huu giang schrieb:
Hi Zeeshan
I know a solution using DRBD, Heartbeat and RedFone hardware to
provide failover ability to Asterisk.
If I have two Asterisk Servers, and each server has a TDM card and a
PRI line connect to each card, how your solution can provide failover
ability to
it is definitly worth a try ;)
Kind Regards,
Tobias Wolf
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happened ;-)
Kind regards,
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the opposite ;)
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?
Generally receiving faxes works fine, but sometimes they break and i assume i
might have something to do with echo can.
Kind Regards,
Tobias
Tobias Wolf schrieb:
Olivier schrieb:
2009/2/10 Tobias Wolf tobias.w...@evision.de
mailto:tobias.w...@evision.de
Hello all,
i was just
Hi Kevin,
Kevin P. Fleming schrieb:
Tobias Wolf wrote:
If it is not possible, why is that so ? Is there really no need to do this
and i
am totally mistaken?
This is generally true. Any standards-compliant FAX machine or modem
will generate a CED tone during the beginning of the call
Kevin P. Fleming schrieb:
Tobias Wolf wrote:
Does this only take place if fax detection is enabled in DAHDI or is it
something that happens everytime a CED tone is send over the line?
FAX detection is not done in DAHDI, it's done in chan_dahdi (in
Asterisk). CED detection is done
Kevin P. Fleming schrieb:
Tobias Wolf wrote:
I am absolutly positive that DAHDI was not built with NO_ECHOCAN_DISABLE, since
i have only downloaded the package and done a 'make; make install' without
touching anything of the source code.
By the way, i am using DAHDI-linux 2.1.0.3
fax calls, they break nearly 100%. I am
guessing
that disabling echo can might improve this.
Kind Regards,
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Olivier schrieb:
2009/2/10 Tobias Wolf tobias.w...@evision.de
mailto:tobias.w...@evision.de
Hello all,
i was just made aware on the Bristuff-Mailing list, that it is
possible to
disable echo cancellation per dialplan application.
This comes in very handy
sip.conf:
[81.201.83.14]
host = 81.201.83.14
type = friend
insecure = port,invite
context = voxbone
canreinvite=no
Hope this helps ...
Regards
Tobias Wolf
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Hi,
i am running Asterisk 1.6.0-beta4 and i have some trouble with the
Bridge-Application.
Here is what i want to do:
1) Caller A calls an extension and is connected to an AGI-Script.
2) Doing stuff and originating a second call per Manager Interface
3) Call will be set to an extension with
myself a little bit ...
Tobias Wolf schrieb:
Hi,
i am running Asterisk 1.6.0-beta4 and i have some trouble with the
Bridge-Application.
Here is what i want to do:
1) Caller A calls an extension and is connected to an AGI-Script.
2) Doing stuff and originating a second call per Manager
Alex Balashov schrieb:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
Well, i am not getting the correct meaning of 'defunct', but from the
last part of your suggestion i guess you value Kamailio/OpenSIPS more
than SER.
Are there some hard reasion for this.
I
analog fax machine
With this scenario it would be possible to use the old fax machine you
already have, but also use reliable fax transmission over IP.
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Tel: +49 (0)231
is already a digital stream
of bits).
But ZapRAS can only be used to dial-in with another ISDN Modem on my
side, right. If i have a simple analouge modem there will be no data
connection because of the different protocolls. Is this correct?
Regards,
Tobias Wolf
is
my base number. I make an outgoing call from a phone which sets its
CallerID to 1500.
Can anyone be so kind to tell me what is shown to the callee in either case?
Thank you very much ...
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Hi,
Steve Totaro schrieb:
I would avoid IAX and use SIP if at all possible.
Thanks,
Steve Totaro
Can you some reasons for this? Would interest me a lot why SIP is better
in a large Enviroment. than IAX.
Kind regards,
Tobias
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Hi,
Matt Watson schrieb:
There is really no reason why you cannot.
Personally… I’d avoid using Java for AGI’s that you think are going to
receive heavy use… simply because the JVM adds a lot of overhead, and
possibly a very real performance impact from having the load the JVM
everytime.
this myself the moment i got my password back for the Wiki ;)
Thanks for any help/suggestions offered,
Tobias Wolf
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in answering my questions according
ZapRAS, or are they too trivial ;)
Have i overlooked some useful source of information ?
Tzafrir Cohen wrote:
On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote:
Hi all,
i have found the two possible solution for doing RAS with Asterisk:
1) PPPD
.
This is not very elegant, but cheap and saves one the hazzle with
patching software that runs smoothly or has to be upgraded ...
regards
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Evision GmbH
Wittekindstr. 105
44139 Dortmund
Tel: +49 (0)231 - 47790
Tzafrir Cohen schrieb:
On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote:
Hi all,
i have found the two possible solution for doing RAS with Asterisk:
1) PPPD (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD)
I have downloaded the tgz proposed in the Wiki
provider or Marterna.
Has anybody some hint for me?
If i get it working stable i will update the wiki page ob voip-info
Regards,
Tobias Wolf
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Hi all,
does anyone got this app ported to Asterisk 1.6?
I would greatly appreciate any hints what i have to do, to make it
working ...
Thx ...
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Wittekindstr. 105
44139 Dortmund
Tel: +49 (0
if there was an issue
with callerid events? Or is it possible that i have screwed up my
configuration and since i didn't watched the events for callee channels
didn't recognized.
Kind Regards,
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it in ... I have found the
patch in the Bug Tracker and it works, if the telephone listens to it.
Regards.
--
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Leiter Softwareentwicklung / Kommunikationslösungen
Evision GmbH
Wittekindstr. 105
44139 Dortmund
Tel: +49 (0)231 - 47790 307
Fax: +49 (0)231 - 47790
is
definitly an improvement and sinve the Snom 360 listens to the SIP
message mentioned above i hope this one will too.
But from the data sheet i see, that the M3 also has no means to access a
central adress book. Sad.
Regards
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Tobias Wolf
Leiter Softwareentwicklung / Kommunikationslösungen
straight-forward.
But if you have KIRK equipment you should be able to call their support
to get you started, my test was some time ago, and i don't really know
any more what you have to do exactly to connect to asterisk, but it is
possible ;)
Regards
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Leiter Softwareentwicklung
the phone in all the
telephones and sent new phonebook files to them.
Regards,
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Tobias Wolf
Leiter Softwareentwicklung / Kommunikationslösungen
Evision GmbH
Wittekindstr. 105
44139 Dortmund
Tel: +49 (0)231 - 47790 307
Fax: +49 (0)231 - 47790 500
http://www.evision.de
say that the most accurate
answer is, your mileage may vary, but don't hope for a lot :-)
I dont't think that chan_modem was designed for *analog modems*. See here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Modem+channels
Anyway it seems to be depreceated.
Cheers,
--
Tobias
to some source of information ?
Thanks a lot ...
Tobias Wolf
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Kirk but they are working on an solution without any
information when it will be available.
Does anyone knows an vendor which supports this feature ?
Regards,
Tobias Wolf
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Humberto Figuera schrieb:
HI Tobias,
look in www.soft-switch.org/unicall/unicall/index.html ;p
Thank you. Not very complete but it has given me an idea what to think
of unicall.
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they have done it and with what setup. Don't let us stay in the
rain ;)
Thanks for you help,
Tobias Wolf
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Noc Phibee schrieb:
Hi
i read the list and see a lot of personn say T38 it's not possible
with asterisk and other says that he use T38 with asterisk ??
i don't understand ;=)
Well, if i understand it correctly then Asterisk currently only supports
T.38-Passthrough, which means, you have to
Carlos Chavez schrieb:
On Thu, 2007-02-15 at 16:11 +0100, Tobias Wolf wrote:
Callerid is not defined by the hints. You need the line:
callerid=asreceived
This should be in the definition of your zap channel so it passes the
callerid information without modification to your
Hi,
I use two hint-extensions to monitor my two ISDN-Lines:
exten = 10,hint,Zap/10
exten = 11,hint,Zap/11
My Snom subscribed to the hints and then one line gets busy i have a LED
assigned to the line, that flashes til the call is up and then stay on
til the call is over. So far so good.
If a
Savoy, Kevin - Williston, ND schrieb:
I'll give this a try but seems silly to require 2 different extensions
for one conference room. Thanks for the input.
Actually you don't need 2 different extension, but two different
parameter-sets for the meetme-App. So, you have to implement some logic
with
Dialplan-Logic, or AEL.
I am sure, you will come up with an solution :D
Take care,
Tobias Wolf
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Hi,
Conrad Wood schrieb:
On Wed, 2006-11-15 at 17:50 +0100, Tobias Wolf wrote:
Thx, for you answer ;)
I have some trouble with setting my CallerID if i make an international
Call. No Problems with National Calls, i can set whatever I want. We pay
for this service but our telephone provider
Conrad Wood schrieb:
You might have more luck asking on a telecom specific list, rather than
here.
Well, thx for your time anyway. I just wanted to double-check that there
aren't any asterisk config faults that i were not aware of.
Tobias Wolf
is from Germany to England, maybe it is a
problem between the providers and not of my setup.
Maybe someone could explain to me how CallerID transmission is done on
the technical level I could guess where i have to look for an solution.
Thx.
Tobias Wolf
Raphael Jacquot schrieb:
At some point in my dial plan, I need to find out the length of a sound
file in seconds (to weed out things that are way too short)
the record application doesn't seem to have any facilities to do that.
any ideas ?
i am wondering ... the voicemail app, does
David Gagnon schrieb:
Are you having this problem with an analog line or PRI ?
David
Sorry, forgot to include that information: It's a PRI.
My Asterisk is: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q with Zaptel 1.2.6.
Tobias
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where i don't know what type of phone is to be called it
becomes quite difficult to set an appropriate timeout.
Is there someway to get Dial() to start the countdown, when the channel
state changes to ringing ?
Thx for any advice,
Tobias Wolf
Allan Kamau schrieb:
I would like to run a fast-agi application in the
background.(cmd agi())
This is because I would like to implement a
disconnect after so many seconds feature or at least
a log of the duration of the call.
What about using an Option of the Dial-App instead ??
S(n):
On 7/5/06, David Beckerdite [EMAIL PROTECTED] wrote:
Is there an archive for this list that can be searched? If so, could
someone
tell me where it's located?
http://dir.gmane.org/gmane.comp.telephony.pbx.asterisk.user
works for me :)
Tobias
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Greg Oliver schrieb:
On Thu, 2006-03-16 at 18:39 -0500, Alexander Lopez wrote:
I have offered but I don't think he (owner) id open to that.
This matter was discussed earlier, the problems seems to be in the
difficulties of mirrowing an database-driven wiki.
Tobias
Douglas Garstang schrieb:
Is it just me or is the voip-info web site down right now?
I was experiencing problems accessing voip-info, too. But i guess the
problems derived from accessing http://www.google-analytics.com, because
i could see that voip-info was resolved rather quickly and after
Benny Amorsen schrieb:
MG == Michael George [EMAIL PROTECTED] writes:
MG I may be way behind here, but I see that digium redesigned their
MG site. I cannot find the mailing list search screen.
MG I have found the mailman list page, but that doesn't have have a
MG nice search ability.
Tomislav Parcina schrieb:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I am aware of the possibility to add the option t or T to dial, so #33
transfers the call to extension 33.
It needs to be deined in feautres.conf file. So when you dial #1 you'll
hear transfer and than
Victor Alvarez schrieb:
Hi,
I'm afraid I don't know how to use the command Transfer.
I am also interested how the command Transfer should be used.
I am aware of the possibility to add the option t or T to dial, so #33
transfers the call to extension 33.
Is there any use of this command
' and 'internationalprefix' to
zapata.conf, as proposed in
http://www.asteriskguru.com/tutorials/pri_zaptel.html
but while restarting * tells me that these options are unknown.
Is there any way to access NPI or TON information of an incoming call on
Zap Channels?
Tobias Wolf
Hello,
i've got the following setup:
exten = _X.,1,Dial(Zap/${EXTEN},15,T)
exten = 9000,1,AGI(agi://localhost/myagi.agi);
Now i want to do the follwing. With the catchall extension i make an
outbound call to another person. This Person will then get transfered to
extension 9000 and will be
Hi,
the following szenario leads to a problem:
I connect an CAPI channel to an AGI-Script per Manager API. This Agi
script starts the MeetMe-Application. The Person on the Capi Channel is
now able to speak with the other conferess in the MeetMe-Room. But if
the CAPI channel hangs up, the
Hi there,
i've become curious about chan_unical and found the information site at
the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+MFC+R2 .
i understand that it uses a different architectue as zaptel with libpri
and chan_zap.
can anybody explain briefly the main advantage of
Hi,
Emanuele Pucciarelli schrieb:
Tobias Wolf ha scritto:
I tried that successfully with my own SMS rig a couple of years ago. As
far as I could tell from experimenting and from the ETSI docs, the phone
knows it shouldn't ring, but it should answer and talk FSK to the SMSC,
by looking
sends to * an SMS
with the number 555, if i want to send to this phone it will have to
come from 5550 so that the phone accepts it and does not ring?
Are there any brand specific settings?
I hope that anybody can help me a little bit ;-)
Thx in advance ...
tobias wolf
Armin Schindler schrieb:
Hi,
this should already be fixed in current CVS version and will be part of
next release.
Maybe you want to try it. (Note: capi.conf and dial syntax has changed)
Armin
Yes, thank you. updating to cvs-version did solve the issue :)
tobias wolf
the
same problem? I am using Linux Kernel 2.4.27 and an Fritz!PCI v2.0 Card.
Thx in advance :)
Tobias Wolf
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. if so i would
gladly hear any advice how to produce useful information for debugging
on my side :)
have a nice day
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Hello all,
can anybody how usable app_sms is? I want to use it in england (but not
with the BT) and in Germany. Is this possible with * 1.0.9 and either
with PRI lines or with simple ISDN and an AVM Fritz! Card??
Thx in advance,
Tobias Wolf
hi,
Adeel Ali schrieb:
Jul 18, 2005 2:55:13 AM net.sf.asterisk.util.impl.JavaLoggingLog info
INFO: Received connection.
Jul 18, 2005 2:55:14 AM net.sf.asterisk.util.impl.JavaLoggingLog error
SEVERE: Resource bundle fastagi-mapping is missing.
Jul 18, 2005 2:55:14 AM
statement in the code, does indeed wait.
any thougts ??
maybe someone has an idea how sip+agi+meetme could work ?
thx in advance ...
tobias wolf
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Lee Azzarello schrieb:
The README in the source code states:
app_conference doesn't have DTMF-activated features or anything like
that.
I'm curious how you got audio working on your compliation. I am running
CVS HEAD + app_conference in a Xen virtual machine. I can connect to the
channel but
is muted for me and i
and menu is read to me. Something like the ${MEETME_AGI_BACKGROUND} in
MeetMe.
Thx in advance :)
Tobias Wolf
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Hi,
i have just started to configure access to the * over SIP-Phones.
Therefore I have defined this SIP-Phone in sip.conf:
[tobias]
type=friend
username=tobias
secret=tobias
auth=md5
host=dynamic
reinvite=no
dtmfmode=inband
callerid=Tobias 1087006
allow=all
context=javaAgi
dtmfmode=rfc2833
Hi,
Rich Adamson schrieb:
The s extension matches only when no digits are dialed. Dialing a 1
is a digit, so no match.
Oh, ok, i think now i understood. So the s extension is mainly the
starting point for contexes which i reaches from other contexes, eg.
because of a goto. When I receive a
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