Displaying 20 results from an estimated 100 matches similar to: "HELP on AVM Fritz with CAPI drivers for SMP RH 9"
2004 Sep 03
2
OH323 0.6.3b compilation problem with 1.0 RC2 on RH9
Hello,
I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2
installed but failed. I applied the patch to the required OpenH323 library
according to the instructions, and set the proper directories in the Makefile.
Here is what I receive after I issue make:
*******************************
g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -
2011 Apr 15
5
Possible bug in Hangup() (Asterisk 1.4.x)
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten => _X.,1,Dial(SIP/12345 at peer01,,,)
exten => i,1,Hangup(${HANGUPCAUSE})
exten => t,1,Hangup(${HANGUPCAUSE})
exten => h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(<cause
code>) commands, if the call is not answered by peer01 for any reason,
the actual cause code
2003 Apr 01
2
CE certification for Europe
Hello,
I'd like to ask if there are any news about CE certification of the E1
boards. I know that the T1 boards are FCC certified but I'd also like to
know what is the status for CE certification.
Thanks for any input,
Vlasis Hatzistavrou.
2004 Sep 16
1
Problems with native h323 channel on Asterisk RC2: No early audio and codec negotiation issues
Hello all,
We have been testing Asterisk RC2 with the native H323 channel driver.
We followed the instructions with the needed OpenH323 and PWLib versions
and everything compiled ok. Operation of the driver seems ok, except
from 2 main points:
1) Audio is passed between the two ends of the call only after the call
is answered. This was not the case with previous versions of Asterisk
(0.9.2
2006 Feb 17
1
FW: AGI onAnswer function: does it exist?
Hello,
Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list to ask this question?
Best regards,
Vlasis Hatzistavrou.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vlasis Hatzistavrou
Sent: Thursday, February 16, 2006 3:43 PM
To: asterisk-users@lists.digium.com
Cc: 'Vlasis
2017 Jul 03
2
DMTF in clock rates other than 8000 for chan_sip
Hello,
Does anyone know whether chan_sip in Asterisk supports DTMF in clock
rates other than 8000? I looked for telephone-event/16000 in the
changelog and in Jira but no luck.
Any help would be appreciated.
--
Best regards,
Vlasis Chatzistavrou.
2004 Dec 07
3
can't compile chan_capi 3.5 after patch applied :-(
Hi I use RH 9 + asterisk v1.0 stable + 2 PCI fritz cards + chan_capi
3.5 and it works fine,
Since my users want fax fonctionnality and customers know 1 of the msm
as fax number I wanted to try the chan_capi-0.3.5 patch
if I patch chan_capi and run make, I get an error message , as you
can read below, orginal chan_capi compile works, when patched I get
an error, no CID ?
Any idea ?
anybody
2003 Mar 31
2
modem.conf i4l issues
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2005 Jul 19
1
spandsp - fax is just blank pages
I've done quite a bit of googling and haven't found a solution to my problem.
I've got the Digium dev kit (wctdm11b) set up and working. I've
compiled spandsp and can receieve faxes from eFax (www.efax.com) but
the pages are blank. The page count is correct, in that if I fax a
two page document, my tiff file has two pages, but they are white
blank pages.
I found one similar
2005 Oct 05
1
Caching DTMF tones for get_data AGI?
I'm using get_data in an AGI script and am having a problem when, after
a long time in my IVR, when I ask for a 10-digit phone number, the first
few tries are always invalid -- the number it reads back is very
strange, almost like the DTMF tones from other answers were being cached
and then dumped on the call to get_data.
Anyone ever experienced this before? I have to do some major
2008 Dec 04
1
OT - Is sourceforge OpenH323 active ?
Hi,
A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if
this location is the one to use (I got trouble in the past when google
pointed to an obsolete site) :
some quite old messages remain unanswered.
Cheers
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2009 Mar 16
1
T.38 - Which endpoint shall reINVITE ? caller or callee ?
Hi,
I've been playing with T.38.
I observed that mostly but not always, it's the "calling endpoint" that
reINVITE the other party to drop current SIP/G711 session and start a new
T.38.
But sometimes, it's also the callee party that reINVITE the calling party.
Which is the "standardized" or most common, way to start a T.38 session ?
Shall it come from callee or
2006 Nov 30
1
CAPI module issue
Hi List,
I am experiencing an issue in a server that I have installed asterisk;
configured an AVM FRITZ card to work with the capi module.
Once istalled the card works perfect; however every time I reboot the
machine I found that I have to re install the capi4k-utils before I can load
asterisk otherwise the capi module will not loadup.
Can anyone direct me in the right direction in order to
2004 Jan 16
7
CAPI not installed, after changed from i4l to CAPI
I had unexpected hangups from my asterix box using the i4l driver. (SIP
<-> SIP calls worked execellent, but SIP<->ISDN didn't.)
Then I changed the i4l driver in modem.conf with the chan_capi from
jungham. (http://www.junghanns.net/asterisk)
I followed his instructions in the INSTALL file, and first encountered
some errors compiling it. It help by deinstalling several
2003 Feb 27
7
Interest in E1 channel banks?
Our company manufactures an E1 channel bank that is approved for use in
Australia (it should also be compatible with Euro standards). It is modular
and available in 10, 20 or 30 analog port configurations. Signal monitoring
and configuration is via Ethernet.
These units are manufactured in low quantities for specific telco
requirements. However if there was enough interest, we would be able to
2005 Jan 30
3
Callgroup with bristuff ISDN?
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 extension.conf?
How will this work for ISDN BRI/PRI?
I don't want some extensions to get all calls from the BRI/PRI, just
2003 Apr 05
0
Re: Asterisk-Users digest, Vol 1 #237 - 11 msgs
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2004 Jul 15
4
Kernel panic with two Fritz cards
So I've got a PBX running with one Fritz card with the fcpci module and that
works fine, but add another one and... kersplat... Kernel Panic.
It boots fine with two cards, but panics as soon as the first fcpci.ko
module is 'modprobe'd - I don't even get as far as modprobing the second.
Any ideas on what's going on? Should I just back-up to an older version of
the fcpci module
2004 Jul 12
2
Oz ISDN
In Australia, Telstra, the local telco provides isdn modem for isdn
connection. The modem has 2 analogue telephone jacks and a serial port
for connection to dialup internet.
My question is that will it be possible to use Zaptel TDM02B to connect
to the analogue jack instead of getting a fritz card to do the
telephony. Will there be less feature if doing so?
--
David Kwok, CISSP
Tel: 612
2010 Aug 10
1
Dial option 'r' not working anymore?
Hello,
I have used the Dial option 'r' before in older Asterisk versions and it
behaved as expected, that is, Asterisk would always give ringback audio
before the call was answered no matter what.
It has been a while that I have used version 1.4.33.1 without any the
'r' option. Now that I had to use it for a while, I noticed that 'r'
would not give ANY audio until the