i don't think there are channel driver's for dialogic cards yet...
On Thu, 23 Dec 2004 09:49:54 +0200, Tasos Daskalopoulos [EMAIL PROTECTED]
wrote:
Hi there
I have a Dialogic VFX/40 ESC plus installed on Redhat Linux 8.0
and looking for Channel drivers for this Card.
where cann i found
who can tell me how to do TDM over enthernet ?
pc a connect pc b only use TDM card?
thank you
John.
--
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED]
: 20041223 11:47
: asterisk-users@lists.digium.com
: Asterisk-Users Digest, Vol 5, Issue 336
Send Asterisk-Users
Hello,
I have some issues with Asterisk, and I see them fixed in bug
database. It is also said the fix is updated in CVS. However, when I
checkout the most current release from CVS, I cannot find the
modification as described in the difference file in the bug database.
How can I apply the
As SIP phones become more prevelant, it would be great to have
dialplans that recognized SIP URIs.
latest CVS
you can use ${SIPURI}
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OK with CVS-HEAD-12/22/04-09:47:32 incoming (to ata) sip calls on a
ATA0186 now seam to work fine. However transfers still do not work.
With CVS-HEAD-12/22/04-12:46:47 transfers still do not work.
Fixed and working in the latest CVS
tested with cisco 7960 and 7905
Hi everyone,
I'd like to implement with * a Reservation call on busy service.
Have you got any ideas? Does exist any application that can do this? ...
or any perl script+ AGI module?
by Gianluca
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Hello,
Is there any possibility to get full calling party INVITE SIP headers
into AGI script somehow?
I'm using SER as * guard - all calls are passed to SER, and it decides
where to send the
connection. I'd like to set some (or only one) headers at SER - which
is not a problem, and then parse
Trying now to set up the final part of my * switch. I must admit I've had
great fun over the last week or so playing with it, and would like to thank
you guys for all the assistance (past and present, since I've been trawling
a lot of old posts!!!).
Scenario - using voiptalk.org to supply the
Rich Adamson schrieb:
Rephrased: Why do folks think they have to use Answer in the sequence
when Playback (etc) is _not_ used?
Because they don't think or they love the telephone companies ...
I think its stupid because the call is established and the caller has to
pay for it even when the call
Using IAX as recommended by most - and therefore my IAX config goes
somewhere along these lines:
[general]
bindport=4569
bindaddr=192.168.1.150
language=en
bandwidth=low
[voiptalk]
type=peer
username=username
secret=password
host=iax.voiptalk.org
qualify=yes
[08450number]
type=friend
Hi,
-Original Message-
I am looking for a message onto my mobile phone, when I got a
voicemail
on my Asterisk!!
I have on my mobile phone ICQ, AOL, MSN, YM and Jabber.
Is there an application, that just send me a message like the email
notification?
You can hook up prety
Hello everybody,
Ive been pulling my hair for a week now over a problem, and I really dont
know where to look anymore. Heres my setup:
There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I
can use it to send and receive calls from physical phones attached to it.
I have setup
Silviu Herchi wrote:
Hello everybody,
Ive been pulling my hair for a week now over a problem, and I really dont
know where to look anymore. Heres my setup:
There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I
can use it to send and receive calls from physical phones attached
Ken Godee wrote:
More than happy to try and share my set up, but
I'll be out of reach to my system and notes till
first of the year.
Thanks for the offer Ken, hopefully we'll get it figured out. Have a
good time, and I'll leave a message as to how it went.
Doug
At 07:00 PM 12/22/04, you wrote:
What registration failure is that?
from the asterisk messages log:
Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25'
The only way to tell is a complete SIP trace of what's going on.
That may be, but the point is when the registration failure
On Thu, 2004-12-23 at 00:12 +0100, Thierry wrote:
Hi
I have something like this but it's in french and it uses teh res_config
Best regards
Thierry wehr
Thierry,
If you are willing to share your billing solution with the community,
I'm sure there will be people pitching in to translate
Fabian Stelzer wrote:
i don't think there are channel driver's for dialogic cards yet...
On Thu, 23 Dec 2004 09:49:54 +0200, Tasos Daskalopoulos [EMAIL PROTECTED]
wrote:
Hi there
I have a Dialogic VFX/40 ESC plus installed on Redhat Linux 8.0
and looking for Channel drivers for this Card.
Hi ALL;
I am new to IAX2, I tried to register an endpoint
to Asterisk with phone number(E.164 address)but it failed.
Can you plz send the desired iax.conf to register a
user on Asterisk.
Regards
Mohammad
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YOu can get commercial licenses from Digium as far as I remember for a
dialogic channel driver
Thanks
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To UNSUBSCRIBE or
Sorry, I mistakenly sent my mail before it was complete... Here it is again.
--
Subject: One-way audio in incoming calls with Asterisk + OpenGK +
Innovaphone IP3000
Hello everybody,
Ive been pulling my hair for a week now over a problem, and I really dont
know where to look anymore. Heres my
Qui, mais je ne parle pas français ;)
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Patrick
Verzonden: donderdag 23 december 2004 13:28
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: RE: [Asterisk-Users] Asterisk billing solution
Hi all.
Does anyone integrated a Siemens Hicom 300E with Asterisk using FXO
interfaces!?
I created an extension group in Hicom and connected my 4FXO(TDM04B) into the
telefony internal network.
What issues I have to care about it!?
Thanks for any help!
Regards,
Denis.
On Thu, 2004-12-23 at 11:00 +0100, Sergio Chersovani wrote:
OK with CVS-HEAD-12/22/04-09:47:32 incoming (to ata) sip calls on a
ATA0186 now seam to work fine. However transfers still do not work.
With CVS-HEAD-12/22/04-12:46:47 transfers still do not work.
Fixed and working in the latest
On Thu, 2004-12-23 at 10:09 +0100, B G wrote:
Hello,
I have some issues with Asterisk, and I see them fixed in bug
database. It is also said the fix is updated in CVS. However, when I
checkout the most current release from CVS, I cannot find the
modification as described in the difference
Hello Rich,
First of all, thank you very much for your help and patience.
I've just arrived home from work (yes, I'm one of the midnight oil burners
:-))
and implemented and tested your suggestions; unfortunatelly it didn't work,
the
same behaviour remains.
More details follow
Silviu Herchi wrote:
Sorry, I mistakenly sent my mail before it was complete... Here it is again.
--
Subject: One-way audio in incoming calls with Asterisk + OpenGK +
Innovaphone IP3000
Hello everybody,
Ive been pulling my hair for a week now over a problem, and I really dont
know where to look
Is there some reason the sipbuddies table structure was
designed with sip config values as column names?
Doesn't look very flexible
It really should take the form of ast_config so when
a new sip feature is implemented, you don't have to
re-write the entire data structure too.
Regards
Greg Cirino
Is there any winblows softphone available offering g729 *and* IAX?
I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones
The best choice should be dIAX, but it is only GSM.
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I've ran into a strange problem
Using current HEAD version (CVS-HEAD-12/22/04-20:45:05), Whenever anyone
calls in to my FWD number it locks the machine up good. If I go back to
the stable version (CVS-v1-0-12/23/04-07:05:33) It works fine.
I've recompiled the current HEAD version (make clean),
Hi Hermann,
- Original Message -
From: Hermann Wecke [EMAIL PROTECTED]
Is there any winblows softphone available offering g729 *and* IAX?
I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones
The best choice should be dIAX, but it is only GSM.
DIAX is supporting now aLaw, uLaw,
Apparently, the realtime system in asterisk is faulty.
Implementing realtime, begins a host of seeding messages
along with registration messages visible at the CLI prompt.
This is not the case with .conf file configuration
Unfortunately, it is not clear where the bug originates
but is shows it's
On Thu, Dec 23, 2004 at 11:45:04AM +0100, Michael Vogel wrote:
Rich Adamson schrieb:
Rephrased: Why do folks think they have to use Answer in the sequence
when Playback (etc) is _not_ used?
Because they don't think or they love the telephone companies ...
Ok, this is probably a stupid
It was written the way it is because that is how RealTime works. =P If you
don't like the schema design, talk to Mark so he can rewrite RealTime for
you.
Read up some more on how RealTime works then you will understand why all the
tables are designed the way they are.
Read docs/README.extconfig
On Thu, Dec 23, 2004 at 02:10:12PM +0100, E. Versaevel wrote:
Qui, mais je ne parle pas français ;)
On Thu, 2004-12-23 at 00:12 +0100, Thierry wrote:
Hi
I have something like this but it's in french and it uses teh res_config
Best regards
Thierry wehr
Thierry,
If you are
At 09:53 AM 12/23/04, you wrote:
It was written the way it is because that is how RealTime works. =P If you
don't like the schema design, talk to Mark so he can rewrite RealTime for
you.
Read up some more on how RealTime works then you will understand why all the
tables are designed the way they
Has this version improved the install process from what it was, to
something where a guy with average intelligence (AKA dummy) can install
without the need of a consultant.
Happy Holidays
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan
Andrei,
Do you have X-Windows running on the linux box? I had a similar issue
that was eliminated when I stopped this process and samba from running.
Now samba is only allowed to come up during non-business hours, for
changing BG music.
Also, make sure your registration period in either
Hi,
I have 3 PAP2 connected to*, they work fine
but there are some things which I would like to improve, some of them
are:
- double ring tone when placing
a call (I hear two tones it seems like the PAP2 is generating it's own
tone)
- some kind of noise (like
glitches or something) when I
For those of you who may be interest
IAX2 loads are now available for the standard builds...
http://www.aredfox.com/edownloadsiax2.htm
Just a word of caution...
Remember to change the ports over to 4569 from whatever.
And don't forget to grab the palmtool from
Rephrased: Why do folks think they have to use Answer in the sequence
when Playback (etc) is _not_ used?
Because they don't think or they love the telephone companies ...
Ok, this is probably a stupid question ;)
If I have a setup like the following:
One TDM400P with one FXO
I have users that use Cisco 7960 with asterisk, if a user sets the
call forward on the phone than asterisk will just go to that extension
priority one. However I want to put in some logic in the Dial Plan
that if the call was either transfered or if it was call fwd (302
moved temporaryly) I should
just poor data modeling
How so? How would you change it? Are you aware that they have written
code into app_voicemail.c that allows you to store the actual soundfiles for
voicemail in the database itself? You want to talk about poor database
design...sheesh..
Read it, makes no difference,
Even though they make the cards and advertise that they support data modes,
digium won't support data mode on the $500 card they sold to me, so I must
turn to the list.
Has anyone configured a T100P to use a T1 strictly as bandwidth? Is there a
HOWTO somewhere? Wiki has nothing I could find. I've
Firmware v 4.63 has been released on the Uniden website. No docs yet to
explain the extras.
Does anyone know how to turn off the call logs on the phone? It's very
annoying in my SOHO environment as all incoming calls ring this
phone(business line and personal line) and then if you pick up the
On December 23, 2004 10:37 am, Matthew Boehm wrote:
Even though they make the cards and advertise that they support data modes,
digium won't support data mode on the $500 card they sold to me, so I must
turn to the list.
If Digium won't support it return the card and get a Sangoma A101u, it's
i have problem with upgrade
i have phone like this http://www.voip-info.org/wiki-Atcom
tested firmware is
http://www.aredfox.com/download/English/program/iax2/PA168S.zip
from debug
192.168.1.100: PHONE
192.168.1.100: V1.37.008
192.168.1.100: Updating ...
Please Wait
192.168.1.100: upgrade
1) I would verify the pin outs of that cable. (I have the tools to make my
own and would do that myself.) And also try a straight cable.
2) Try SF/D4 AMI to make sure that the Zphone and the T100P don't have a
secret compatibility problem trying to do ESF/B8ZS.
Let me know if you get it to
Matthew Boehm wrote:
How so? How would you change it? Are you aware that they have written
code into app_voicemail.c that allows you to store the actual soundfiles for
voicemail in the database itself? You want to talk about poor database
design...sheesh..
So, providing a uniform access model
Hi,
I have a few accounts with sipgate.co.uk to get some different DiD
numbers. However, when an incoming call comes in, it seems to pick the
wrong peer from sip.conf which sends the call into the wrong context and
it fails because there is no extension in that context to match the
register.
At 10:32 AM 12/23/04, you wrote:
just poor data modeling
How so? How would you change it? Are you aware that they have written
code into app_voicemail.c that allows you to store the actual soundfiles for
voicemail in the database itself? You want to talk about poor database
design...sheesh..
Hi Jared,
Thank you for your reply. That server is for asterisk only, things like
X-windows and samba were not even installed there. I limited what I
could from system point of view. Digium support has qualified the box as
clean for TDM400P operations.
It is not clicks and pops, it's just
At 10:37 AM 12/23/04, you wrote:
Even though they make the cards and advertise that they support data modes,
digium won't support data mode on the $500 card they sold to me, so I must
turn to the list.
Has anyone configured a T100P to use a T1 strictly as bandwidth? Is there a
HOWTO somewhere?
Hi,
Currently * is registered to 1 FWD #. If that line is busy people can't
call in? Do I need multiple FWD registrations?
Regards,
Norman Zhang
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At 10:43 AM 12/23/04, you wrote:
On December 23, 2004 10:37 am, Matthew Boehm wrote:
Even though they make the cards and advertise that they support data modes,
digium won't support data mode on the $500 card they sold to me, so I must
turn to the list.
If Digium won't support it return the
Has anyone configured a T100P to use a T1 strictly as bandwidth? Is there a
HOWTO somewhere? Wiki has nothing I could find. I've got plently of public
IPs I can assign to it but don't know how.
Matthew,
I fought with this for quite a while, too. I feel your pain. There
is a wiki page that
-Original Message-
i have problem with upgrade
i have phone like this http://www.voip-info.org/wiki-Atcom
tested firmware is
http://www.aredfox.com/download/English/program/iax2/PA168S.zip
from debug
192.168.1.100: PHONE
192.168.1.100: V1.37.008
192.168.1.100: Updating ...
Hi,
I'm struggle to set up a simple round-robin queue, where I can
determine the order in which the members are tried.
My queues.conf is currently:
[TechSupportQ]
strategy = roundrobin
context = techSupportQueued
timeout = 10
retry = 1
wrapuptime=3
maxlen = 0
joinempty = yes
member = SIP/405
So, providing a uniform access model that doesn't depend on file paths,
capitalization, or numbering is a bad thing?
I'm just speaking from that store-binary-data-in-database war that
went on on the -dev list. I personally have no experience in storing binary
data in a database so I guess I
Kanuri, Seshu (Company IT) wrote:
Has this version improved the install process from what it was, to
something where a guy with average intelligence (AKA dummy) can install
without the need of a consultant.
The Installation Guide provides step-by-step instructions:
For incoming calls, Asterisk matches peer's on IP, meaning that the
first peer it finds will match. This is the *last* one you have in
sip.conf. The context given in that peer must have *all* extensions you
need for incoming calls, which is the extension at the end of the
register= line in the
On December 23, 2004 10:37 am, Matthew Boehm wrote:
Even though they make the cards and advertise that they support data
modes,
digium won't support data mode on the $500 card they sold to me, so I
must
turn to the list.
If Digium won't support it return the card and get a Sangoma A101u,
Hi,
For incoming calls, Asterisk matches peer's on IP, meaning that the
first peer it finds will match. This is the *last* one you have in
sip.conf. The context given in that peer must have *all* extensions you
need for incoming calls, which is the extension at the end of the
register=
On December 23, 2004 10:59 am, Greg - Cirelle Enterprises wrote:
I spoke with a fellow, (can't remember his name, but had a british accent,
there
are only about 10 folks working there) at sangoma, and he specifically said
the
sangoma card will only work with a pri t1 (24channel isdn) not with
Awesome now i'm a minister!
On Wed, 22 Dec 2004 12:45:51 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Alexander Lopez wrote:
Agreed, You have a strong point about the Monopoly aspect of the whole
thing. My .02 would be to have this be a Digium product. Heck, Mark DID
invent the
Hi all,
I'm a bit confused about how to use the switch statement.
I've got an IAX2 link between 2 servers (SA SB).
I have use the switch statement to include extensions from SA onto SB which is
happening perfectly. I've read that I can't use the switch statement the
other way (ie SB-SA) at the
Ok. Thanks a lot anyway. BTW, do you know how many g729 licenses I
need in this situation? Maybe 1 is not enough. Maybe I need 2: 1 for
decoding and one for encoding.
RODOLFO
As the documentation states, you need 1 license for each instance of
g729 in use. That license is used to
I tried to configure a TE110XP using HDLC (mainly to see if it worked) I
called the guys at Digium because HDLC support is not natively compiled,
they told me that HDLC was not covered in the free support provided with the
card, I had to rebuildl the kernel but as I had not that much time
Thanks! :)
On Thu, Dec 23, 2004 at 08:57:22AM -0600, Rich Adamson wrote:
As in many cases with *, there are usually multiple ways to accomplish
a task. Here's a couple that you'll need to tailor to your environment.
[in5100]
exten = s,1,Dial(SIP/sip1,20,tr)
The above assumes the pstn
Hi Jason.
First of all, thanks to become AMP a GPL software!
What about zap channels support in AMP!?
Denis.
Em Qui 23 Dez 2004 14:08, Jason Becker escreveu:
Kanuri, Seshu (Company IT) wrote:
Has this version improved the install process from what it was, to
something where a guy with
At 11:17 AM 12/23/04, you wrote:
On December 23, 2004 10:59 am, Greg - Cirelle Enterprises wrote:
I spoke with a fellow, (can't remember his name, but had a british accent,
there
are only about 10 folks working there) at sangoma, and he specifically said
the
sangoma card will only work with a
Sounds like it could be clipping due to canceling echo on the POTS line.
Just my 2 cents... non refundable btw.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrei
(MPI)
Sent: Wednesday, December 22, 2004 7:09 PM
To: Asterisk Users Mailing List -
On December 23, 2004 11:14 am, TC wrote:
but thats the bitch Mark has put years of blood sweat into it,
now as asterisk start to become much bigger than the single developer/co
how do you divest
that control in a fair/equitable manner
I agree with you on all points -- If Digium needs to
Has anyone had success with the TDM400 in production? I have multiple
boxes where these cards lock up and the only thing that will fix them is
to unload *, modprobe -r wctdm, modprobe wctdm, load asterisk. Does not
matter if it is a FXS/FXO module.
I know this topic has been discussed many times
Hello,
We've released another update to our Asterisk GUI Client suite: 1.0.6
http://astguiclient.sf.net/
Screen shots: http://astguiclient.sourceforge.net/screenshots.html
The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not
Dec 23 09:24:57 NOTICE[12406]: chan_sip.c:7742 handle_request:
Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26'
happens with app_realtime, doesn't happen without realtime
there are previous posts identifying issues with this.
All I can say on this is that I am using
I want it to *always* call SIP/405
first, and if they're busy or don't answer, it should fail-over to
SIP/403 etc.
I've been using RR and what you described is EXACTLY what has been
happening. So I don't know why yours isn't doing that. I'm using agents but
I can't imagine why that would
On Thu, Dec 23, 2004 at 09:58:19AM -0700, Damon Estep wrote:
Has anyone had success with the TDM400 in production? I have multiple
boxes where these cards lock up and the only thing that will fix them is
to unload *, modprobe -r wctdm, modprobe wctdm, load asterisk. Does not
matter if it is a
I also have additional columns in my MySQL sip_buddies table with no problems.
If there is a problem updating, selecting, or inserting into your table then
you should check your MySQL Configuration or verifiy that it is working
properly.
On Thursday 23 December 2004 05:10 pm, Matthew Boehm
Rich Adamson wrote:
Ok. Thanks a lot anyway. BTW, do you know how many g729 licenses I
need in this situation? Maybe 1 is not enough. Maybe I need 2: 1 for
decoding and one for encoding.
RODOLFO
As the documentation states, you need 1 license for each instance of
g729 in use. That
I want it to *always* call SIP/405
first, and if they're busy or don't answer, it should fail-over to
SIP/403 etc.
I've been using RR and what you described is EXACTLY what has been
happening. So I don't know why yours isn't doing that. I'm using
agents but
I can't imagine why that would
What I am confused about the most is this section in /etc/zaptel.conf:
# Next come the dynamic span definitions, in the form:
# dynamic=driver,address,numchans,timing
#
# Where driver is the name of the driver (e.g. eth), address is the
# driver specific address (like a MAC for eth), numchans is
Short answer: try new FXO modules or a new card.
I've struggled with this for about a month. I've returned one TDM400
card, got a new one. Had same problems, Digium support installed a patch
for zaptel, no difference. Then I diagnosed one FXO was dead. Got a
replacement for that FXO.
Hello,
I was curious how people had
the timing setup for their T100P and Total Access 750.
We have been getting Red
Alarms once a day for 5 seconds. I think the line is losing sync and
resyncing.
Currently, it is set like
this:
span=1,1,0,esf,b8zs
Should I set the channel
On Thu, 2004-12-23 at 11:24 -0600, Matthew Boehm wrote:
What I am confused about the most is this section in /etc/zaptel.conf:
# Next come the dynamic span definitions, in the form:
# dynamic=driver,address,numchans,timing
#
# Where driver is the name of the driver (e.g. eth), address is
Hi all,
I have asked this before, but got no reply yet:
Does Call Completion on the Snom 190 work with
Asterisk? I have tried many things but just cannot get it working, I even wrote
to Snom Support in Berlin,
but just got the reply that they did not have a clue, but suggested I
At 11:52 AM 12/23/04, you wrote:
On December 23, 2004 11:14 am, TC wrote:
but thats the bitch Mark has put years of blood sweat into it,
now as asterisk start to become much bigger than the single developer/co
how do you divest
that control in a fair/equitable manner
I agree with you on all
Inline...
As in many cases with *, there are usually multiple ways to accomplish
a task. Here's a couple that you'll need to tailor to your environment.
[in5100]
exten = s,1,Dial(SIP/sip1,20,tr)
The above assumes the pstn line is _not_ sending any digits to you. If
it does,
At 12:14 PM 12/23/04, you wrote:
On Thu, Dec 23, 2004 at 09:58:19AM -0700, Damon Estep wrote:
Has anyone had success with the TDM400 in production? I have multiple
boxes where these cards lock up and the only thing that will fix them is
to unload *, modprobe -r wctdm, modprobe wctdm, load
I understand some of the basic Goto() forms,
such as Goto(context,extension,priority) and
Goto(extension,priority) [within context I presume].
Can someone Explain Goto(6275|1) as found in the
sample extensions.conf? Is this the same as
Goto(6275,1) just with a different delimiting
character?
I wonder if anyone has come across this odd behavour with a T1 PRI using
NI2 signalling from a Nortel switch.
Sometimes, when bringing up a PRI trunk, a channel gets into a state
where asterisk can't request a channel, and gets reason 0, but the
channel is not busy. The only thing so far that
Fabian Stelzer wrote:
i don't think there are channel driver's for dialogic cards yet...
On Thu, 23 Dec 2004 09:49:54 +0200, Tasos Daskalopoulos [EMAIL PROTECTED]
wrote:
Hi there
I have a Dialogic VFX/40 ESC plus installed on Redhat Linux 8.0
and looking for Channel drivers for this Card.
I was curious how people had the timing setup for their T100P and Total
Access 750.
I pull timing from the ta-750 in the un-scientifc belief that the
timer is more accurate than the low end t100p crystal
span=1,1,0,esf,b8zs
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Whoever was listed first in the list always got the call first. This isn't
what I was expecting RR to do. I was expecting call #1 to goto agent 1. if
call 2 comes in and 1 is still on phone it goes to 2. if 1 is not on phone
it still goes to 2. and then 3, 4 etc..until it loops back around.
but
Certification works to show that the person can pass a test, not that
they can do anything in the real world.
...and a Microsoft Certified Systems Engineer is like a Lego Certified
Building Entrepreneur... ;-)
It's Must Consult Some Experienced or possibly Mimesweeper Consultant
and Solitaire
Can you try to add some parameters to zapata.conf before defining channel 3,4
cidsignalling = dtmf
cidstart = polarity
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Should I set the channel bank to
provide timing or receive timing?
My Atlas 550 provides
timing and I set my zaptel.conf to:
span=1,0,0,esf,b8zs
Never had a red alarm, ever.
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Hi,
I'm having trouble getting IAX calls through in one direction. On the
receiving end I see:-
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00014ms SCall: 1 DCall: 0 [1.142.164.215:4569]
VERSION : 2
CALLED NUMBER : 2003
On Thu, 2004-12-23 at 12:35 -0500, Greg - Cirelle Enterprises wrote:
At 11:52 AM 12/23/04, you wrote:
On December 23, 2004 11:14 am, TC wrote:
but thats the bitch Mark has put years of blood sweat into it,
now as asterisk start to become much bigger than the single developer/co
how do
On Thu, 2004-12-23 at 09:40 -0800, Richard Lyman wrote:
Fabian Stelzer wrote:
i don't think there are channel driver's for dialogic cards yet...
On Thu, 23 Dec 2004 09:49:54 +0200, Tasos Daskalopoulos [EMAIL PROTECTED]
wrote:
Hi there
I have a Dialogic VFX/40 ESC plus
I was curious how people had the timing setup for their T100P and Total
Access 750.
We have been getting Red Alarms once a day for 5 seconds. I think the
line is losing sync and resyncing.
Currently, it is set like this:
span=1,1,0,esf,b8zs
Should I set the channel bank to
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