Remarque : message transféré en pièce jointe.
Hello ,
To people who told me I'm cretin !!
Harry
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On Fri, May 12, 2006 at 02:25:44PM +0300, Cosmin Prund wrote:
Unfortunately the latest misdn-mqueue does not compile on my system, it
issues all sorts of blah-blah that I'm interpreting in only one way:
there's a problem with the parameters the Makefile passes to the
compiler (the .h files
Thank you very much for making this patch available!
I'm not all that clear on the development process in Asterisk -- does this
mean that your patch will be present in the next release of Asterisk (1.2.8
or 1.3 or whatever the next version is)? Or do I have to manually apply the
patch every
David Gomillion wrote:
Does anybody know anyone who offers encrypted IAX termination at
reasonable rates? I googled, searched the WIKI, but didn't find a whole
lot of information.
You might want to take a look at link2voip. I don't currently use them,
but was planning to get a DID from them
2006/5/15, Rich Adamson [EMAIL PROTECTED]:
Which fax-modem would you pick if you had to test fax capabilities ?The fax modem is not really the issue with asterisk. By far, themajority of existing analog fax machines installed and being sold today
will function just fine with asterisk.Hi Rich,I
Hi all
I have setup sips accounts to an asterisk server from a
provider, I know that there are using asterisk real time for sip users
definitions.
Sometimes in a ramdom basis I receive:
chan_sip.c:9596 handle_response_register:
Forbidden - wrong password
Hi Armin!
I now tried 2 versions:
A) I also executed make prepare in the kernel sources before building
with make KDIR=/usr/src/kernel-source-2.6.8
B) I compiled with the kernel headers:
make KDIR=/lib/modules/2.6.8-2-386/build/
Both times I got the same error message (see below). It would
Hi,
I cannot make my TDM2400P (with Echo Canceller module) detect faxes. I
tried with a TDM400P and it worked at 80% (20% of faxes were lost). My
test conf.files are:
_zapata.conf_:
context = inbound
faxdetect = incoming
language = it
musiconhold = default
signalling = fxs_ks
callerid =
And some more:
Compiling with 2.6.8 is just for testing. The production system uses
2.4.27-2. On this system, I got other errors:
gcc -D__KERNEL__ -I/usr/src/kernel-headers-2.4.27-2-386/include -Wall
-Wstrict-prototypes -Wno-trigraphs -Os -fno-strict-aliasing -fno-common
On Tue, 2006-05-16 at 08:24 +0200, [EMAIL PROTECTED] wrote:
Remarque : message transféré en pièce jointe.
Hello ,
To people who told me I'm cretin !!
Donc?
--
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Tzafrir Cohen schrieb:
On Tue, May 16, 2006 at 01:09:41AM +0200, Bastian Schern wrote:
Hello,
is it possible to delete global variables during runtime?
Is setting the variable to an empty value good enough? How do you use
it?
I will use it to check if a Caller has already a open
Ralf Schlatterbeck wrote:
On Fri, May 12, 2006 at 02:25:44PM +0300, Cosmin Prund wrote:
Unfortunately the latest misdn-mqueue does not compile on my system, it
issues all sorts of blah-blah that I'm interpreting in only one way:
there's a problem with the parameters the Makefile passes to
Hi,
I am investigating getting a wifi VoIP phone because its may be a better
option than an ATA and a cordless phone..
Does anyone have any experience with the whats out there??
Do they support things like WPA etc??
I have heard the battery life can be a problem.. Is this the case?
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi Armin!
I now tried 2 versions:
A) I also executed make prepare in the kernel sources before building with
make KDIR=/usr/src/kernel-source-2.6.8
B) I compiled with the kernel headers:
make KDIR=/lib/modules/2.6.8-2-386/build/
Both times I
Hi,
I'm using asterisk 1.2.7.1 and somehow my iax trunking is getting these
problem :S.
Sometimes iax acts weird and start to drop calls randomly and give these
at the log:
May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6,
having received INVAL
May 16 13:44:22 DEBUG[5264]
Hi All,How can i use Microsoft SQL server with asterisk, Can the unix ODBC diriver interface MSQL?? and what module would i be using to access ODBC from asterisk??
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Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi Armin!
I now tried 2 versions:
A) I also executed make prepare in the kernel sources before building with
make KDIR=/usr/src/kernel-source-2.6.8
B) I compiled with the kernel headers:
make
steven,,
u not understand the situation,,,
there only have 30 channel for voice,, the other channel resiverdfor control channel..
good luck for you.
On 5/10/06, Steve Underwood [EMAIL PROTECTED] wrote:
陈帆 wrote: there have only 30 channel for E1,,He has defined the correct 30 channels for his
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi Armin!
I now tried 2 versions:
A) I also executed make prepare in the kernel sources before building with
make KDIR=/usr/src/kernel-source-2.6.8
B) I compiled with the kernel headers:
make
It looks like SPA-3000 has both FXO and FXS. You could connect FXO to PSTN
and FXS to fax machiine. Maybe it work.
We have it worked on MG3000-R from www.telecomchinasourcing.com.
hawk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Hi steven,
I am from China. we havemany E1 gateway running
in China including ShangHai.
Inyour case, it is impossible to negociate with
ShangHai Telecom. To debug the E1 link problem, why not use a PBX or E1 trunk
gateway to test the link first. We have successfully used the E1 gateway
WipeOut wrote:
Hi,
I am investigating getting a wifi VoIP phone because its may be a
better option than an ATA and a cordless phone..
Does anyone have any experience with the whats out there??
Do they support things like WPA etc??
I have heard the battery life can be a problem.. Is this
WipeOut wrote:
Hi,
I am investigating getting a wifi VoIP phone because its may be a
better option than an ATA and a cordless phone..
Does anyone have any experience with the whats out there??
The only phone Senao I tested has a problem when roaming between APs.
There's a very high
** Want to see who you're talking to? Video telephony is growing.
A couple of developers has formed the Asterisk Video Task Force in
order to improve the support for video telephony
in Asterisk for the 1.6 release this fall. There is already support
for video in the SIP and the IAX2 channel,
Hello, is there any way to anounce on agents phone (e.g. beep tone like
in gsm), that some call is waiting in queue?
PJ
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On Tue, May 16, 2006 at 11:05:01AM +0200, Klaus Darilion wrote:
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi Armin!
I now tried 2 versions:
A) I also executed make prepare in the kernel sources before building
with
make KDIR=/usr/src/kernel-source-2.6.8
B) I
Tzafrir Cohen wrote:
On Tue, May 16, 2006 at 11:05:01AM +0200, Klaus Darilion wrote:
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi Armin!
I now tried 2 versions:
A) I also executed make prepare in the kernel sources before building
with
make
Hi,
I'm trying to match a few of numbers in a GotoIf; numbers are not starting
with but contain some strings:
GotoIf($[${CALLERIDNUM} =~ 984836|984899|498993|644110]?8:11)
Expression result is always '0'.
Where am I wrong?
Domenico Viggiani
___
On Tue, 16 May 2006, Klaus Darilion wrote:
Tzafrir Cohen wrote:
On Tue, May 16, 2006 at 11:05:01AM +0200, Klaus Darilion wrote:
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi Armin!
I now tried 2 versions:
A) I also executed make prepare in the
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Hi,
I am investigating getting a wifi VoIP phone because its may be a
better
option than an ATA and a cordless phone..
Does anyone have any experience with the whats out there??
Do they support things like WPA etc??
I have heard the battery life can be a problem.. Is this the case?
On 5/16/06, Avi Miller [EMAIL PROTECTED] wrote:
Michael J. Tubby B.Sc (Hons) G8TIC wrote:
call then transfers it on to another extension transferee (recipeient)
sees the Caller*ID
This behaviour changed in Asterisk 1.2 -- add o to your Dial options
and Asterisk will retain the original Caller
Hi Armin!
thanks for the new package - I will try it.
Meanwhile I used the 2.4.27 debian kernel sources, imported the debian
config, and executed make dep
After that I could successfully compile the v3 drivers, but failed
during config:
'DIDD' driver load failed. Please check system
Hi Armin!
With the new v3 package compilation works fine. But I still have
problems when starting the config script:
'DIDD' driver load failed. Please check system installation
(kernel version, missing files)
Do I have to load any modules before? The Eicon Card is found by lspci:
I believe there are quite different levels for the Asterisk market, so
most people who run call centers wont feel confident in downloading a
couple of ISOs from the internet and setting things up themselves.
l.
In data Mon, 15 May 2006 21:47:09 +0200, Steve Totaro
[EMAIL PROTECTED] ha
Hi All,
Is it possible to configure asterisk to play a beep at a regular
interval when a conversation is being recorded / monitored?
There are a number of ways indicating to a user that a conversation is
being recorded, one is to play an announcement, another accepted way is
to play these
[EMAIL PROTECTED] wrote:
I believe there are quite different levels for the Asterisk market, so
most people who run call centers wont feel confident in downloading a
couple of ISOs from the internet and setting things up themselves.
l.
Absolutely correct. I could not agree more.
Senad
Klaus Darilion wrote:
Hi Armin!
With the new v3 package compilation works fine. But I still have
problems when starting the config script:
'DIDD' driver load failed. Please check system installation
(kernel version, missing files)
Do I have to load any modules before? The Eicon Card is
Hi Klaus,
what happens when you try to load the DIDD module
sh# insmod /usr/lib/divas/divadidd.o
and what is the kernel (syslog) saying?
Armin
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi Armin!
With the new v3 package compilation works fine. But I still have problems when
starting the
Sorry! Looks like I got confused and forgot to execute make install!
regards
klaus
Klaus Darilion wrote:
Klaus Darilion wrote:
Hi Armin!
With the new v3 package compilation works fine. But I still have
problems when starting the config script:
'DIDD' driver load failed. Please check
On 5/16/06 12:23 AM, Richard Lyman wisely said:
A.J. Paxson wrote:
On 5/15/06 10:55 PM, Richard Lyman wisely said:
A.J. Paxson wrote:
Hi All!
I've really been struggling trying to get around this. Instead of the same
announcement being played over and over again, I
Hi Armin!
After make install I'm able to run the config script. I configured all 4
ports of the card and saved the configuration.
After that:
...
Load Diva XDI driver ... OK
Check for saved MAINT debug/trace buffer ... idle
Write Diva configuration to CFGLib ... succeeded
Check Diva
This is some bad behaviour of the Config wizard.
You should then start with
sh# /usr/lib/divas/divas_stop.rc
to clean all of it. And then really start the
divas:
sh# /usr/lib/divas/divas_cfg.rc
Armin
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi Armin!
After make install I'm able to run
Armin Schindler wrote:
This is some bad behaviour of the Config wizard.
You should then start with
sh# /usr/lib/divas/divas_stop.rc
to clean all of it. And then really start the
divas:
sh# /usr/lib/divas/divas_cfg.rc
seems to work, thanks.
Is there a utility which shows the status of the
Olivier Krief wrote:
For example, it seems that Brother 8360P uses Super G3 mode.
Is there a fax-modem offering such capability so that I could easily
check if I still cannot hangup when I enable or disable Super G3 mode ?
MultiTech 5634-series and MainPine RockForce fax modems (Agere
That is what Signate does too.
lenz wrote:
I believe there are quite different levels for the Asterisk market, so
most people who run call centers wont feel confident in downloading a
couple of ISOs from the internet and setting things up themselves.
l.
In data Mon, 15 May 2006 21:47:09
I have a question about FOP.
I am using amp since lasto August, and I am happy with it and its new
version FreePBX
Unfortunately, in all the asterisk servers I installed so far (about 10) I
was never able to make FOP correctly running.
I see the extensions, I see the queue, sometimes I also see
2006/5/16, Lee Howard [EMAIL PROTECTED]:
Olivier Krief wrote: For example, it seems that Brother 8360P uses Super G3 mode. Is there a fax-modem offering such capability so that I could easily check if I still cannothangup when I enable or disable Super G3 mode ?
MultiTech 5634-series and MainPine
I know that this is a more strictly FOP related question than Asterisk but
I'd like to know if regexp buttons support a '-' char, i.e.:
[_Zap/1-.*]
...
In fact, I have:
Zap/1 to Zap/10 as incoming channels
Zap/11 to Zap/15, Zap/17 to Zap/21 as outgoing channels
(it is an E1 PRI)
and I'd like
I am using amp since lasto August, and I am happy with it and its new
version FreePBX
Unfortunately, in all the asterisk servers I installed so far (about 10) I
was never able to make FOP correctly running.
I see the extensions, I see the queue, sometimes I also see the trunks, if
they are zap
Thank you very much
I will jump to http://www.asternic.org and search for m y problems
Andrea
Nicolás Gudiño
[EMAIL PROTECTED]
Asterisk can certainly perform under high-stress, high-availability situations like call centers. Keep in mind that while that current generation of call center operators may not be comfortable replacing their equipment with Asterisk, the next generation seems to be ready and willing. It doesn't
After trying multiple times to change the code for res_features.c to read in a
variable for the park timeout destination, I gave up
and hardcoded it into the source.
I attached my patch if anyone needs to do a similar thing.
I am not sure why I had to edit it twice, I would think that the area
Hi dCAP certified people,
I want to do dCAP certification and need advice fro you guys. How
difficult is it to do it? I am working in asterisk for about 2 years,
have installed asterisk systems for a few companies and know quite a bit
about asterisk. Is it necessary to go through asterisk boot
Which fax-modem would you pick if you had to test fax capabilities ?
The fax modem is not really the issue with asterisk. By far, the
majority of existing analog fax machines installed and being sold today
will function just fine with asterisk.
Hi Rich,
I got problems
Ok,
So I just installed some aastra 9133i phones. They work great. One
small problem. When I do an 'all page' anyone who is on the phone
gets their call placed on hold for the duration of the page! THIS IS
NOT GOOD!How do I make the page only go to people who are not on
the phone, and
On Monday 15 May 2006 08:42, Steven wrote:
What is the trick to call it announce the Park position during the transfer
instead of a call back??
*CLI show application ParkAndAnnounce
-= Info about application 'ParkAndAnnounce' =-
[Synopsis]
Park and Announce
[Description]
Speaking as one of those call centers we are looking at doing a turn over to
Asterisk from our Nortel systems and are doing it ourselves. We've looked at
a lot of packages from Fonality, Signate, Aheeva and others and none fit our
needs. Each has good aspects but none have all of what we need.
I use the Zyxel wireless SIP phones with Aerospace (now Cisco) APs and
they work just great. My battery is good for all day. Charge them at
night. Expensive though.
Got them registering to Asterisk, calling Cisco phones via SIP trunks
and they sound great.
Thank you.
Phil Menico
171 Madison
I checked the online documentation
everything seems again correct.
So I tryed to start ./op_server.pl with debug =7
The output seems to be correct, ie when a call arrive it is reported in the
output; if i call from 557 the extension 567, I see a lot of these rooms
127.0.0.1 - Event:
Chris,
When I tried background it waited until the message was done before dialing,
just like playback. Am I missing something?
Thanks,
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt
Sent: Monday, May 15, 2006 4:36 PM
To: Asterisk Users Mailing
Yes you can use MSSQL with the ODBC driver, I have it working for CDR logs, I had to install unixODBC and configure it then use the cdr_odbc.conf file to specify. Check in your source files or svn checkout for a docs folder and the
cdt.txt file.On 5/16/06, Dumpolid Exeplish [EMAIL PROTECTED]
I tested out a UTStarcom F1000 that has WEP support. The only problem I found is that the original firmware dropped calls when roaming between AP's. It is also hard to hear when in a noise enviroment without an earbur or headset of some sort. I have been curious about Linksys 2 new offerings and
On Tuesday 16 May 2006 08:53, Steven wrote:
Perfect would be that is a call is parked by a SIP or IAX extension, it
would return there, but if parked by a ZAP Trunk (legacy PBX), it would
return to ZAP/g2/5100.
exten = 700,1,Set(RTN=${CUT(CHANNEL,-,1)})
exten =
Hi all,
Simple question;
What are the possible reasons for a SIP channel to hang there for hours
after
the call has terminated ?
Calls are sent from 1.2.6 to SER using DeadAGI over the internet, and
yesterday there
were 100 calls that hanged for hours over a total of 15k calls.
Typical
Olivier Krief wrote:
Would you use such modems to run a batch script to check various fax
protocols ( V.34, super G3, ...) ?
You could, yes.
Would it be easy with Hylafax to change modem settings so that first
attempt is made in V.34, second attempt with ECM, third without ECM
etc.. and
Agents beat up on headsets and voip gear is expensive. Look at the
TenorAX or ATAs.
Kevin Savoy wrote:
Speaking as one of those call centers we are looking at doing a turn over to
Asterisk from our Nortel systems and are doing it ourselves. We've looked at
a lot of packages from Fonality,
I discovered the problem
the TCP port 4445 should be opened from browsers to HTTP server runnin
./op_server.pl
I think that this need is not sufficintly pointed out, at least for me !
The FOP now works like a charm. Usually I have separated networks for
VOIP and PC,
I have two IAX trunked *, there are loud crackles and pops,
they are dialing out a T-1 and are sip devices, it also drops words, any help or
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A recent version of Unicall has a small bug in tone generation (a
missmatching use of spandsp library), please try to upgrade to latest
spandsp and unicall libraries, im pretty much sure your problem will
be solved that way.
Regards
On 5/16/06, acriollo [EMAIL PROTECTED] wrote:
Hi every body,
James Harper wrote:
I was looking for something like this a while back (actually, a wifi +
gsm combo), and came to the conclusion that a dect + gsm phone would be
a better option, except that they don't exist (much).
Maybe a VoIP capable DECT base station would be a better option for you?
These
Title: Meetme and authentication
Hi all,
I have thoroughly read the available documentation and I can't seem to find a workaround for my setup
I'm trying to create a phone conference line that users would call using a unique phone number (no matter if they are moderators or just plain
It sounds like you could pass it. The Boot Camp is not strictly necessary.
However I am not sure dCAP would be worth it - I did mine last year and was
rewarded with a little plaque saying that I was dCAP certified for asterisk
v1.0 - I received the plaque about 1 month before asterisk 1.2 came
The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote,
wireless handsets via DECT.
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice - 800.398.VoIP X3402
fax - 716.630.1548
e -
Mark,
While these samples are pretty good they do not work out of the box -
there are a couple of issues:
1. the samples are 44100 samples/second and Asterisk needs them to
be at 8000 samples/second. This is what happens if you prune out all of
the Amercian voicemail prompts and substitute
Hi!
We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb
RAM. It was working 24/7 without any for a month, but for not related causes I
rebooted it a week ago. Yesterday the machine suddenly stop working, with a
kernel panic. We was watching logs, and found in
On Tue, 16 May 2006, Klaus Darilion wrote:
Armin Schindler wrote:
This is some bad behaviour of the Config wizard.
You should then start with
sh# /usr/lib/divas/divas_stop.rc
to clean all of it. And then really start the
divas:
sh# /usr/lib/divas/divas_cfg.rc
seems to
If the patch makes it past all the code checks and whatnot, and any kinks
are worked out of it, it would make it into the next version of asterisk.
Anyone that would want it in the older versions would have to backport it
(I'll see about posting a patch somewhere that'll work for 1.2). We just
Edu wrote:
Hi!
We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb
RAM. It was working 24/7 without any for a month, but for not related causes I
rebooted it a week ago. Yesterday the machine suddenly stop working, with a
kernel panic. We was watching logs, and found
WipeOut wrote:
James Harper wrote:
I was looking for something like this a while back (actually, a wifi +
gsm combo), and came to the conclusion that a dect + gsm phone would be
a better option, except that they don't exist (much).
Maybe a VoIP capable DECT base station would be a better
I'm using Asterisk + OH323 with CCM 3.2. There are a few quirks with codec negotiation and callerid. We're moving as fast as we can to dump the CCM, so we're not too worried about them.
On 5/15/06, Gustavo Souza Queiroz [EMAIL PROTECTED] wrote:
Hello,
I´m have a CCM 3.3 and Asterisk in my
LAN.
I
Hi!
I have problems when receiving overlap calls. I'm used to behaviour of
zaptel. overlap=yes takes care of collecting the digits.
I tried the digit colleciton setting in the diva config tool, but it
stops collecting digits when a certain amount of digits is received
(isntead of waiting a
*snipped
I like the idea that I can just pick up the phone, dial an extension, record
an announcement, and be sure that announcement will play during extended
hold times.
Thanks for the ideas, Richard!
It just came to me, about using timeouts on the queues, play an announcment,
update the
Hello,
I have configured my TDM01B Card (1 FXO Port ) as follows (below) but it will
not pick up an incoming call.
Any suggestions/tools to see what the problem is? I have looked at zttool
where this line changes but I don't understand what it means (The last digit
changed from 0 to 1)
Yes, the dialplan field in the PAP (not asterisk's dialplan) was the
problem. The dialplan used to have *xx in it (as well as lots of other
stuff which we left alone), we changed that to *xxx (leaving the double
*'s in all of the vertical service activation codes) and it now works.
thanks
On Tuesday 16 May 2006 18:38, Pieter Claassen wrote:
Hello,
I have configured my TDM01B Card (1 FXO Port ) as follows (below) but it
will not pick up an incoming call.
I failed to mention that my service provider is UPC (Cable telephone).
Pieter
Hi!
I have problems with the ToN configurations in chan_capi-cm. I
understand how incoming calls are rewritten using national and
international prefix. But for outgoing calls - what is the ToN?
Further, is there any debug info of the Ton? capi debug and the
divactrl dchannel both show the
Has anyone seen this (version 1.2)?
The following dialplan should result in the voicemail
message being delivered two both mailboxes ([EMAIL PROTECTED] and
[EMAIL PROTECTED]);
exten = 0,1,SetCIDName(OPER ${CALLERIDNAME})
exten =
Kevin Savoy wrote:
Speaking as one of those call centers we are looking at doing a turn over to
Asterisk from our Nortel systems and are doing it ourselves. We've looked at
a lot of packages from Fonality, Signate, Aheeva and others and none fit our
needs. Each has good aspects but none have
Matt,The developers are also very receptive to feedback andhave been steadily shaping QueueMetrics to fit our needs.I couldn't agree with you more. One of the things that impressed me most was the great support I got from the QueueMetrics team. Especially when something didn't work quite as we had
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi!
I have problems when receiving overlap calls. I'm used to behaviour of zaptel.
overlap=yes takes care of collecting the digits.
I tried the digit colleciton setting in the diva config tool, but it stops
collecting digits when a certain amount
Greetings,
I'm running 1.2.7.1, and I'm trying to using the REGEX function in my
dialplan. However, the Asterisk parser doesn't seem to understand
what's going on. I'm trying to use REGEX to determine if a variable
matches a standard 10 digit US/Canada number. To do this, I started
with the
You have in /etc/zaptel.conf
fxsks=4
But in /etc/asterisk/zapata.conf
channel=1
I could be wrong but methinks that should be channel=4? I've never set
up a TDM card that didn't start at the first module socket, so I very
well could be wrong. If that doesn't fix it you could move your FXO
Joshua Colp schrieb:
Edu wrote:
Hi!
We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850
with 4Gb RAM. It was working 24/7 without any for a month, but for not
related causes I rebooted it a week ago. Yesterday the machine
suddenly stop working, with a kernel panic. We was
I can't agree more. We are currently evaluating QueueMetrics. They
gave us a second evaluation term so that we could evaluate longer.
We are currently replacing our ?legacy? Nortel. I say legacy cause it
isn't that old... but boy does it lack in features (well unless we
want to buy
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi!
I have problems when receiving overlap calls. I'm used to behaviour of zaptel.
overlap=yes takes care of collecting the digits.
I tried the digit colleciton setting in the diva config tool, but it stops
collecting digits
AFAIK UPC cable phone is a voip phone, and not PSTN
Regards,
Moutaz
-- Original Message ---
From: Pieter Claassen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-
[EMAIL PROTECTED]
Sent: Tue, 16 May 2006 18:59:09 +0200
Subject: Re:
Hi Damon -
The following dialplan should result in the voicemail message being
delivered two both mailboxes ([EMAIL PROTECTED] and
[EMAIL PROTECTED]);
The actual result is an error copying the message to the second mailbox as
follows;
I do the same thing (with version 1.2.7.1), and it's
Ok i had a loop where i would dial a number and it would ring my extensioni made a change in ;For outgoing calls this setup would dial my extension back.. loop loop exten = _1NXXNXX,1,dial(SIP/1002,60)
exten = _1NXXNXX,2,congestion() exten = _1NXXNXX,102,busy() to this ;For
I've finished a patched version of asttapi that will work with asterisk
1.2. There were fundamental changes to the Asterisk Management
interface between 1.0 and 1.2 that broke asttapi. I think my patched
version will work on 1.0 and 1.2 branches, but I have no way of testing
since I don't have a
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