:10380
/var/lib/asterisk/mohstream-chatfire-2.sh:
#!/bin/bash
/usr/bin/mpg123 -q -r 8000 -b 0 --mono -s http://stream.laut.fm/chat-fire
That's it :)
Regards
Markus
Am 27.08.2012 21:29, schrieb Matthew Jordan:
- Original Message -
From: Markus unive...@truemetal.org
To: Asterisk
Background music during a call
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg254252.html
Does anyone have the right solution and is available to create a
dialplan for me for cash? Please get in touch!
Thank you!
Markus
of misunderstanding here. Maybe I'm not
explaining it right...
Thanks!
Markus
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Hi Larry,
Am 26.08.2012 03:57, schrieb Larry Moore:
On 26/08/2012 3:45 AM, Markus wrote:
When I receive an incoming call from a SIP peer where I've configured
disallow=all
allow=alaw
(and no other codec)
I can see the following NOTICE on the console:
Dropping incompatible voice frame SIP
puzzled. :)
Regards
Markus
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asterisk
,
but just one ] closing? Is that a typo?
Also, the doc shows:
GotoIf(condition?[label1]:label2)
Why is label1 in square brackets and label2 isn't?
I'm confused. :)
Thanks so much!
Markus
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? There are two opening square brackets, but just
one ] closing? Is that a typo?
Also, the doc shows:
GotoIf(condition?[label1]:label2)
Why is label1 in square brackets and label2 isn't?
I'm confused. :)
Thanks so much!
Markus
---
You need to run your logic the other way. What you're doing now
on the voice quality but the messages
on the console are quite annoying.
PS: The peer doesn't support ulaw.
PPS: Asterisk 10.7.0
Thanks a lot!
Markus
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New
.
PS: The peer doesn't support ulaw.
PPS: Asterisk 10.7.0
Thanks a lot!
Markus
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Am 06.05.2012 13:46, schrieb Shahid H:
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
Log the actual DTMF to your console, set in
Am 06.05.2012 13:46, schrieb Shahid H:
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
PS: You are only allowing the GSM codec for your
know once the list is ready.
I've already registered opennumberingplan.org :)
Am 29.03.2012 11:12, schrieb Lenz Emilitri:
DO you know if the doc files from the ITU are available somewhere for
download?
l.
2012/3/22 Markus unive...@truemetal.org mailto:unive...@truemetal.org
I hope
Am 28.03.2012 20:17, schrieb C. Savinovich:
The Way to make money is to help folks use the open source items
in the most efficient manner
Nobody wants to pay me $2,000 to install and configure A2billing, which
in my view, is a fairly low price for my time. There are people who do
that for
I hope this is not too off-topic. As a kind-of follow up to rate sheet
normalization I'm still trying to figure out a solution for: throw 10
ratesheets at a program and get the cheapest codes/providers as output.
For this purpose I believe I need a real, detailed, accurate list of all
the
Am 16.03.2012 04:14, schrieb Ast Coder:
I would be more interested in a system where quality routes are tested
with different providers because rate really doesn't matter if a call
can't be placed or if a destination is a fake one. We have seen many
fake destinations with top tier providers but
all the time on FreeSwitch Freenode channel and it probably
does on OpenSIPs as well. Check there for some good advice.
On Wed, Mar 14, 2012 at 7:35 PM, Benny Amorsen
benny+use...@amorsen.dk mailto:benny%2buse...@amorsen.dk wrote:
Markus unive...@truemetal.org mailto:unive
Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर):
On Thursday 15 Mar 2012, Markus wrote:
With like 10 different ratesheets from 10 different providers, of
which many change their rates every few days, manually doing it in
Excel is too time consuming...
Is it possible to get samples? I'd
Am 14.03.2012 00:34, schrieb James Sharp:
ping + arp isn't going to work if they're on a different VLAN.
I believe this will work:
Too complicated. Just have a look on the switch(es) the phones are
connected to.
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Am 15.03.2012 00:35, schrieb Benny Amorsen:
Markusunive...@truemetal.org writes:
Does such a thing exist?
How does a2billing do it? It should be pretty easy in an AGI. If you can
afford a linear lookup per call, just grep through the array of prefixes
to find the ones matching a particular
Am 07.03.2012 02:04, schrieb Mike Diehl:
I tried the chat as well with no effect. My German is a bit rusty, or I'd call
them
Most Germans speak English. :)
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But wouldn't that mean that every customer line is busy every 30 minutes
for a few milliseconds for real callers? Unless there is more than 1
concurrent call enabled on the customers line.
:)
Am 09.02.2012 15:59, schrieb Bryant Zimmerman:
We designed our solution the following way.
We have
Reply to self, missed the line count part. Nevermind then :)
Am 09.02.2012 18:10, schrieb Markus:
But wouldn't that mean that every customer line is busy every 30 minutes
for a few milliseconds for real callers? Unless there is more than 1
concurrent call enabled on the customers line
Can't help you with the SIP account but for geographical numbers in all
3 countries that you mentioned try http://www.globalnumbers.de -
forwarding to any SIP destination is free.
Am 01.02.2012 13:29, schrieb Christian Gansberger:
Hello List!
I'm searching for SIP-Providers in the following
Here's a script to call a bunch of numbers (list-of-numbers.txt), trigger a
new call every 10 seconds. Adjust for your needs:
-snip-
#!/bin/bash
for i in `cat list-of-numbers.txt`
do
echo /usr/sbin/asterisk -rx originate local/@from-local extension
$i@voipout
/usr/sbin/asterisk -rx originate
Hi again,
no one got an idea? :-( Or did my request not make any sense? Or is the
answer to obvious that no one bothers to reply? :-)
Thanks again!
On a second thought, I don't need the predetermined delay. I can probably
just set that with additional w's in the DialBackground command
Hi,
new to the list. Wondering if anyone has / knows of, a good rate importer
tool that can be used to standardize and normalize the ratesheets / rate
decks etc. obtained from various carriers so they can be analysed and
imported into a DB or be saved as a CSV or something?
I'm using
Hi,
has the following been done before respectively is it possible with
Asterisk? I searched the archives but couldn't locate anything.
1. Call to comes in via SIP.
2. Call is not answered yet but progress continues.
3. At the moment the call comes in something like this gets spawned in the
On a second thought, I don't need the predetermined delay. I can probably
just set that with additional w's in the DialBackground command (which I
made up).
So rather something like:
_X.,1,Progress
_X.,2,DialBackground(SIP/123456@provider,,D(ww${EwwXwwTwwEwwN}ww))
_X.,3,ConnectLegs
not its own
see also https://issues.asterisk.org/view.php?id=16901
I Guess the Problem applies mainly to Germany because it's an ISDN Message.
are there any solutions??
cheer Markus
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Hi,
I still don't know if it's a bug or if it's already fixed esp. what
exactly is the source...how could i find this out?
or where could i open the bug report?
thanks
Markus
Am 02.01.2010 19:16, schrieb Steve Totaro:
Did you open a bug report?
On Sat, Jan 2, 2010 at 12:37 PM, Markus Weiler
Could anybody give me a hint how to investigate that problem?
cheers Markus
Am 31.12.2009 18:17, schrieb Markus Weiler:
Sorry wrong topic...
Hi,
I'm issuing a Bridgeaction through the manager interface.
One Person is called, when answered second one is called first gets MoH.
After
/lib/tls/i686/cmov/libc.so.6
thanks all
Markus
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0xb7e7a49e in clone () from /lib/tls/i686/cmov/libc.so.6
thanks all
Markus
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late. 28 sec instead of 30 secs??
thanks
Markus
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pragmas in C/C++ programs.
in /etc/vimrc
set showmatch
although not really an asterisk-java issue :-)
Markus
sean darcy wrote:
Philipp Kempgen wrote:
sean darcy schrieb:
On 1.6.1, I must be losing my eyesight:
exten = _6000XXXNXXX,n,Set
Hi,
I installed Digiums Free Fax for Asterisk and found out, that it
automatically retries failed faxes, is there a way to stop that?
Thanks
Markus
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tried Digiums solution to test if it´s better and it is, all the
testfaxes went through.
T.38 worked instantly.
Configuration was pretty easy and well documented.
I think $50 per channel is not too much money either, just the
support...well there is none.
hope i could help
Markus
David
, please help. What am I missing?
Sorry for my bad English.
Regards,
Markus
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On Tue, 2009-03-10 at 15:44 -0500, Kevin P. Fleming wrote:
markus wrote:
I installed Asterisk following the instructions of the book
Asterisk: The Future of Telephony. (very nice book)
However, I failed.
You 'failed' because you installed Asterisk 1.6.0.6, which contains a
very large
On Jan 27, 2009, at 10:50 PM, Wilton Helm wrote:
YMMV. Mine certainly did. For the better.
My comments were more negative than I intended. My installation is
worthless at this point because it is only a cookbook example and
I haven't tried to modify it to meet my needs. I didn't intend
No, because total_cost = billable_seconds * destination_cost
--
Markus
On Jan 21, 2009, at 1:15 PM, Sriram wrote:
Hi
I am using Trixbox 2.4 and PRI lines..on the CDR i see many calls
that have duration of 0 seconds, but they are still shown as
ANSWERED . how come its possible when
really
customize what you are graphing in cacti.
--
Markus
2009/1/10 Markus A. Wipfler mar...@infocom.co.ug
Another way to monitor this via cacti (for example if you don't have
snmp support for asterisk or need to customize what you are
graphing) is to create a new data input method
zap, iax, sip channels, how many concurrent calls from network
A to B, and more...
http://www.cacti.net/downloads/docs/html/making_scripts_work_with_cacti.html
I would also suggest to run the cacti poller every 1 minute rather
than the default 5.
--
Markus
On Jan 10, 2009, at 11:24 PM
Hello list
I am new in this list.
Before I wrote this email, i search with google and in the list
arichves for the question.
I look for a possibility to install FXO ports not over RJ11 Ports. I
will install the Ports by LSA+ Patch panel. Someone an idea ore link?
Thanks for help.
Bye MZ
PS:
Hi Group,
I have a problem with my FXO interface. It seems that asterisk cannot
see any configured channels, only pseudo. Pls see the error and my
config below.
Rgrds
Markus
-
May 2 15:09:00 NOTICE[17327]: app_dial.c:1010 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0
Hi,
try to set the TDMV_DCHAN = 16 (E1) or 24 (T1).
I had the same problem while updating from 2.3.4-(2|3) to 2.3.4-7 .
I think, that wancfg did not set this value correctly.
In my setup i updated an running system to new version, so the LinkLayer
is still ok.
Best Regards,
Markus
On 1/16
Peter Bowyer schrieb:
On 11/01/07, Markus Amann [EMAIL PROTECTED] wrote:
Hi
i have a trunk up and running with Asterisk and Sipgate.de and i can
make call out but no call in but the Enddevice Status on the Sipgate
Webpage says offline.
Maybe somebody had the same problem in the past and can
Peter Bowyer schrieb:
On 11/01/07, Markus Amann [EMAIL PROTECTED] wrote:
Hi
i have a trunk up and running with Asterisk and Sipgate.de and i can
make call out but no call in but the Enddevice Status on the Sipgate
Webpage says offline.
Maybe somebody had the same problem in the past and can
Hi
i have a trunk up and running with Asterisk and Sipgate.de and i can
make call out but no call in but the Enddevice Status on the Sipgate
Webpage says offline.
Maybe somebody had the same problem in the past and can give me some hints ?
Regards
MArkus
Markus
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I fix this problem ?
Bye
MArkus
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John Joseph wrote:
--- Markus Schuster [EMAIL PROTECTED] wrote:
Could you please post some details (or even better:
write them in some sort
of Wiki) on the configuration you did on the Nokia?
I tried to put some details on Voip-info.org ,
please check the link
http://www.voip-info.org
a short web search there
seem to be some problems about the correct configuration of the phone.
Greetings,
Markus
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) '04321234567' ]
it is like having a user provided number not screened, and a
networkprovided number
screened.
Best Regards,
Markus
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-46,48-62,63-77,79-93
Asterisk: 1.2.4
Zaptel: 1.2.3
Libpri: 1.2.2
Best Regards
Markus
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in pthread_start_thread () from /lib/libpthread.so.0
is this a known problem and we should switch an other version?
best regards
Markus
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While using Blindtransfer #Extension
everything works fine.
But how do i activate announced transfer
with an Grandstream GPX2000 ?
Greets
Markus
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,
Markus Hakansson
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-interface to the
console, but still want to be able to recieve calls without
autoanswering them...
Sincerely,
Markus
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()
Sincerely,
Markus Hakansson
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?
When the OSS-driver works (sometimes on one of the boards) it has very
low latency and good sound-quality.
Sincerely,
Markus Hakansson
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to reregister again on the SAME server. * is not
rereding the srv entry!
i want the * to reconnect complete by changing to the next SRV entry. (
if i restart gracefully it works in the same second. if i wait it takes
minutes)
is this possible???
thanx for helping
Markus Dörfler
try something like this:
# Uncomment this if you want to use digest authentication
if (!www_authorize(, subscriber)) {
www_challenge(, 0);
break;
};
maybe you have Problems with your realm.
And this seems not to be the list where you can find good help for your
Problem!
Best Regards
markus
Hi,
is it possible to switch to an extension to
send a SIP 404 Message?
Something like:
[404]
exten = i,1,SetVar,PriCause=0
exten = i,2,Hangup()
I know this is for PRI Signaling but i did not
find the same for SIP.
Greets
Markus
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Markus
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;allow=g729
secret=password
username=sipgateid
fromuser=sipgateid
fromdomain=sipgate.de
type=friend
host=sipgate.de
context=incomming
canreinvite=no
nat=1
Best Regards,
Markus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Patrick
Gesendet
Hello Jason,
No, in this case I only needed to remove the old source code completely and
make a new checkout. After that compiling works fine without changing the
make file.
Thanks,
Markus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von
oad_module':chan_capi.c:2607:
error: `iflock' undeclared (first use in this function)chan_capi.c: In
function `usecount':chan_capi.c:2820: error: `usecnt_lock' undeclared (first
use in this function)make: *** [chan_capi.o] Error
1deskpro:/usr/src/asterisk/chan_capi-0.3.5 #
Best
Regards,
Markus
at present
Asterisk CVS-HEAD-07/07/04-18:53:32 ,same time we
checked out libpri.
Any ideas?
thx,
Markus
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ax2.so
noload => chan_zap.so
noload => chan_alsa.so
noload => chan_oss.so
[global]
chan_capi.so=yes
And the capi.conf is:
[general]
nationalprefix=0
internationalprefix=00
[interfaces]
msn=9142829
incomingmsn=9142829
controller=1
softdtmf=1
devices=2
Does anyone have an idea whats going wrong
Hello,
which firmware are you using? I don't have problems like that. If I dial and
no one picks up the phone it stops ringing on the other side. I didn't have
a trace on hand right now but I'm pretty sure that I saw those Cancel
during other tracings.
Best Regards,
Markus
-Ursprüngliche
Hello Pertti,
I'm running the ZyXEL with WEP (128BIT) here at home and I don't have
problems with the voice quality.
Best Regards,
Markus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von
Pertti Pikkarainen
Gesendet: Donnerstag, 3. Juni
= national
to
switchtype = 5ess
did the trick.
Thanks.
-Markus
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Calltrex Corporation
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numbering plan.
The question now is: how do I tell Asterisk to send everything
starting with 011 as unknown numbering plan?
Thanks for your help.
-Markus
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Calltrex Corporation
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bindaddr = 0.0.0.0
context = incoming
disable=all
allow=alaw
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
canreinvite=yes
register = 8009090:[EMAIL PROTECTED]/8009090
[100]
type=friend
context=out
callerid=100
username=Markus
host=dynamic
dtmfmode=info
yes i am using stable,
ok i will try unstable,
thanx a lot ,
Markus
Am Mit, den 12.05.2004 schrieb Karl Brose um 18:55:
Sounds like you are using CVS 1.0stable?
Proxy Authentication is broken in that CVS head, and it may not get fixed.
Using development head will fix this.
see also
R=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o say.o say.c
say.c: In function `ast_say_number_full_de':
say.c:769: parse error before `int'
say.c:770: `thousands' undeclared (first use in this function)
say.c:770: (Each undeclared identifier is reported only
Failed to authenticate on INVITE to 'xyz
sip:[EMAIL PROTECTED];tag=as4ddd4a6f'
what says sip debug?
greets
Markus
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getting zero as p-lastresult (CSeq
== 0 )?
Also it looks like a bug in the grandstreamfirmware sending CSeq zero?
would something like this solve the Problem?
if (!p-lastinvite = 0 !strlen(p-randdata))
^
?
Best Regards
Markus
the callerID of the 1st box. Apparently the 1st Asterisk box replaces
the original callerID with its own.
Is there a way to keep the original (real) callerID in place.
Thx.
-Markus
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Calltrex Corporation
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The one I know of is X-Pro/X-Lite from http://www.xten.com/
I doubt that there is a Linux version available...
Markus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Martin Mielke
Gesendet: Dienstag, 6. April 2004 15:30
An: [EMAIL PROTECTED
no clue what to do anymore
Regards,
Markus
Virus checked by G DATA AntiVirusKit
Version: AVK 14.0.635 from 31.03.2004
Virus news: www.antiviruslab.com
-0357
Cell: (309) 657-6365
Fax: (309) 213-3500
E-Mail: [EMAIL PROTECTED]
Customer Service: (877) 976-0711
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Markus Miertschink
Sent: Tuesday, April 06, 2004
10:09 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk
of Asterisk or is there function to
force the check of the external IP on a regular base?
Hope, someone can
help me...
Best
regards,
Markus
to killall -9 asterisk on ast-2 (because 'stop now' would hang
indefinitely) and restart it. After that ast-2 would immediately
re-register with ast-1 and everything would start working again.
Any pointers of what's wrong here?
Regards,
Markus
that follows a pattern. Any ideas how to find out what exactly is
going on here and which box is tropping/losing packets or if it happens
somewhere in betweeen (hops we can't control)?
Best regards,
Markus
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I wonder if there is a possibility to access the
InfoElement (provided by chan_capi debugging)
in context - maybe thru a variable - It would be a perfect
way of getting our HiPath to deliver Display-Names to
SIP(Soft)Phones.
Does anybody know about a solution?
Greetings,
Markus
simultaneous calls cause simultaneous rings (note: that's not the same
as 'ringall').
Thanks,
Markus
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Ok, has already been answered. Thx, folks.
-Markus
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The original posting seemed to be buried inside a thread so I reposted.
But then the original thread sprang back to life.
-Markus
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doesn't work for us), but we also want more than one phone
ringing if there's more than one call coming in at the same time.
Thanks,
Markus
On Thu, 2003-12-11 at 14:36, Brancaleoni Matteo wrote:
so, from queues.conf.sample
A strategy may be specified. Valid strategies include:
ringall
like this and still get everything to work again? Rebooting can be a
huge pain.
Thanks.
Regards,
Markus
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