
card, and not on the system CPU.
Is there a mailing list for Asterisk where people treat each other in a civil
manner?
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Dana Olson
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