Have you tried without reinviting ?? (canreinvite=no)
Is your * box behind a nat ?
maxx
Scott Pinhorne ha scritto:
Hi All
I have setup my asteriks to use voipcheap.com for the outgoing trunk
on local calls (because they are free), my setup is below:
register = username:[EMAIL PROTECTED]
Koopmann, Jan-Peter ha scritto:
On Thursday, January 19, 2006 6:53 PM Massimo De Nadal wrote:
If you are using bristuff add the m option like this:
Dial(Zap/g1m/numbertodial)
This will turn off the modulation (rxgain/txgain) and should (!) turn of
the echo cancellation but I am not sure
Anybody knows if it's possible to disable zap echo cancellor from
dialplan only for certain outbound calls ??
I share the same phone lines for voice calls and faxes. Iaxmodem works
fine for me only turning off the echo cancellor, but I need it for
voice calls.
Any ideas ?
maxx
Guillermo Salas M ha scritto:
Hi all, I was checking the TDM2400 features and seems to me very
interesating. I think is that I need :)
I want to know your experience with this card and if you know abouts
bugs, configuration and everithing thah I need to know before acquire
it :)
The
Hi Guys,
I'm trying to send and receive some faxes using iaxmodem and hylafax
through an hfc isdn board and a bristuffed asterisk.
All seems to work fine, but the faxes are sent randomly truncated
without any reported error.
Any idea or suggestion ? Anybody owns a working
Giovanni Miano wrote:
I've 2 hfc billion and one TDM400P 1fxs/1fxo with bristuff 0.2.0-RC8o
and * 1.0.9
I dont recive callerid from TDM400P fxo port but isdn hasnt problems
If i try to use only TDM400P 1fxs/1fxo without bristuff.. all work ok
is it bug of bristuff ?
Maybe, why not try
Joseph ha scritto:
On Mon, 2005-09-12 at 09:56 +0200, Domjan Attila wrote:
On Mon, 2005-09-12 at 08:51 +0200, Dave Cotton wrote:
On Sun, 2005-09-11 at 20:58 -0600, Joseph wrote:
Does anybody runs Asterisk on AMD64?
I can compile it on Gentoo, and start Asterisk a command line
Alessio Focardi ha scritto:
Hello Lars,
Have you got kernel sources installed ?
I think that are mandatory for Zaphfc.
Regards
Not only, you have to have the kernel config save file too.
Remember to make dep too.
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Lars Dybdahl wrote:
I would like to know how to install asterisk 1.0.9 with zaphfc working
on a SuSE 9.2.
Any ideas?
Forget RPM.
First of all read:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+SuSE
then download
http://www.junghanns.net/downloads/bristuff-0.2.0-RC8n.tar.gz
.
Is it possible to port zap echo cancelor to different channels like
chan_capi ?
Armin Schindler ha scritto:
On Thu, 23 Jun 2005, Massimo De Nadal wrote:
Have you planned to integrate some echo cancel feature ?
Echo cancelling (if the card supports it) is already implemented.
As far
Sergio Chersovani wrote:
As far as I know the Eicon Diva Server cards are the only cards
supporting
echo cancel via onboard DSPs.
AVM active cards do not support it?
No.
Avm active cards are basically multi fritz boards running the same
firmware onboard instead of charging pc cpu.
Armin Schindler ha scritto:
Which boards don't support that? If DSPs on board, echo-cancel should be
available.
I have in my hands right now a DIVA Server BRI-2M-PCI (not the 2.0
version) which own its dsp but doesn't echo cancel, due to old capi
drivers which don't support this feature.
Change the dialplan in your spa3k with something like:
(xx.|*x.|**x.)
This way you can dial any number, even starting with * or **
Martin Roy ha scritto:
How can I dial *67 on a Sipura 3000 if I dial from a SIP phone
connected on an asterisk server. I always get a message saying that
Have you planned to integrate some echo cancel feature ?
Armin Schindler ha scritto:
Hi all,
I would like to announce the first release of the chan_capi
channel driver on sourceforge.net
The package is available for download with name
chan_capi-cm-0.5
and is the current CVS HEAD.
It is
every day I discover something new about chan_capi and zaphfc.
That's really exiting but, is there a place with some docs about
these fine pieces of sw without asking for detail every time ??
Klaus-Peter Junghanns wrote:
Hi,
zapata.conf:
prindication = passthrough
best regards
Klaus
Anybody knows what is and how to use the ISDNguard daemon included in
new bristuff packages ?
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Hi,
I'm using a SPA 3000 as FXO and FXS termination connected to my * box.
I'm using the caller ID prefix trick explained here:
http://www.voip-info.org/wiki-Sipura+3000.
All seems to work really fine, there is only a problem when I hangup my
IP phone after a conversation and the other party
Anybody knows how to patch the music on hold bug on a
bristuffed-0.2.0-RC7j 1.0.6-asterisk ?
Thanks
maxx
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Remco Barende wrote:
Hi list!
I'm still trying to figure out about the groups in asterisk.
If I understand correctly, if you assign a certain group number and
you assign the same call group number to a sip device the device will
reing even though you did not specifically specify it in
Remco Barende ha scritto:
Hi list!
I'm still trying to figure out about the groups in asterisk.
If I understand correctly, if you assign a certain group number and
you assign the same call group number to a sip device the device will
reing even though you did not specifically specify it in
Simply dial a # after the number.
Remco Barende ha scritto:
When I use an analog phone connected to a Sipura SPA-2000 it takes
about 3-4 seconds before the number is actually dialled.
Very annoying especially if you are connecting an intercom to it.
Can I change this behaviour and do I need to
Milos Kocbek wrote:
I have a problem trying to install two avm fritz cards on one asterisk
machine. I am using fcusb2 driver. 1 card works perfectly.
The multiple fritz hack works only with pci cards (and, of course, it's
a hack). Avm decided to not allow multiple installation for fritz
Marco Parmeggiani wrote:
I've downloaded and compiled zaphfc and libpri.
To do that i've downloaded bri-stuff and commented out the asterisk related
stuff because i've installed it from a debian package.
Does this means that i have to rebuild the whole asterisk thing to support
zaphfc?
thanks
I've a problem connecting uniVoice (http://voice.uni.it) from asterisk.
Using my account data I can place a call smoothly using xlite or my
budgetone phone directly, but I'm not able to use uniVoice as a peer
from asterisk.
Registration seems to work correctly, but when I try do dial, the sip
Definitely choose chan_capi.
Chan_modem is almost deprecated, bad quality and very few features.
Chan_misdn seems to be a very good project but it is still young.
Zaphfc in theory it's wonderful (zap echo cancellation, timing etc.) but
you have to use older * versions, (till new kapejod's
Dave Cotton wrote:
On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote:
I've just used chan_capi it's very easy to use with Fritz!Cards and
therefore I like it ;-)
Worked straight out of the box on an AVM C2, hope it does the same with
2 Fritz!Cards in the same machine.
Sadly
only avm active cards permit multiple installation straight forward (b1,
c2 and c4)
with fritz! pci you can do the hack mentioned above, for fritz! usb you
can't install more then one.
This limitiation is due to avm drivers design, they choose to allow
multiple installation only on hi-end
:[EMAIL PROTECTED] De la part de Massimo De
Nadal
Envoyé : mardi 19 octobre 2004 16:24
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ???
Sadly no.
If you want to use 2 fritz! in the same box you have to do a little hack
with the drivers
Try deleting the line
pritrustusercid=yes
in zapata.conf
maxx
- Original Message -
From: Bastian Schern [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 20, 2004 8:34 PM
Subject: [Asterisk-Users] Incoming MSN via ZapHFC - to SIP
Hi there,
I've got a small problem with
I've asked Grandstream tech support about attended transfer.
They told me that in about a month there will be available a firmware
upgrade that supports attended transfer natively.
maxx
Chris Shaw wrote:
I know this must have been asked before, but I was just wondering, the
manual says it can
You cannot compile zaphfc with latest CVS head. You have to donwload
specific date version using the download.sh included script.
BTW I have some problems with RC4. It works fine with my 2 isdn pci boards,
but it seems to be unable to drive my TDM400 ...
Try RC3, at the moment seems to be more
forget the asterisk source you have downloaded.
Zaphfc is not only a driver, it's a patch that have to be applied to
specific source version too,
You have to run the install.sh script that is included in the tarball.
This script before downloads the right asterisk version (download.sh) and
then
Thorsten Huber wrote:
Hi,
On Sun, Aug 01, 2004 at 09:13:52PM +0200, Massimo De Nadal wrote:
...
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,Dial(Sip/cisco1Sip/xlite1,30,tTr)
exten=s,3,HangUp
The problem is that when I receive a call, I can't see the CallerID
neither
on the Cisco
Hi,
On Mon, Aug 02, 2004 at 01:20:32PM +0200, Massimo De Nadal wrote:
we had similar problems and fixed them by setting the CIDName to the
CallerID:
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,SetCIDName(${CALLERID})
exten=s,3,Dial(Sip/cisco1Sip/xlite1,30,tTr)
exten=s,4
Hi,
On Mon, Aug 02, 2004 at 01:20:32PM +0200, Massimo De Nadal wrote:
we had similar problems and fixed them by setting the CIDName to the
CallerID:
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,SetCIDName(${CALLERID})
exten=s,3,Dial(Sip/cisco1Sip/xlite1,30,tTr
I'm not sure that this problem is strictly related to zaphfc, but this is
what happens:
my asterisk (build on bri-stuff-0.1.0-RC2k) handles a single PCI HFC-S based
card.
I own a Cisco 7940 Sip phone (fw 7.1) and a pc running X-Lite.
Zaptel.conf and zapata.conf are taken directly from zaphfc
Is it possible to build a dialplan in which shifting from daytime to
nightime is not hour based but phone driven ???
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Ok guys...
the example is simply perfect ! Thanks a lot and shame on me for not reading
carefully the wiki :-)
maxx
On Thu, 22 Jul 2004 14:54:29 +0100, Steve Hanselman
[EMAIL PROTECTED] wrote:
Yes, you'd have a dialplan entry that set a value in the database, then
acted upon that.
I'm using a Cologne chip card in my Asterisk box with zapHFC drivers
(bristuff-0.0.2). The system works well, but this way I'm not able to run
newer version of Asterisk.
Do you think it's better to use i4l support and newer version of Asterisk or
keep the bristuff with older asterisk ??
Have
going to i4l means... incoming sound sometimes gets interpreted as DTMF
- and when your caller humms a '#' - transfer kicks in... Outgoing DTMF
mhhh almost unuseful but surely funny ;-)
There is an Update patch for bristuff... look carefully in the download
directory.
do you mean
Hi everybody,
I have a problem using zaphfc. When I start asterisk after 8-10 seconds I
get the message Primary D-Channel on span 1 down and my isdn modem stops
to work.
If I place or receive a call before this message all works really fine (even
if the call is very long), but when I hang up,
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