it down to the Music On Hold.
Scenario:
- My boss calls my cellphone
- I do not pick up
- My cellphone's voicemail picks up
- He hangs up
- Asterisk plays On Hold music to voicemail until voicemail on cellphone
times out
- Zap channel is stuck
-- Thomas
to create channel of type 'Zap'
== Everyone is busy at this time
ztcfg shows no errors.
So where's my problem?!?
Regards,
thomas
PS:
zapata.conf:
[...]
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel = 1-15,17-31
zaptel.conf:
span=1,0,0,ccs,hdb3,crc4 #,yellow
bchan=1-15,17-31
. Use a software phone (see
voip-info.org) so you can get
your demo coded. Then once you got the rest of the hardware you can hang
it onto the PSTN. This
will get you going for now. I guess you could just use some sort of SIP
provider to give you an dial
in phone line too.
-- Thomas
Duane wrote:
Grandstream v1.0.4.68 firmware
http://www.hellofone.com/downloads.html
Seems to have loaded ok on my BT100..
Hmmm well I need to kinda figure out how to get the custom ringtones to
ring on the phone... :-)
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trained
Segmentation fault
Any ideas...?
Regards,
Thomas
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that is not that easy. If you can just group 2 of the 4 pots
lines together I guess you can create a dialplan like that. But then you
could only have 2 incoming lines per number but 4 outgoing lines.
Thats all for now (though Im sure this wont be my last post. Thanks
again for any help.
- Mike Stupak
-- Thomas
Mikael Andersson wrote:
Thomas Gallaway -- wrote on den 17 maj 2004 16:57:
Hmmm well I need to kinda figure out how to get the custom ringtones
to ring on the phone... :-)
___ Asterisk-Users
or how to change them
/M
Yeah I can change
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the ring tones, while present on
the server and
Jeremy McNamara wrote:
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the ring tones, while present
Brian Capouch wrote:
Thomas Gallaway wrote:
My ringtones just work on all the grandstream's :-)
Do the URLS for the ringtones at the top show up as something other
than all zeroes?
I've fiddled with this until blue in the face, and the ring sounds
just like the ring it had before
Stephen R. Besch wrote:
Thomas Galloway wrote:
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the ring
Stephen R. Besch wrote:
Thomas Gallaway wrote:
Brian Capouch wrote:
Thomas Gallaway wrote:
My ringtones just work on all the grandstream's :-)
Do the URLS for the ringtones at the top show up as something
other than all zeroes?
I've fiddled with this until blue in the face, and the ring
Jerry Geis wrote:
I just grabbed my fresh Fedora Core 2 final release.
Untared zaptel-0.9.1 dir make linux26 and I get errors on
the compile. Anyone else tried this yet and been sucessful?
I did have to ln -s /usr/src/linux-2.6.5-1.358 /usr/src/linux-2.6
But I still get errors after that... about
or inside and set the
right corresponding parameters. The RTP will still bind on all ports
currently, but that will be fixed in a matter of days.
Also, sipgate.net should be sipgate.de (works ok though since they
don't care)
fromdomain is meant to be realm not a hostname.
Thomas Gallaway wrote:
Hi
WipeOut wrote:
Joshua M. Thompson wrote:
On Thu, 2004-05-20 at 05:12, WipeOut wrote:
When trying to build zaptel it required me to link
/usr/scr/linux-2.6 to the default source dir which is
/usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :)
After than I tried again but
Hi
I try to use zapras. I am using zaptel-bri-0.0.2
I compiled and patched pppd from ftp://ftp.digium.com/pub/zaptel/misc/
pppd is /usr/sbin/pppd
Any idea whats going wrong here?
Thomas
-- Accepting call from '95' to '8526' on channel 1, span 1
-- AGI Script nuller.agi completed, returning
On Tue, May 25, 2004 at 08:37:34AM -0600, Dan Cunningham wrote:
I downgraded to 3.5.7 and that seemed to work, for others with this issue I
just manually downloaded these packages from debian/testing and downgraded
with dpkg -i libtiff*.pkg , this worked with no dependency problems.
Think
Hi folks
I'am looking for the right way to select the outgoing MSN on zaphfc
for Euro-ISDN.
I found some notes on the Wiki and I know it has to be done in the
dialplan.
Does anyone know the right way/code?
THX
--
Tho/\/\as
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On Wed, May 26, 2004 at 05:54:52PM +0200, Julian Pawlowski wrote:
Hi Thomas!
you have to set the MSN this way for zaphfc when you use the dial command:
exten = _0Z.,5,Dial(CAPI/MyMSN:${EXTEN},90,mT)
zap=capi???
For Capi its clear
, 2004-05-26 um 20.04 schrieb Thomas Niesel:
On Wed, May 26, 2004 at 05:54:52PM +0200, Julian Pawlowski wrote:
Hi Thomas!
you have to set the MSN this way for zaphfc when you use the dial command:
exten = _0Z.,5,Dial(CAPI/MyMSN:${EXTEN},90,mT
adam -
can the g729.dll be downloaded somewhere
- is this still required for g.729 support?
Regards,
Steven Thomas
jo [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
31/05/2004 09:19 PM
Please respond to
asterisk-users
To
[EMAIL PROTECTED]
cc
Subject
Re: [Asterisk-Users
Hello everybody,
we've implemented a new Channel Driver for *. It uses the new mISDN
isdn4linux architecture and supports bri te and nt mode for now.
I assume, there are lots of bugs we didn't found yet, and even mISDN is
rarely stable. So we search brave volunteers to test the driver.
Get it
Hallo Matthew Boehm
On Mon, 20 Sep 2004 16:54:33 -0500 you wrote:
I finally got 2.4 recompiled with RTC as a module:
Module Size Used byNot tainted
autofs 13684 0 (autoclean) (unused)
acenic241092 0 (unused)
iptable_filter
Hallo Martin Mielke
On Tue, 21 Sep 2004 14:32:34 +0200 you wrote:
...cut
so the isdn4linux drivers are correctly loaded. I know, CAPI should do
better but I can't compile from the tarball (see my post about it)
When trying to dial the PSTN using the ISDN interface I get:
---
*CLI
Hallo Martin Mielke
On Tue, 21 Sep 2004 17:03:54 +0200 you wrote:
Thomas Niesel wrote:
[ snip ]
Does the phone had the same MSN?
I think so. It could dial outside without a problem...
Is there maybe a PBX needs a leading Digit to get outside line?
No, those
Hallo Daniel Eboa
On Tue, 21 Sep 2004 16:16:44 +0100 you wrote:
Hello to all,
I'm new user of Asterisk. I'm running Asterisk on a RedHat 9 platform.
Everything seems to be ok but I got lot of error messages and I don't
know their meaning. Can somebody help me ??
These are the
On Tue, Sep 21, 2004 at 09:00:18PM +0300, Vlasis Chatzistayrou wrote:
Hello,
I have been wrestling with installing the CAPI drivers for AVM Fritz in order
to use chan_capi with Asterisk.
I use an SMP machine, RH 9. I have found rpm's for CAPI and AVM drivers
(namely:
On Wed, Sep 22, 2004 at 07:15:19PM +0200, [EMAIL PROTECTED] wrote:
Hello all,
I'm trying to setup a AVM C2 card.
I have read the kernel requirements for this card.
M CAPI2.0 support
[*] Verbose reason code reporting (Kernel size +=7K)
[*] CAPI2.0 Middleware support (EXPERIMENTAL)
On Wed, Sep 22, 2004 at 07:15:19PM +0200, [EMAIL PROTECTED] wrote:
Hello all,
Sorry for my first mail which answers the 2nd part:(
I'm trying to setup a AVM C2 card.
I have read the kernel requirements for this card.
M CAPI2.0 support
[*] Verbose reason code reporting (Kernel size
On Thu, Sep 23, 2004 at 01:10:44AM +0200, Gunther Stammwitz wrote:
Hello,
I just got my isdn-card working together with i4l and asterisk.
Everything seems to be working fine: I can accept calls coming from the
outside and I can dial out. Even setting the msn works like charm but my
problem
/area_codes/index.html
I also came across a 1999 version of the whole shebang in a text file
here: http://sd.wareonearth.com/~phil/npanxx/
It would be nice to be able to accurately lookup city and state for
billing. Anybody have a resource?
Thanks,
Thomas Hutton
On Thu, Sep 23, 2004 at 11:03:21AM +0200, [EMAIL PROTECTED] wrote:
Hello,
I?m trying to compile the Fritz CAPI module for Debian stable, following
the steps related in
http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install
But I always get
On Sat, Sep 25, 2004 at 10:12:44AM +0200, Joost Kraaijeveld wrote:
Hi all,
I am trying to get my Debian Sarge to work with 2 Winbond W6692 chipset based ISDN
cards and Asterisk 1:0.9.1+1.0RC1-8. I have installed CAPI and chan_capi (all latest
testing versions).
If I start asterisk I
suggestions?
Thanks,
Thomas Hutton
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Hi all,
i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco
ATA-186 3.1.1 atamgcp
We are used to make an special ;) blind transfer like
(Flash)Number(Hangup before anyone answers or ring).
Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp
If one
hi list,
how can i disable that one sip clients can login twice?
thx for help,
thomas
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sufficient.
As for LDAP, search for that on the Wiki.
http://voip-info.org/tiki-searchresults.php?words=LDAPwhere=pagessearch=go
Thomas Hutton
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ist there any cdr dokumentaion about the cdr format?
thx!
thomas
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-- Asterisk1 --MGCP-- Asterisk2 --SIP-- Softphone2.
Any help will be much appreciated.
Best Regards
Thomas
--
===
Arkateq (Pty) Ltd
Tel: +27 12 6655360 6 Hazel Court
Fax: +27 12 6655201 160 Witch-hazel
, and this
is a version. 2.4..
Can anyone explan what i am doing wrong
?
Best Regard
Thomas H.
Denmark.
cc -c zaprtc.c -D__KERNEL__ -DMODULE
-DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include
-WallIn file included from
/usr/include/linux/prefetch.h:13,
from
/usr/include/linux/list.h:6
give me a hint..
Best Regards
Thomas Hupfeldt.
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the new kernel the default by updating
/boot/grub/grub.conf
every things works.. i think.. but I cant update the grub.conf, because it
didnt exist.
8) Try to rebuild zaprtc.
:o( Still doesnt work..
Best Regards
Thomas H.
Here are what happens (some of the text):
from /usr/src/linux
get this message:
Kelnel Panic: No Init found.. Try Passing init= option to kernel.. :o(
Now i need to install it from scratch..
Regards
Thomas.. :o(
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- Original Message -
From: Thomas Hupfeldt
Hmm.. somehow i did a mistake, so my mandrake is fucked up..
When i boot, i get this message:
Kelnel Panic: No Init found.. Try Passing init= option to kernel.. :o(
Now i need to install it from scratch..
I resqued my mandrake using
scratch, was that i
didnt knew to the rescue function on the cd's.
Do any of you know, wheter it would help to installe the newest mandrake
10.1 instead of the v. 9.2 ?
And as I said before, im a really newbe to linux, and there for i dont know
hot to rescue it...
Regards
Thomas H
was quiet new
in this.. :o)
I return when i get mandrake 10.1 installed..
I thank you for your help so far..
Best Regards.
Thomas Hupfeldt.
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Hello again :o)
I just installed mandrake 10.1, and now i want to
compile Asterisk..
Then i need to install openssl-devel, but where do
i find that to mandrake 10.1 ?
Best Regards
Thomas H.
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http
-devel.
/usr/bin/ld: cannot find -lssl
collect2: ld returned 1 exit status
make: *** [asterisk] Fejl 1
Is'nt there a openssl-devel to mandrake 10.1 ?
Regards
Thomas H.
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I could do chan_dect.so
There are PCI DECT Boards but I don't know if they will do base-station
stuff.
*http://tinyurl.com/3hnq5
Maybe somebody else has more insight into this if those PCI and PCMCIA
Dect boards can be turned into base-stations.
-- Thomas
Hallo rich allen
On Mon, 1 Nov 2004 12:30:43 -0900 you wrote:
iH
i have an IAXy which i can make calls from but am unable to call. when
i dial the extension assigned, i get the following from the console;
-- Executing Dial(SIP/5801-b665, IAX2/[EMAIL PROTECTED]) in new
stack
Hallo Paulo Adriano
On Tue, 02 Nov 2004 10:02:53 + you wrote:
I have just added a Conceptronic ISDN card to use as my only BRI.
1 - modprobe capi detects the card fine (i can see that in the log)
Shure?
Next Asterisk documentation says I should install CAPI drivers ?? My
card
Hi List
I recently got one of those boxes (without wlan).
Works as ata device with my local asterisk.
Just tested the basic stuff like call the box and make call from the box.
It uses sip with alaw/ulaw/g726 codecs.
Runs on linux, kernel 2.4.17, mipsel.
HardWare:
-wan (UR2/annexB)
-ethernet
Hallo HBK
On Thu, 04 Nov 2004 19:07:37 +0100 you wrote:
Hi
I am trying the fine iso at http://www.asterisk.de.ms/ but are having
problems with Capi probably due to having to old Fritz PCI card. Trying
with both non version marked version and version marked V 2.0.
I get following error
Hallo Chris W
On Sat, 06 Nov 2004 14:35:04 +0100 you wrote:
I wonder if someone out there can help.
Asterisk seems not to be recognising my Fritz card despite the drivers
being installed as far as I can tell correctly.
I'm runnign Redhat 9; Asterisk 1.0 from CVS; chan_capi is installed
respond on the old IP's when you ping them. Shut down any of
these machines on your network, clear out the leases, and try again.
Thomas Hutton
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Hi,
Am I correct in saying that the wcfxs kernel module is something of the
past, and is now replaced by wctdm ?
Regards,
Thomas
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What does this error mean:
Nov 14 10:35:12 WARNING[24733]: Unable to get our IP address, Skinny disabled
I looked in channels/chan_skinny.c and it looks like ourhost[] is never
initialised ?
$
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channels configured.
I've used this link to set up the ports:
http://www.digium.com/index.php?menu=faq#Configuration_0
How do I debug this ? Is there supposed to be a log in
/var/log/asterisk/event_log ?
Regards,
Thomas
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On Sun, Nov 14, 2004 at 11:19:26AM +0200, Thomas Andrews wrote:
I hear no dialtone on the telephone plugged into the TDM400 card.
Here's the relevant output from dmesg:
Zapata Telephony Interface Registered on major 196
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed
On Sun, Nov 14, 2004 at 10:32:40PM +1300, Matt Riddell wrote:
Power alarm on module 1, resetting!
Have you plugged the power into the TDM400P?
I have yes.
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On Sun, Nov 14, 2004 at 10:32:40PM +1300, Matt Riddell wrote:
Power alarm on module 1, resetting!
Have you plugged the power into the TDM400P?
The wierd thing is that asterisk refuses to start *until* I've had those
Power alarm error messages. Until then I get these errors:
= (Zapata
them, and it always does afterwards
Thanks for the help!
Thomas
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signalling=fxo_ls
channel=1
context=incoming
signalling=fxs_ls
channel=2
Regards,
Thomas
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is for the handset
I don't agree, but perhaps I'm wrong. The green module (fxs) is number 1
and the red module (fxo) is number 2. As I understand it you plug the
handset into the green one (fxs). Not so ?
Thanks for the help.
Thomas
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On Thursday 19 July 2007 04:27, Tzafrir Cohen wrote:
On Wed, Jul 18, 2007 at 12:44:29AM +0200, Thomas Winter wrote:
Hi,
Frozen or crashed? Do you see the console of the system?
serial console is dead.
kernel is 2.6.18-4 debian Etch.
bristuff is latest zaptel-1.2.19 and asterisk-1.2.22
I
Michael Munger wrote:
I agree it is the NAT in the router.
Does anyone know what the ip tables command would be to pass IAX to an
Asterisk box on the LAN?
It depends a lot on what your current setup is, but something akin to:
iptables -A PREROUTING -t nat -p tcp -i eth0 --dport 4569 -j
Hi!
At the moment i am using a digium tdm400 card for my analog phone lines.
The zaptel driver supports fax detection, so incoming faxes are
redirected to the fax extension automatically.
This works without problems with asterisk 1.2.
But now I would like to switch to ISDN (mISDN) and asterisk
Anselm Martin Hoffmeister wrote:
I did something similar using multiple records in a row.
Something like
exten = 931,1,Answer()
exten = 931,2,Wait(2)
exten = 931,3,Set(E=1000)
exten = 931,4,Playback(beep)
exten = 931,5,Set(E=$[${E} + 1])
exten =
hmmm.. no ideas?! :-|
tom
Thomas Artner wrote:
Hi!
At the moment i am using a digium tdm400 card for my analog phone lines.
The zaptel driver supports fax detection, so incoming faxes are
redirected to the fax extension automatically.
This works without problems with asterisk 1.2
Lee Howard wrote:
Artifex Maximus wrote:
zttest is often on 99.975586% with final result:
--- Results after 67 passes ---
Best: 99.987793 -- Worst: 99.951172 -- Average: 99.973764
This is unacceptable for faxing, and it is evidence of the underlying
problem also causing your faxes to
Lee Howard wrote:
Correct. IRQ 2 bridges to IRQ 8. Thus the priority order is:
0, 1, 2, 8, 9, 10, 11, 12, 13, 14, 15, 3, 4, 5, 6, 7
My zttest results weren't quite as bad as the previous poster.
Home Machine.
--- Results after 113 passes ---
Best: 100.00 -- Worst: 99.987793 --
Michael Munger wrote:
I have this exact same problem with two different Business Edition
systems. Both are using TDM400s.
Do we have an answer for this yet?
I know this sounds silly, but if there is a chance that it is an
improperly tuned echo canceller, has anyone tried using oslec.
Jason Parker wrote:
Administrator TOOTAI wrote:
Hi all,
I receive this error while compiling chan_mobile:
gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
chan_mobile.c: In function `mbl_load_config':
chan_mobile.c:1745: erreur: trop d'arguments pour la fonction «
Seysan wrote:
Hi all,
I want to limit the outgoing trunk to certain extensions, so for example
6 extensions can call long distance, but 4 other extensions are not
allowed to do so.
How can I do it in FreePBX specially?
I don't know about Trixbox per say, but normally you would have all
Joe Acquisto wrote:
I need to ask, to refresh, is the aux power connector on the TDM400P card
*only* to power the ringer on any
analog phones/devices on the system?
Can I still use this board, to terminate POTS lines and use all SIP Phones?
Yes, you only need to connect a power supply
Barton Fisher wrote:
Thanks, OK, a bit confused The cards are TE410P. I really don't
see how the set a codec for this, other than it might default to
something in code like ulaw. Any clue on how to verify codec in use
during a call?
G.711ulaw and G.711alaw are the audio transmission
Atis wrote:
A little caveat - sox doesn't understands file extensions used by
asterisk (or it's just asterisk, trying to use file extensions that
match codec name). So - some sox commandline hints:
ulaw: -t ul
alaw: -t al
slin: -t raw -s -w
Or (since 1.4.0) in the asterisk cli type:
I have replaced my SPA 3000 with a TDM-400P (which strangely isn't
considered to be a timing source)., and have been keeping the SPA3000 as
a power-down failover. (Home system).
Does anyone know of a device that could be used to replace it in this
purpose?
TIA.
Is there an advantage to having a Queue members URI in the form:
SIP/User (or indeed IAX2/User)
Over
Local/number@context
?
I know that the latter will allow you to do things like set counting
logic etc. through dialplan operations, but the former appears to be a
more direct route to calling
Tony Mountifield wrote:
I have a client who wants a Meetme box with 12 FXO ports, to connect
to Analogue lines coming from an Ericsson PBX.
It looks like I could do this with four different hardware configurations:
a) three TDM04B cards (based on TDM400P)
b) one TDM04B and one TDM808B
c)
Mojo with Horan Company, LLC wrote:
Those are all analog though, aren't they? What about a channel bank into
a digital card? Might that be cheaper than shelling out for 12 FXO
ports and the cards to hold them?
Just wanted to throw that out there before the discussion started :)
It
Steve Murphy wrote:
Oh, Julian, I'd imagine what I'm about to say will fuel some flames!
Here's a fairly powerful argument for all you asterisk users, as to why
you
should purchase a Digium product vs. a Sangoma: Because Digium uses a
chunk
of the purchase money to support Asterisk. And
Steve Totaro wrote:
Thomas Kenyon wrote:
Steve Murphy wrote:
Oh, Julian, I'd imagine what I'm about to say will fuel some flames!
Here's a fairly powerful argument for all you asterisk users, as to why
you
should purchase a Digium product vs. a Sangoma: Because Digium uses a
chunk
Philipp Kempgen wrote:
Don Kelly wrote:
http://www.sandman.com/autodial.html
These phones look like the ones we had in Germany
20 years ago. ;-P
I like the way that MSC1J really *is* the batphone.
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On Friday 12 October 2007 04:38, Ken D'Ambrosio wrote:
Hi, all. My company is setting up a branch office in Germany, and I'm
very interested in a VoIP provider over thataway. However, I'd need a few
things:
- Reliability. Can't have my branch office's DID's just going down. A
then you
Hi Everyone,
I cannot seem to get the voicemail gain option g(#) work in Asterisk
1.4.11. I am using it like so...
Voicemail([EMAIL PROTECTED],bg(10)) ; for busy announce and 10dB record gain
This has absolutely NO affect on the resulting voicemail wav file.
I have also tried using format=wav
I'm not sure how the gain option works as an argument to Voicemail(), but I
know
that the volgain option for e-mail attachments requires that you have sox
installed in order to work properly. If you don't already have it installed, I
would suggest installing sox and seeing if that helps.
I get the following error when trying to play wav files for my IVR
menu. Does anyone know what this means or how to fix it?
[Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say fmt
Thanks!
David
___
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Tzafrir Cohen wrote:
By now there are quite a few x86_64 Asterisk users that complain if
something breaks.
Been using it on a 64-bit P4 with debian 4.0/1 (amd64) for some time now
without a hitch.
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Hi,
Iam dialing from NT ptp to SIP provider.
Sometimes Asterisk is doing music on hold but there are no options like t or T
in the dial command. As an result the channel got lost and an Hangup occurs.
Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card.
Any solution for this?
Oct 22
I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.
Is anyone else getting the following error in the asterisk console:
[Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short
every couple of seconds when a handset is in a call?
I didn't notice this
Thomas Kenyon wrote:
I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.
Is anyone else getting the following error in the asterisk console:
[Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short
every couple of seconds when a handset
With wich codec is the channel working now, with ALAW or with g.729A
And what is the relevant value read/write format or nativeformat ?
Thanks for help,
Thomas.
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Hi,
is there anybody who knows this very little detail ???
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Donnerstag, 6. November 2003 10:54
An: Asterisk User
Betreff: [Asterisk-Users] which channel format number is right
(1.4.11) and openh323(1.11.7)
installed, and it worked.
Is here something wrong with this url ? Before i installed the new stuff it
worked so.
Can somebody help?
Thanks,
Thomas.
Here the trace level 2 log in oh323.log:
0:30.808H323 Listener:80f24d8 H323TCP Started connection:
host
Thanks Michael,
for this very special detail :-)
Regards,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
Manousos
Gesendet: Freitag, 7. November 2003 14:00
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Unable dial out
://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
Thomas
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Is it possible to log the CallerID of
an inbound call including the time to a log / text file? Also the
same for outbound? ie., dialed number and time?
Thanks.
Regards,
Steven Thomas
,
Steven Thomas
Technical Project Manager
Network Connectivity Services, IBM Australia
Ph: 0404 099 262
NH011, IBM Centre, St Leonards, 2065
Internet: [EMAIL PROTECTED]
Visit us at http://www.ibm.com/services/au/its
yes. Cisco 2612 Router with 2
x FXO's and 2 x FXS's. Works well using H323, and gnugk.
Steve.
Bruce Hedreen [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
15/12/2003 09:57 AM
Please respond to asterisk-users
To:
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