On Tue, 31 Jan 2006, Ezio Vernacotola wrote:
Look at http://www.uptime.it/gsprov/gsprov.htm, I wrote a small python
library that can generate binary config files
sadly this does not appear to work on gxp2000. the gxp2000 uses some
completely incomprehensible encrypted binary format.
-Dan
HiI am facing very strange problem when i try to use asterisk in media proxy mode by using canreinvite=no i receive no voice at both ends. and when i use canreinvite=yes voice is OK at both endpoints. i tried to use play back application to check if asterisk is communicating well with UA and
On 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
The genzaptelconf doesn't work with E1/T1 cards in my experience.
You will have to configure it by hand.
PsulH
ok, thanks,
do you know where can i get the configuration to set in /etc/zaptel.conf?
1. T1 or E1
2. Line Encoding
On 02/10/06 09:55 kevin ling said the following:
Hi,
You need the unattended transfer (blind transfer) featuer. That implemented
in Asterisk (#) button. Not attended transfer.
right, but adding in this behaviour into attended transfer would allow us
to then retire blind transfer.
--
Sorry, that's correct - so when experimenting with s/w echo can try the different options.RobOn 2/11/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I thought that if the VPM was detected then you didn't have any control as to which algorithm was used.I was under the impression that the algorithms
Why don't you think it is correct
behaviour? The purpose of attended transfer is that you consult with the party before transferring with hooking, otherwise it would be a blind transfer for which there is a blind transfer option.Rob
On 2/10/06, Moises Silva [EMAIL PROTECTED] wrote:
this is a
Nik,
Start here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels
It will give you some background info. Also, be sure to learn the
difference between zaptel.conf and Zapata.conf. It took me two weeks to
realize what each one does:
Zaptel.conf handles the lower-level stuff
This afternoon I finally figured out more with regarding to a strange
clock-slip problem we have on our asterisk box.
We have two TE410s, in E1 mode:
TE410P version c01a009b
They have their own interrupts:
66: 781648298 783747388 IO-APIC-level t4xxp
233: 253890977 1311504670
Sorry for re-posting this message -
Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor.
Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency.
i.e a sentence which should finish in 4 secs finishes in much
On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote:
Sorry for re-posting this message -
I am trying to run the latest stable Asterix version 1.2.4. on 64 bit
amd procesor.
Things are working but the playback sounds that I hear when tring to
connect over IAX are of very high frequency.
i.e a
Dear All,
I've got a weird problem with my asterisk box which
has fxo interfaces (TDM400). Well, the problem is that
the interface answers the call, but no caller id is
being received. Also, sometimes this error happens:
fsk_serie made mylen 0
Any idea what is going on?
Thanks,
Kaveh
sorry can you elaborate a little, what exactly is timing issue?
Thanks
On 2/12/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote: Sorry for re-posting this message - Iam trying to run the latest stable Asterix version
1.2.4. on 64 bit amd procesor. Things
On Sun, 12 Feb 2006, Florian Heer wrote:
Hi!
I am playing around with Asterisk and have a problem :-)
(Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4)
I have a sip-phone at my desk and an ISDN-phone (independent of the
Asterisk-server) in my living room, when I'm not at my desk, the
Latest firmwares of bt100 and gxp2000 have an option to authenticate the
config file. I haven't looked yet at it and don't know what is needed to
make my library compatible. I am using some gxp2000 with Authenticate
Conf File: No and the generated binary config works.
Ezio
[EMAIL PROTECTED]
On Sun, 12 Feb 2006, Ezio Vernacotola wrote:
Latest firmwares of bt100 and gxp2000 have an option to authenticate the
config file. I haven't looked yet at it and don't know what is needed to make
my library compatible. I am using some gxp2000 with Authenticate Conf File:
No and the generated
On 2/12/06, Michael Collins [EMAIL PROTECTED] wrote:
Nik,
Start here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels
It will give you some background info. Also, be sure to learn the
difference between zaptel.conf and Zapata.conf. It took me two weeks to
realize what
for example, now i've set this in /etc/zaptel.conf
span=1,0,0,ccs,hdb3,crc4,yellow
span=2,0,0,ccs,hdb3,crc4,yellow
bchan = 1-15, 17-31
dchan = 16
bchan = 32-46,48-62
dchan = 47
loadzone= it
defaultzone = it
if i try
[EMAIL PROTECTED] ~]# ztcfg -vvv
Zaptel Configuration
ok guys, it seems that i've found the solutions
i had to switch a jump on the card, as is explained here
http://www.asteriskguru.com/tutorials/wildcard_te405p_wildcard_te410p.html
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On Sunday 12 Feb 2006 01:04, Kevin P. Fleming wrote:
[...]
(and by 'you' I don't mean you specifically, Michael, I am referring to
those in this thread who think we should change the code used to write
to every file we write to in Asterisk)
I assume you are referring to me. I did not say that.
On Sunday 12 Feb 2006 10:57, Bob Goddard wrote:
On Sunday 12 Feb 2006 01:04, Kevin P. Fleming wrote:
[...]
(and by 'you' I don't mean you specifically, Michael, I am referring to
those in this thread who think we should change the code used to write
to every file we write to in Asterisk)
Cosmin Prund wrote:
I've got a Mobile-to-PBX gateway installed and I want the ability to dial
from my mobile phone into my PBX and next dial a land-line from the PBX so I
can make cheep mobile-to-land-line calls while on the go.
I've contemplated using the WaitExten application but it only
Matt Schulte wrote:
Could anyone either recommend a website or howto on optimizing Linux to
run asterisk. Such examples of what I mean are..
Renice of asterisk pid's
Forcing irq smp_affinity (For interupt hogging T1 cards)
.. That kind of stuff, I looked on the wiki and nothing directly
well if I pass the parameter aswhennext/4 instead of whennext/8 in file.c = ast_readaudio_callback()= ast_sched_add(.,whennext/4,...)
things start working fine.
I debugged the sched.c and time.c didn't find why this should happen.
Since the scheduler calculates time-interval and keeps
got the answer the gettimeofday() is twice as fast as the one in older box, problem with system clock.
Nitin
On 2/12/06, Nitin Gupta [EMAIL PROTECTED] wrote:
well if I pass the parameter aswhennext/4 instead of whennext/8 in file.c = ast_readaudio_callback()=
I agree that systems should be well-adminstered, I also prefer
programs that don't run amok even when there are lapses in
administration.
Since detecting which file caused the SIGFSZ is impractical, how about
if we do this.
a) don't rotate logs on SIGFSZ if it's done it recently.
b) when it
Hi,
I tried to connect SPA 3000 to bell phone line but it didn't work as it
should. Incoming calls worked but outgoing didn't. I need help on its
configuration. Couldn't find useful information on the Internet so far.
Zach A.
___
--Bandwidth and
Ok,
I don't know what's up with this mailing list.. But I swear it's
broken. Mail doesn't seem to get out to the list for hours, and this
message never came through.
I'll post again.
Hi,
I'm currently running CVS-HEAD 2005-09-03
I do plan to upgrade to the newest version, but need to do some
Hi!
I am new to asterisk and I'd like to know wheter the following scenario
is possible:
Someone press the Button on the door station.
The door station dials lets say the extension 333.
I take the call on 333 and talk with the person on the door.
Now I'd like to activate the door opener by
Tamas wrote:
Matt Schulte wrote:
Could anyone either recommend a website or howto on optimizing Linux to
run asterisk. Such examples of what I mean are..
Renice of asterisk pid's
Forcing irq smp_affinity (For interupt hogging T1 cards)
.. That kind of stuff, I looked on the wiki and
On Sun, Feb 12, 2006 at 12:52:29PM +0100, Tamas wrote:
we have exactly the same questions and weak points in our system.
What I found is that it's not every time good to give highest nice value
[-20] to asterisk because the system can be unaccessible if asterisk
process starts eating CPU (due
Probably because this isn't the way that a lot of other PBX systems
work. It's not always easy to educate users about the difference
between a blind and attended transfer when the systems that they've used
in the past don't make this distinction.
Disconnecting the outside caller certainly
The following is my dialplan for outgoing international call. What I want are:
- when people dial 9011604xxx , 9011605xxx, 90114411xxx,
90114421xxx, use voipstunt to dial out
- otherwise, use my pstn to dial out.
What I've found is when i dial 9011604xxx , 9011605xxx,
Hi
I am experimenting Asterisk , so far I am
able to talk from two sip clients under one server and
in the same network, [ Thanks to the mailing list ]
Now I want to have two or more Asterisk
server and SIP clients from one server communicating
to the other sip clients in
On Sun, 2006-02-12 at 10:05 -0500, Wooi Koay wrote:
The following is my dialplan for outgoing international call. What I want
are:
- when people dial 9011604xxx , 9011605xxx, 90114411xxx,
90114421xxx, use voipstunt to dial out
- otherwise, use my pstn to dial out.
What
Warren Burstein wrote:
b) when it does rotate files on SIGFSZ,. it should rotate the csv file, too,
and
any other files that are written to (maybe only of they are larger than the
file
size limit)
And there's the important point: any other files that are written to.
Asterisk writes to
Hello everybody,I have set
dtmfmode=auto in my sip.conf, but that does not work and I still got the
following message:
WARNING[4980]: dsp.c:1422 ast_dsp_process:
Inband DTMF is not supported on codec g729. Use RFC2833
According to
Hello,
I have an IAX2 trunk like this running well with IAX2 and SIP users mixed at
each side.
Runing like a charm :-)
Don't forget to add username definition from this example.
To avoid too much load for your CPUs with transcoding, tempt to have only
the same CODEC choice for all phones and
Hi!
Armin Schindler wrote:
there is a bug, I would need a full log (set verbose 5 ; capi debug) to
find out.
Of course you would, I just didn't know if it was one.
But: if there is a call signaled, the switch has a timeout (about 4 or 5
seconds), this timeout can be extended by sending
Do you have the following set in your zapata.conf?
callerid=asreceived
- Original Message -
From: KaveH Aasaraai [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, February 12, 2006 10:25 AM
Subject: [Asterisk-Users] Zap, Caller ID problem
Dear All,
I've got a
Aryanto Rachmad wrote:
Why does it not work as the wiki said?
It does work exactly as the wiki said. The SIP peer did not offer to
send/receive RFC2833, so we assume it wants to use inband DTMF. However,
inband DTMF _does not work_ over G.729 codec, so you get a warning.
You need to enable the Exchange SMTP gateway to receive email from your
* server's IP address. There is a way to do that if you look closely in
the Advanced section of the SMTP connector. It's under relaying.
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)
Thanks a lot Kevin,
I am aware that inband DTMF does not work over G.729 codec. So in this case my
provider does not offer RFC2833. I can not do anything about this, can I? Or is
there anyway to simulate inband DTMF over any other codecs, but G.711a or G711u?
Cheers,
Anto
- Original
On 1/23/06, Ira [EMAIL PROTECTED] wrote:
At 05:06 AM 01/23/2006, you wrote:
http://asteriskdocs.org deserves all mentions it receives and the
Though you really should mention that it's a 1.0 document and trying
to make a 1.2 installation work using that book is somewhat futile.
That's
I tried to connect SPA 3000 to bell phone line but it didn't work as it
should. Incoming calls worked but outgoing didn't. I need help on its
configuration. Couldn't find useful information on the Internet so far.
Check www.voxilla.com and look for a 'wizard' to help configure the box.
Lots
Aryanto Rachmad wrote:
I am aware that inband DTMF does not work over G.729 codec. So in this case
my provider does not offer RFC2833. I can not do anything about this, can I?
Or is there anyway to simulate inband DTMF over any other codecs, but G.711a
or G711u?
You are correct, if your
On Sun, 12 Feb 2006, Florian Heer wrote:
Armin Schindler wrote:
But: if there is a call signaled, the switch has a timeout (about 4 or 5
seconds), this timeout can be extended by sending ALERT (Ringing).
Okay, is the timeout necessary? Or: is this short timeout necessary? It
appears
Hi
iam testing my asterisk server
some time i get one way voice
some time i get voice breaking
i dont see any bandwidht throuttle
and i see the RAM is low
is this cause will drop the Voice quality ??
ram
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Title: IP phone with many speed dial buttons
Hello,
Im looking for IP phones with at least 10 or so speed dial buttons. Can you recommend something which works with Asterisk
and does not cost fortune?
An option can be analog phone combined with ATA adapter. So hints for good analog
You all may like this http://mundy.org/blog/index.php?p=82RobOn 2/12/06, Aryanto Rachmad
[EMAIL PROTECTED] wrote:Do you have the following set in your
zapata.conf?callerid=asreceived- Original Message -From: KaveH Aasaraai [EMAIL PROTECTED]To:
asterisk-users@lists.digium.comSent:
SNOM-320 has 12 - see www.snom.com - the SNOM-360 can take a key pad with another 42.RobOn 2/12/06, David Hajek
[EMAIL PROTECTED] wrote:
Hello,
I'm looking for IP phones with at least 10 or so speed dial buttons. Can you recommend something which works with Asterisk
and does not cost
On 13:51, Sun 12 Feb 06, David Hajek wrote:
Hello,
I'm looking for IP phones with at least 10 or so speed dial buttons. Can
you recommend something which works with Asterisk
and does not cost fortune?
Have a look at the Snom 320.
If you need more go for the 360, it can take a sidepad with
Hi,
does anybody know if following is possible:
Asterisk-- Softphone -- Bluetooth cell phone
I.e. is it possible to programm a softphone to forward calls from/to a
regular cell phone via bluetooth? Evidently the cell phone would need
to be programmed accordingly.
In my opinion the most
At 06:38 AM 02/12/2006, you wrote:
Can anyone give me some hints where to start looking in the docu?!
I only need to know how to execute a script when I press - lets say the
* Button while i am talking.
Look at features.conf and also try searching for flash() on the
wiki. I can flash a call
Hello Armin,
Armin Schindler wrote:
On Sun, 12 Feb 2006, Florian Heer wrote:
Armin Schindler wrote:
But: if there is a call signaled, the switch has a timeout (about 4 or 5
seconds), this timeout can be extended by sending ALERT (Ringing).
Okay, is the timeout necessary? Or:
I am running * 1.0.9/E1 PRI line with telmex/16 FXS phone clients and
after succesfull running for almost 4 hours, suddenly, SIP phones loss
registration and begin to reregister as if they were rebooted. No
Unregister messages logged, nor level 2 link lost. Calls regarding this
UACs, of course,
At 08:44 AM 02/12/2006, you wrote:
That's incorrect as the book was written for 1.2. Old documentation
was written for 1.0 (as there was no 1.2 work going on at that time).
That might be so, but that would indicate to me that 1.2 changed
quite a bit between when the book was written and when
At 12:57 AM 02/12/2006, you wrote:
Why don't you think it is correct behaviour? The purpose of attended
transfer is that you consult with the party before transferring with
hooking, otherwise it would be a blind transfer for which there is a
blind transfer option.
So let's consider an
Hello,I have a few astersk servers, talking via SIP to an upstream provider. I decided to launch a callshop using two Atcom AG168V ATAs, each talking to a central asterisk server via IAX. PhoneATA -(iax)---asterisk(sip)-upstream
I am using a billing system , that sits
I am running * 1.0.9/E1 PRI line with telmex/16 FXS phone clients and
after succesfull running for almost 4 hours, suddenly, SIP phones loss
registration and begin to reregister as if they were rebooted. No
Unregister messages logged, nor level 2 link lost. Calls regarding this
UACs, of course,
That certainly is the way it SHOULD work. Blind and attended transfer
should be able to be initiated the same way. It certainly is the most
efficient logical way. Attended transfer should revert to blind simply
by the initiating party hanging up.
Most legacy hybrid key/pbx systems work that
Since your EC only needs to support a tail long enough to handle the
PSTN part of the call, I suspect even fairly short tails are fine.
Steve Underwood wrote:
I don't know about the Tellabs cancellers in particular, but I think any
echo canceller built in the 80s will be a fairly poor
Does anyone know if it is possible to upload a common directory to all
Aastra phones (480i, 9133)? Is there someting equivalent to the way Polycom
phones do it?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
Did any one succeed in using Rhinobell with asterisk? If so, could you
please share setup instructions? Thanks.
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
[EMAIL PROTECTED] is believed to have said:
Do you have the following set in your zapata.conf?
callerid=asreceived
Dear all,
I add my half cent on the subject.
I do have the following zapata.conf:
*
[channels]
usecallerid = yes
signalling = fxo_ks
callerid = A 2302
context =
What version of Asterisk and Zaptel you were using? Did
you try latest Asterisk 1.2.4 and Zaptel 1.2.3? Anyone has good feedback for
TE411P?
Isaac Xiao
Stagg Shelton wrote: It was Digium's opinion that perhaps the card had a VPM. We got a replacement TE411P, I implemented it tonight
I've seen a similar problem before. Span 3 was throwing errors for
(what seemed to be) no reason at all. After some testing it seemed
that the number of errors thrown on Span 3 had a relationship to the
temperature inside the servers. After installing additional cooling
the errors had
Nope only bad feedback here. The software EC in Asterisk worked much better for me than did the VPM on the TE411P.On 2/13/06, Isaac Xiao (KVB Kunlun Pty Limited)
[EMAIL PROTECTED] wrote:
What version of Asterisk and Zaptel you were using? Did
you try latest Asterisk 1.2.4 and Zaptel
I am using asterisk 1.2.4 and zaptel 1.2.3. Also, I tried the latest
zaptel out of subversion.
Stagg Shelton
www.oneringnetworks.com
Isaac Xiao (KVB Kunlun Pty Limited) wrote:
What version
of Asterisk and Zaptel you were using? Did
you try latest Asterisk 1.2.4 and Zaptel
I'm
using the Varionboards with no problem.
Now,
about echo...
Sagnoma says if YOU have echo, it is THEIR problem and they will fix
it.
James TaylorMetroTel3505 Summerhill RoadSuite
11Texarkana, Tx 75503903-793-1956
-Original Message-From:
[EMAIL PROTECTED]
Case
sensitivity? The CLI references Goodbye but your filename is goodbye.gsm.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam Lee
Sent: Friday, 10 February 2006
1:22 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Hi Gerard --
I found that I get the really loud buzzing sound in the handset earpiece
when I set echocancel=256 instead of echocancel=yes (the default = 128
taps).
It seemed to occur irrespective of the actual echo canceller chosen.
Mike.
-Original Message-
From: [EMAIL PROTECTED]
Hi
That's a known problem with 1.2.2. Upgrade to 1.2.4.
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kamran Ahmad
Sent: Saturday, 11 February 2006 9:09 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] No Voice when
The problem is with my new machine's hardware clock, which was running twice as fast as normal(bios bug). After searching through google the fix for it I could find is to disable apic in bios.
This is my machine configuration:
HP a1230nATI XPress chipsetAMD Athlon 64 X2 3700+ATI X800 XLLinux
Hi Rob
Is it possible
to disable the onboard echo canceller so that one may try the software
cancellers instead?
I have the
TE110P and am experiencing the same bad echo problems that I cant seem
to effect by fiddling with the echo canceller settings in zconfig.h
Cheers,
Mike.
Hi.
I have some great door intercoms for sales. You can find them on our
website. It comes with door opener and a door code to activate the door
relay (built-in relay). Any user at the asterisk end can dial a code to
also open the door or talk to the person at the door before opening it.
John is absolutely correct - in the PBX world a transfer is a transfer,
regardless of whether it is blind or attended. How many PBX phones out
there have two different transfer buttons, one for blind and one for
attended? Zilch.
It's the user's behavior that determines whether or not the
Questions for the community: is an integrated transfer feature
valuable to you?
Yes, merging blind and attended transfer would be valuable for me!
If so, would you be willing to put out a bounty?
Maybe. Depends on how much it would be.
Tom
On Mon February 13 2006 02:15, [EMAIL PROTECTED] wrote:
Hello,
I have an IAX2 trunk like this running well with IAX2 and SIP users
mixed at each side.
Runing like a charm :-)
Don't forget to add username definition from this example.
To avoid too much load for your CPUs with transcoding,
At 01:49 PM 02/12/2006, you wrote:
Does anyone know if it is possible to upload a common directory to all
Aastra phones (480i, 9133)? Is there someting equivalent to the way Polycom
phones do it?
If there is, it's in the recently released XML documentation which
you can find in the
At 04:58 PM 02/12/2006, you wrote:
Questions for the community: is an integrated transfer feature
valuable to you? If so, would you be willing to put out a bounty? (In
other words, is it just a nice feature or is it so important that you'd
be willing to pay a few bucks for it...)
I'm
Hello Florian,
I spoke to soon, thought you were referencing something else... I have
been having a problem post 8015 build of asterisk that has been
preventing me from going up any higher...
It's an odd one too, and I narrowed it down, tested like crazy, etc...
You could see my previous post
Carlos,
I'm planning to use the Aastra 9133i for a new installaton.
Can u please comment on your experiences with this equipment. Please let me know if u have found any specific issues with it.
thanks in advance.
On 13/02/06, Ira [EMAIL PROTECTED] wrote:
At 01:49 PM 02/12/2006, you wrote:Does
For the Aastra IP Phones I think you can specify up to 2 common delimited
directory files to be downloaded to all the phones through their aastra.cfg
file.
Its done using the directory 1 and directory 2 parameters in the cfg
files. You should also be able to download a copy of the phone's
Hello All,
I am trying to figure out which way to go for a quad port fxo solution
with a good echo can on it. My options are the sangoma remora, a
mediatrix fxo, or something similar.
The issue is that I would need a good EC. This would be on about a 9000
foot loop, and the lines don't
Maybe do a transfer to a dedicated extension, which calls the script
with the system() command to open the door? Or use the feature keys for
a blind transfer. Seems like it could work.
Btw, what kind of door phone opener do you have? I've been looking for
something similar...
Greg
On Sunday, February 12, 2006 9:36 PM John Novack wrote:
That certainly is the way it SHOULD work. Blind and attended transfer
should be able to be initiated the same way.
...
I would consider that a defect or bug, not a new feature request.
I second that. Regarding the bounty: Once
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