Hi Jeremy,
What about this in extentions.conf
exten = 5000,1,Dial(SIP/phone1SIP/phone2SIP/phone3,50,r)
--
mvh. Hans-Henrik Andresen
Jeremy Kenney [EMAIL PROTECTED] wrote in message
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Hello I am new to asterisk I want to setup the call queues where it will
ring
Hi,
Are there realy no-one who can help here
--
mvh. Hans-Henrik Andresen
--
Telefon for en flad 20'er - www.telefin.dk
--
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi
hmm - this is the bad thing about open source etc.
Should we make a bugreport ? or are we just doing something wrong ?
--
mvh. Hans-Henrik Andresen
--
Telefon for en flad 20'er - www.telefin.dk
--
usedcanon
When I google'ed this problem I can see other users also found this error
(bug ?) But no-one seems to have solved the problem.
Any clue ?
--
mvh. Hans-Henrik Andresen
--
Telefon for en flad 20'er - www.telefin.dk
clue ? or any how-to for the zapras ?
--
mvh. Hans-Henrik Andresen
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that it will be 5-700 sim. users only talking sip, and IAX2 to my
PSTN-Gateway.
The system is suposed to scale to 15000 users.
I'm ready to receive input :)
/Hans-Henrik Andresen
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agi, cause of
heavy load on the server, but an extensions.conf with 5000+ entrys is that
good ? I would preffer agi and a very littlte and simple extensions.conf.
Any experience with asterisk and 5000-15000 users ?
--
mvh. Hans-Henrik Andresen
Hi,
I had followed the installation-guide to festival
http://www.voip-info.org/wiki-Asterisk+festival+installation
speech-tools compiles OK, but I got this error when compiling asterisk
if I compile without the patch it compiles, but of cause did'nt work with
asterisk.
any clue ?
/Hans-Henrik
Hi,
When I make a call from my asterisk and it is passed thru another astrisk
eg. iaxtel, I got 'Answered' in my astrisk, and bill-sec is start counting
as soon I get connected to the other asterisk, and not if the party on the
other asterisk server pick up the phone. So IF the other party are
Hi,
Another one.
I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get
this in my logfile
Attempting native bridge of [EMAIL PROTECTED]/5 and IAX2[dialout]/6
Asterisk Version is CVS-04/19/04-22:17:41
What's wrong ?
I gues it has somethnig to do withe my bilsec-problem
Problem at partner site, some perl-problem with answer-command
/HHA
Whats wrong ?
Can I do something about it ?
/HHA
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does this meen ? Or what can I do ?
The router is behind nat, but if I put the router on the same network as
asterisk it work ok
/Hans-Henrik Andresen
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Hi,
My asterisk fails and stops after running the reload command ~20 times (I'm
testing) - is this a kown problem ?
Therefor I wil reload only sip, extensions and iax, it works with sip and
extensions, but it seem that there are no reload for iax - or what ?
--
mvh. Hans-Henrik Andresen
If you want
MusicOnHold and Conferencing however, you will need one card for the
timing.
Why - I had used only ztdummy that works for MOH and conf.
uncomment ztdummy in the makefile for zaptel and compile.
/Hans-Henrik Andresen
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Hi,
At my company we have a large CCM-installation, is it possible to / how to
connect between asterisk and CCM.
I'm quit shure that the CCM only use Skinny.
Any idea of the hardware-size for 1000 users ?
/Hans-Henrik Andresen
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Tanks
This was exatly what I needed,
/Hans-Henrik Andresen
Nicolas Gudino [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi Hans,
http://bugs.digium.com/bug_view_page.php?bug_id=773
This patch plays a tone 40,30,20 and 10 seconds before absolutetimeout
Hi,
I saw somewhere that it was possible to set a limit for how long time a call
could be, for an extension in extension.conf. But I can't find it anymore.
Can someone please help.
Calls to '411' an operator may max. be 5 min.
I have this in extension.conf.
[shortcuts]
exten =
Did you compile the zap and lipri and installed ?
/HHA
app_dial.c:533 dial_exec: Unable to create channel of type
'ZAP'
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, and if they make a VPN to my
asterisk it works.
(We had tried to use stun-server as well)
Any clue ?
/Hans-Henrik Andresen
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I have no problem transfer from one GS adaptor to another GS adaptor.
/Hans-Henrik Andresen
Can anyone confirm that this problem exists?
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Hi
I tried to raise it to 5000, but still unreachable.
But as I wrote earlyer, for the same config, sjphone and a Grandstream 286
works.
/HHA
qualify=1000
If the client turns UNREACHABLE, you might want to change the qualify=
setting to qualify=yes,
that defaults to two seconds, instead of
Hi,
Thank you, but this I cant get to work.
/HHA
so that should enable you to do the following:
Call timeout = 20 sec
Max Call Duration = 300 sec = 5 min.
exten =
411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20,S(300))
however, I have not tried it yet so someone
Thank you This works, but. It just cut the line, I had hoped for some
bip bip bip to remind that now your about to be disconected, is this
possible as well ?
/Hans-Henrik
Senad Jordanovic [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
exten = 1,AbsoluteTimeout ($SECONDS)
'
cdr_odbc: Query Successful!
-- Søren
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi,
Thank you, but this I cant get to work.
/HHA
so that should enable you to do the following:
Call timeout = 20 sec
Max Call Duration = 300 sec
arhh - I did a checkout the 4th of marts - I will do a new checkout
/HHA
When did you checkout your version of Asterisk from CVS ??
This feature was put into CVS on the 6'th as a fix for bug #1107 but I
have not seen it in v1-0_stable.
___
What checkout name should I do ?
Just asterisk ?
# cd /usr/src# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot#
cvs login - the password is anoncvs.# cvs checkout asterisk
/HHA
This is a new feature, that's why it is NOT in 1.0-stable.
Only bugfixes go into
Hi,
Anyone got danish voice-files who wants to share ?
/Hans-Henrik Andresen
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Hi,
On my Suse90-out of box I had downloaded from CVS asterisk.
I'm running kernel 2.4.21-99-smp4g with 4cpu's, and the kernelsource is in
/usr/src/linux
Asterisk compiles with no problem.
But when compiling zaptel I got this error
..
zaptel.c: In function
Greate -
/usr/src/linux/include/linux/version.h:6:2: #error The kernel sources in
/usr/src/linux are not yet configured.
/usr/src/linux/include/linux/version.h:7:2: #error Please run 'make
cloneconfig make dep' in /usr/src/linux/
/usr/src/linux/include/linux/version.h:8:2: #error to get a kernel
Hi,
I was thinking if it was possible to get this list as news ?
It would be much easier that 'hotmail-account'
/HHA
_
Scope out the new MSN Plus Internet Software optimizes dial-up to the max!
Greate - it works.
Thank you
/HHA
http://www.gmane.org offers many mailinglists as a newsfeed.
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Hi,
Anyone know how to set up tftp server for grandstream.
I gues it should be somethink like
tftpserver-dir
mac-address
firmware.bin
config.txt
Is this correct ?
And how should the config-file look like. ?
I had search sipphone.com but did'nt find anything.
/HHA
Thanks.
How is the directory structure ?
or do you add all you phone to the one file cfg.txt and have it in the root
of your tftp-dir ?
/HHA
Attached is the config file I send to my Grandstream.
Change IP address Phone ID to suite.
Thank your for the link - now I wil try it :)
/Hans-Henrik Andresen
This is the URL I got the config file from, http://www.plugndial.com/ it's
on a link from the SipPhone URL.
_
Learn how to choose, serve, and enjoy wine at Wine
Hi,
Yesterday I finaly got my handytone sip adaptor. It works
But when dialing to and from ISDN I got echo in both ends, I had tried diff.
codecs, but then the GS wont work at all - It can do a call, but after 3
'ring' it disconnect.
Any hints ?
Hi,
Are there any hardware for ISDN30 ?
if yes any problem with this ?
is i out-of-box like ISDN2 but with 30 linies ?
Do I need more than the cable from my teleprowider and a PCI-card ?
/HHA
_
Find high-speed net deals
Yes, ISDN30 ( I am assuming you are in the UK) is basically an ISDN PRI
line which is basically an E1 line, so you would need to get an E100P card
from Digium to be able to connect your ISDN30 into Asterisk..
I'm from Denmark (else my english would had been better:( )
As for the rest of the
] On Behalf Of
Hans-Henrik Andresen
Sent: January 12, 2004 05:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP-Client for Handheld PC
Anyone know a sip-client that will work on a Handheld PC
running WINCE for
HPC.
I can find some for PocketPC, but the wont work on my HPC
??
/HHA
Hi,
How much bandwidth do I need for 1 conversation ?
I know it depends on the codecs, in X-lite I can see a codec called gsm, and
the grandstream aha analog/ip converter have a codec called 721.
Doc. I have found the asterisk handbook, but only a draft from marts 2003
anything newer ?
Anyone know a sip-client that will work on a Handheld PC running WINCE for
HPC.
I can find some for PocketPC, but the wont work on my HPC
??
/HHA
_
Scope out the new MSN Plus Internet Software optimizes dial-up to the max!
Hi,
I had asterisk installed, ISDN-adapter, some x-lite software-phones and I
can call betweens the softphone- and 'normal' phones during the ISDN-card.
2 questions now
1) Is it posible to create policies, so that some SIP-users can dial ALL
numbers, and some SIP-users not are allowed to dial
Thank you both, I will start reading, and had already get something to work
:)
/Hans-Henrik Andresen
Look at contexts and the include statement. Read the draft handbook
linked from www.asterisk.org, support section. Or look here:
http://www.voip-info.org/wiki-Asterisk+howto+dial+plan
Hi,
I got this setup.
analog phone (ext7) --- analog pbx - (ext 6 analog) voicemodem (ext 3
asterisk) ttyS0/asterisk sipphones
q1:
I got the voicemodem to work, but oneway only. I can talk from my analog
phone, to my sipphone, but not the other way ? I know it only suppose to
Hans-Henrik Andresen wrote:
How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ?
The same way you recieve videos through your fax machine.. :)
HMM. greate sarcasm.
I had read about a driver for asterisk for voicemodems, that why i'm asking.
So if anyone had tried
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