Displaying 19 results from an estimated 19 matches similar to: "SkypeForAsterisk"
2009 Dec 29
1
SkyHost is set to expire
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Hi all,<br>
I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last
version).<br>
Everything was going fine, but yesterday I've got this messege when
I've tried to restart asterisk
2003 May 22
1
Experimental Design
I don't know if this is the best place to post this question but I will
try anyway. I have two experiements for which I use one-way
matched-randomized ANOVA for the analysis and I would like to compare
different treatments in the two experiments. The only common group in
the two experiments are the controls. Is there any ANOVA design that
allows me to make this comparison taking into
2008 Feb 18
2
Failure of Sending Voicemail As an attachment in E-mail
Hello all,
I am struggling with sending voicemail as an attachement in Email.
When i have given the email like someone at gmail.com it is delivering
to my gamil account perfectly(of course to spam folder).
But when i given the email like someone at mycompanymail.com it is not
delivering to my company email account..
What should i do ?
Actually my company is using a third party email server..
2008 May 05
3
MeetMeAdmin() working problem
Hello users,
I have been working with a conference setup.
My setup includes:
1)There will be an interface number provided to the user
which might be a DID number or A Toll free number
When user calls the number it asks for the conference room number
and the user pin .
on successfull authentication he will be participated in the conference
2)by didaling the same DID number the
2009 Nov 30
3
Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.2.37,
1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases have been created in response to a SIP remote crash
vulnerability.
Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an
SDP regression
2009 Nov 30
3
Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.2.37,
1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases have been created in response to a SIP remote crash
vulnerability.
Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an
SDP regression
2007 Dec 05
2
Text-To-Speech synthesizer--help required
Hello users,
Actually i wanted to implement Text-To-Speech engine
from cepstral voice using swift application
i tried the documentation of doing this and i was unsuccessful
at doing this work with asterisk
can anybody please help me out finding the solution to installation
thanks in advacnce
srinivas Antarvedi
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2004 Jan 15
3
ISDN CAPI and anonymous callers
Hello,
I am trying to use * to handle anonymous ISDN callers.
Something like
exten => 5150/0,1,Congestion
should work but doesn't. Apparently because the ISDN CAPI doesn't
use 0 for callers who don't send their number.
Is there a way to make * identify ISDN callers who use CLIR?
-Walter
--
Walter Doerr =*= wd@infodn.rmi.de =*= FAX: +49 2421 962001
2009 Dec 30
2
Skype for Asterisk
Hi Sir,
We have integrated Skype with Asterisk (skype user id:-
rexesbposolutions). Each call which is coming to skype account is
getting transfered to Asterisk Queue. It has following two cases:
case 1: When we call from normal skype account to skype account
(rexesbposolutions), everything is working fine.
case 2: This skype account (rexesbposolutions) has been assigned with a
online virtual
2004 Jan 20
4
CAPI: Early-B3 working with AVM-B1?
Hi,
I tested the capi_chan with latest cvs of * and I have problems with
Early-B3. The following dialstring works for me (without Early B3):
exten => _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30)
But if I add the 'b' for using Early-B3
exten => _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30)
nothing changes (no dialtone). If in this example the called party
discards the call, there is no
2004 Jan 05
1
CLIR and isdn4linux
hi
I have a passive isdn port configured in modem.conf
in extention.conf i use this two channel (ttyI0 and ttyI1) with the string:
exten => _NXXXXXXXXX,1,Dial,Modem/g1:${EXTEN}|60|r
how can i hide my msn?
is it possible to activate the clir with the @ before the ${EXTEN}?
thanks
Cristian
2009 Jul 22
3
CallerPres SIP headers Analog Phone
hello all...I have been trying to get a handle on CallerPres with an
analog handset. I have usecallingpres=yes in my chan_dahdi.conf file
and when I dial *67 on my analog handset I see Disabling Caller*ID on
DAHDI/4-1 but when the call is then forwarded to my outbound SIP
provider the RPID header is not correct privacy=off;screen=no instead
of full and yes how can I correct this?
2005 Aug 29
3
How to use * and # as part of number indialcommand
Michel
Send me the same output for a dial string that only sends the *31*
Is this an ISDN line? What type of card/signalling/switchtype are you
using?
It looks as if the PSTN switch accepts the *31* and then hangs up so you
can make the NEXT call with the *31* feature enabled. If so I assume the
*31* feature will be enabled for the next call on the ENTIRE SPAN if it
is an ISDN trunk group.
If
2004 Jan 22
1
chan_capi: suppress calling number on outbound dialing?
Hi,
I just wonder, if it is possible, to suppress my own number on outbound
dials with chan_capi. I took a look into the sources and think it might
work with toggeling the "@" in front of the outbound msn in the
dialstring. (Dial(CAPI/@msn... vs. Dial(CAPI/msn...
But it doesn't work. Maybee I'm wrong and misunderstood the code.
Thanks for any answers!
Karsten
2007 Feb 28
1
voicemail advanced options problem with mysql datbase
Hello all
i have an asterisk setup integrated with mysql via odbc driver
myproblem is:
when i reading my voicemails, in advanced options the following functions
not working with realtime asterisk but working with flat files.
1. Reply to the message(option no:1)
2.Leave a message(option no:5)
i have following settings in my general section
_ searchcontexts=yes
_sendvoicemail=yes
[test1]
1001
2007 Dec 20
1
Asterisk.NET API --help required
Hello all,
Here is the requirement from my side
to use Asterisk.NET API to generate
an automated call (outgoing) from asterisk
and then link to one of the extensions which
plays a sound file for the callee.
For this i have worked out in the follwing way
1)modified manager.conf to facilitate this API to talk to asterisk
2)used the command Originate to call a Registered user under
2009 Dec 18
2
Asterisk 1.6.2.0 Now Available!
The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and
Asterisk-Addons 1.6.2.0. These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.0 is the first feature release since Asterisk
1.6.1.0, which was released April 27, 2009. Many new features have been included
in this release. For a
2009 Dec 18
2
Asterisk 1.6.2.0 Now Available!
The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and
Asterisk-Addons 1.6.2.0. These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.0 is the first feature release since Asterisk
1.6.1.0, which was released April 27, 2009. Many new features have been included
in this release. For a
2010 Jan 04
1
Free FaxForAsterisk ReceiveFAX not working
Hello users,
Recently i have installed the free version of FaxForAsterisk and trying to
work with it by sending a fax
on T38.
My version information is as follows
i)Asterisk 1.6.0.20
ii)res_fax-1.6.0.14_1.1.6-x86_32
iii)res_fax_digium-1.6.0.14_1.1.6-i686_32
sip.conf
[general]
t38pt_udptl=yes
extensions.conf
[default]
exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call)
exten =>