We would be a hour 0 user. And probably would also be abel to get some
partners to test SS7 interconnect with since it would rid us of a hell of
problems :)
On Thursday 29 May 2003 2:22 pm, Mike M wrote:
On Thursday 29 May 2003 05:27, Patrick wrote:
On Thu, 2003-05-29 at 02:36, [EMAIL
When an incoming call is attempted Asterisk displays the following:
Warning [131081]: chan_sip.c Line 1991: (__transmit_response): Unable to
determine sequence number from ''
I've been seeing this problem within the last little while (sorry I
can't be more specific).
This probably isn't going
Hi list,
I have the follow configuration:
===
extension.conf:
===
[pstn]
ignorepat = 0
exten = _0,1,Dial(${TRUNK}/${EXTEN:1})
[default]
exten = 120,1,Dial(IAX/[EMAIL PROTECTED])
include = pstn
But, when I dial from my gnophone something like 097991269, asterisk
Hi all,
ISDN is not an option for me... I don't want to subscribe to ISDN services
just to use Asterisk.
I have just tested a Cisco 1700 router as a 2 x FXS + 2 x FX interfaces.
It is connected to Asterisk through ethernet.
I have connected 2 phone lines and two analog phones.
The box is seen by
Title: Message
Potential Bug? CSV
as of yesterday.
Scenario: When Music
on Hold is enabled, initiating a transfer from a Cisco 7960 using its built in
transfer function (either transfer or blind transfer) to an analog phone on a
TDM10B, the transfer is not sucessfull.
The analog phone
On Thursday 29 May 2003 09:38, Michael Bielicki wrote:
We would be a hour 0 user. And probably would also be abel to get some
partners to test SS7 interconnect with since it would rid us of a hell of
problems :)
:-)
I've been following the 2-4 port T1 cards thread closely because
that's the
I am ordering T1-PRI service from local service provider and have a few
questions.
Is there framing and coding considerations (or is it all one standard), if
so what is best?
How are calls routed based on DIDs - are these just dtmf tones passed after
the call is picked up and treated as normal
[EMAIL PROTECTED] writes:
Easy solution -- Have * talk to SS7-certified equipment. Cisco comes to
mind.
have you checked the price of e.g. cisco sip/ss7 gw lately? i did a few
months ago and it was huge.
-- juha
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hi all
what sort of fault tolerance (if any) exists for asterisk?
roy
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http://lists.digium.com/mailman/listinfo/asterisk-users
On Thu, 2003-05-29 at 10:04, John Harragin wrote:
I am ordering T1-PRI service from local service provider and have a few
questions.
Is there framing and coding considerations (or is it all one standard), if
so what is best?
How are calls routed based on DIDs - are these just dtmf tones
B8ZS/ESF I believe is the usual for a PRI
DID calls in asterisk are routed just like dtmf dialed extensions, but there
are not DTMF tones passed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Harragin
Sent: Thursday, May 29, 2003 11:05 AM
To:
What price range are you looking for? Does anybody know if the FXO port
of the Dlink DVG-1120 would work?
http://www.dlink.com/products/voiceservices/dvg1120/
Have you considered a S100U and one of those $35 FXS to FXO converters?
It is a nice thing overall, but I still need something much
The Snom200's sourced from the UK are apparantly patched (hardware) so that
the PC headset plugs will work.
I haven't used the PC headset plugs, but a colleague using an RJ10 (call
centre type) headset into the bottom, reports that it works ok.
Tan
- Original Message -
From: Simon
Our prices (for singles):
SNOM 100: £169+VAT (free delivery)
SNOM 200: £189+VAT (free deliver)
Personally (as I'm sure the guys on this group would agree) I would go
for the SNOM 200. Look on the emailing list for a lengthy discussion
on this subject.
Tan
Do you have a website?
On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote:
B8ZS is required for PRI. It's a digital service and can not handle the
loss of data required for AMI.
I wasn't aware that AMI lost data. AMI just inverts polarity on the line
for every other 1. B8ZS does the same thing but intentionally
Good day,
Our installation needs a robust ACD application (as I'm sure others
do) that can be dynamically reconfigured (if possible) maybe by a MySQL
database. I have looked at Bill Heckel's ACD work and Andreas Otto's
DynExtendb as well as James Sharp's ACD. None of these seem to be quite
Try running asterisk like this:
screen -d -m asterisk -vvvc
or
screen -d -m asterisk -c
or
screen -d -m asterisk -f
Martin
On Thu, 29 May 2003, Tjardick van der Kraan wrote:
When we have the G.729 codec (ordered from digium) active in * we have the
following problem:
running * in standard
Do you have your zap channel in asterisk when you type zap show channels
?
If not than make sure you have a proper config files (zaptel.conf
zapata.conf)
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
Hi list,
I have the follow configuration:
===
extension.conf:
Lets say that your E1 channels are assinged to
context=incoming
channel = 1-15,17-31
Then in extensions.conf in context
[incoming]
exten = fax,1,Dial,Zap/1 ;if your Zap/1 port is FXS where the fax
;is attached
(all other extensions)
regards
Martin
On Thu,
I am trying to get asterisk to dial this address
:
sip:[EMAIL PROTECTED]
Using a softphone on my PC
(217.168.168.49)
it dials immediately and I get a voice prompt
..
I have configured an extension, 1303 on
asterisk,
modifying the demo configuration :
exten = 1303,1,Dial(SIP/[EMAIL
Hi,
I look for something in the price range of a X100P for one FXO port.
regarding the Dlink device, I think that there is not a real FXO port, more
somethink like in Actiontec's InternetPhoneWizard, just to be able to use
the analog phones for both IP and PSTN calls.
It just switch one of the
Ok, I want to be able to set a different callerid for each Zapata
channel.
-[zapata.conf]-
callerid=Reception 0
channel=3
callerid=Batman 2000
channel=4
callerid=Robin 1001
channel=5
callerid=The Joker 1002
channel=6
group=2
channel=3-6 ;TDM10B
Whenever I dial an extension, the callerid
Hi,
Check to have a common set of codecs.
If X-Lite is used and at the other end is a phone
without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if
X-Lite).
BR,
Dan
- Original Message -
From:
Dave Alan Caruana
To: [EMAIL PROTECTED]
Hi All,
With the help and patience of this forum, I have been able to set my
asterisk box to make outbound
calls to iconnecthere. My intention is to make two such calls and bridge
them( three way calling) . Based on a earlier suggestion, I have created two
accounts with iconnect and have
On Thu, 2003-05-29 at 13:17, Derek Beaumont wrote:
Ok, I want to be able to set a different callerid for each Zapata
channel.
-[zapata.conf]-
callerid=Reception 0
channel=3
callerid=Batman 2000
channel=4
callerid=Robin 1001
channel=5
callerid=The Joker 1002
channel=6
group=2
Hi all,
For some reason VSAT or Satellite Internet services are not mentioned
(or searchable) in this list's archives. I thought I'd let you know
that I tested Asterisk using IAX (not IAX2) to make a phone call from
an analog phone hooked up to an Asterisk system behind a Linksys router
The zapata.conf file is parsed from the top down, so Asterisk uses the
value since the last channel keyword.
so Asterisk really only sees:
callerid=The Joker 1002
group=2
channel=3-6 ;TDM10B
in your zapata.conf file
Jeremy McNamara
Derek Beaumont wrote:
Ok, I want to be able to set a
Then propably your board stoped taking interrupts. Try changing the PCI
slot or IRQ. Make sure you don't run X-windows.
regards
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
On Thu, 29 May 2003 11:41:01 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
Do you have your zap channel
What bandwidth do you have available for you connection (upsteram and
downstream)? Do you have any CIR for VSAT connection ?
Martin
On Thu, 29 May 2003, Jim Ockers wrote:
Hi all,
For some reason VSAT or Satellite Internet services are not mentioned
(or searchable) in this list's archives.
On Thu, 29 May 2003 12:08:32 -0700
Andrew Gillham [EMAIL PROTECTED] wrote:
Does it work without the group? e.g. Zap/1
Also, does 'zap show channel 1' look ok?
-Andrew
yeap, I tried Zap/1 and it didn't work. :~(
*CLI zap show channel 1
Channel: 1
File Descriptor: 17
Span: 1
Extension:
On Thu, 2003-05-29 at 14:27, Eduardo Goncalves wrote:
On Thu, 29 May 2003 14:08:01 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
Then propably your board stoped taking interrupts. Try changing the PCI
slot or IRQ. Make sure you don't run X-windows.
My box has only one slot. I
Check whether strace -xx cat /dev/zap/1 gives you any output
If it stops and waits than your board is not taking interrupts.
Is the board running on the separate IRQ ?(/proc/interrupts)
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
On Thu, 29 May 2003 14:08:01 -0500 (CDT)
Martin
On 2003-05-28 at 22:39, Mark Spencer ([EMAIL PROTECTED]) wrote:
exten = 4044633/_213.,1,OurApp,losangeles-queue
exten = 4044633/_.,1,OurApp,default-queue
Take out the _. rule and just leave it 4044633 and it should work fine.
That did it. Works great! Thanks.
Not postive the _ is
On Thu, 29 May 2003 14:32:37 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
Check whether strace -xx cat /dev/zap/1 gives you any output
If it stops and waits than your board is not taking interrupts.
Is the board running on the separate IRQ ?(/proc/interrupts)
Martin
The command
Hey Jim,
All sounds good.
We tried a satellite system here a few months ago but couldn't get on with
it. Glad you've had more success. In theory, it shouldn't matter whether the
TCP/IP link between your sites is going over satellite, modem or any other
medium but the issues we found with
On 29 May 2003 14:32:01 -0500
Steven Critchfield [EMAIL PROTECTED] wrote:
What MB are you using, and what chipset is on it?
Silicon Integrated Systems [SiS] 620
Eduardo
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[EMAIL PROTECTED]
So now that I finally relize that you're using T1 or E1 circuit
Do you have a ISDN PRI or an analog ciruit ?
What's the status of the span in zttool or in (/proc/zaptel/1).
Is it OK, RED, YELLOW ?
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
On Thu, 29 May 2003 14:32:37 -0500
On Thu, 29 May 2003 14:32:37 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
Check whether strace -xx cat /dev/zap/1 gives you any output
If it stops and waits than your board is not taking interrupts.
Is the board running on the separate IRQ ?(/proc/interrupts)
Sorry Martin, I
Didn't you just write a post before that it was running ?
The EBUSY means that you propably have asterisk running and the port is
busy or you have strace line on some other console
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
On Thu, 29 May 2003 14:32:37 -0500 (CDT)
Martin Pycko
On 29 May 2003, Steven Critchfield wrote:
On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote:
B8ZS is required for PRI. It's a digital service and can not handle the
loss of data required for AMI.
I wasn't aware that AMI lost data. AMI just inverts polarity on the line
for every other
On Thu, 29 May 2003 14:58:09 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
So now that I finally relize that you're using T1 or E1 circuit
Do you have a ISDN PRI or an analog ciruit ?
What's the status of the span in zttool or in (/proc/zaptel/1).
Is it OK, RED, YELLOW ?
Martin
Martin,
What bandwidth do you have available for you connection (upstream and
downstream)? Do you have any CIR for VSAT connection ?
I think we have 400Kbps downstream and 56-112Kbps upstream. No CIR that
I know of, it's first come first served for the bandwidth, and it's all
shared all the
So it means that the board is working all right but there is problem with
the telco or you're using diffrent signalling for your circuit.
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
On Thu, 29 May 2003 15:06:12 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
Didn't you just
I would like to take advantage of my soundcard/OSS system but so far
haven't come on to examples of what the specs would look like,
particularly for bridging a call onto the console.
Also I wonder whether the kernel version of OSS works all right for
this, as opposed to the official OSS
On Thu, 2003-05-29 at 15:06, Charles E. Youse wrote:
On 29 May 2003, Steven Critchfield wrote:
On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote:
B8ZS is required for PRI. It's a digital service and can not handle the
loss of data required for AMI.
I wasn't aware that AMI lost
On Thu, 29 May 2003 15:26:25 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
So it means that the board is working all right but there is problem with
the telco or you're using diffrent signalling for your circuit.
Martin
I've just called my telephony provider and reliaze that the
On Thu, 2003-05-29 at 15:46, Steven Critchfield wrote:
On Thu, 2003-05-29 at 15:06, Charles E. Youse wrote:
On 29 May 2003, Steven Critchfield wrote:
On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote:
B8ZS is required for PRI. It's a digital service and can not handle the
loss
Remember that a ping is round trip so the other user should only
experience a 325ms delay on a 650ms circuit. What you would be
expieriecing is the overlap in conversations as a result of the delay.
i.e. when someone stops talking, it takes about 300ms for the other side
to start getting the
I think they are hardcoded. But what do you exactly refer to by
signalling bits ?
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
On Thu, 29 May 2003 15:26:25 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
So it means that the board is working all right but there is problem with
I think this is the company that makes them but it is hard to tell.
http://www.artech.com.tw/html/english/AX300/AX300.htm
This company sells them
http://www.aislecom.com/
A rep. for them posted this thread, claimed to be the manufacturer.
On Thu, 29 May 2003 16:16:29 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
I think they are hardcoded. But what do you exactly refer to by
signalling bits ?
Martin
Bits To tell the status of a channel.
It's four (ABCD) Transmit/Receive signaling bit patterns for the Idle and Seized
Hello All,
I'm having a weird problem when connecting up to a TA 750 from adtran. The
problem I'm seeing is that the third wire on my 66 block is behaving as the
tip (or ring) for every extension. Is this indicative of a bad BCU? The
only extension that works properly is extension Zap 2.
you sure you don't have a multiplying block (I use the nordx stuff but I am
sure there is an equivalent on every manufacturer's stuff)
did this once accidentally, and was so pissed when I realized the problem,
I made sure that block would never get used again.
I check the product id every time
Steven Critchfield wrote:
On Thu, 2003-05-29 at 15:46, Steven Critchfield wrote:
On Thu, 2003-05-29 at 15:06, Charles E. Youse wrote:
On 29 May 2003, Steven Critchfield wrote:
On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote:
B8ZS is required for PRI. It's a digital
Ok, so I figured out my problem with my pt480s. But, now I have a few more.
1. When I dial into the voicemailmain or voicemailmain2 application, the
phone and * start talking adsi, but then the phone tells me programming
download canceled, services is full., but my services list isn't full, only
Hi Jim,
This is an interesting product, especially for Cisco ATA-186 users..they can
use one of the FXS ports to connect to the PSTN, but. you have a very
limited functionality: when you call the phone number allocated to that
specific port, you will get the tone for the PSTN line and can
One more thing which can be a big issue with this device.
It hangs the line ONLY based on busy tone... if not correctly detected, then
it will keep the line open for ever, or you can select a call limit
(15/30min.)/
Dan
- Original Message -
From: Jim Flagg [EMAIL PROTECTED]
To: [EMAIL
hi all
I'm trying to use the manager interface to do some nagios (http://nagios.org/)
integration, and I find some parts of it not really optimal. What I'd like to
change, is to make \r\n\r\n an actual terminator, something it isn't today,
AFACS. Below is the Status output - it shows Response,
hi!
anyone try siemens optipoint 400 economy SIP phone with * ?
--
http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf
Thomas
___
Asterisk-Users
Has anyone found a solution how to use the directory button on the Cisco
7960?
If configured correctly it should point to an external directory url. So far
I
failed to find any documentation regarding the format to set up a phone
directory
on my asterisk server.
How can the dial tones on a
Hi
all,
Just a quick
one. Should I be able to call myself through IAXTEL using my 1700
number? I'm behind a NAT firewall and can call other numbers, I just want
to test incoming calls somehow to make sure I can accept them from
IAXTEL.
Regards,Jamie Carl
Email:
I concur! It would also help in parsing out the occasional junk I get on the
socket.
(I'm currently writing a wxwindows version of gastman)
Also... I'm still not sure wheter I can be absolutely sure that the
Responses will always be in the correct order...
-Original Message-
From: [EMAIL
Thomas,
On Fri, 2003-05-30 at 08:22, Tomaz Izanc wrote:
hi!
anyone try siemens optipoint 400 economy SIP phone with * ?
Yes, it works pretty well and has message waiting indication.
and has rfc2833 if you apply a workaround.
But you need:
- Patch to replay contact address as is
*
morning roy,
yes, it's possible. the settings will move into the global section
in 0.2.2.
actually there is a use for a per-device gain configuration. you might
like to have a capi device for outgoing calls to SCREAM at people
(txgain=10) ... ;-)
but i will add an option in the global section
Hi,
How can the dial tones on a CISCO 7960 be modified? Compared to the ATA
186,
I
could not find any settings that make a change possible.
Go to Settings SIP configuration 9 (Out of Band DTMF)
You can choose between avt, avt_allways and none
BR,
Dan
- Original Message -
From:
hi,
we are experiecing the following probem, if anybody
have come across such a problem or a solution to this please let us
know.
our set up is, an
Asterisk server equipped with,4 port
station interface card,single port fxo card and several soft sip
phones
we have found problems with the
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