Tarun Banka wrote:
Hello,
We are planning to buy following Hardware for Asterisk TestBed. Please let me know
if this seems fine to you.
1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa
2. Wildcard T100P interface card, that will connect Asterisk server to
Our
You're calling the script using EAGI not AGI - This caught me out the day.
Changing extensions.conf to use AGI solved my problem :-)
Technical explanation: Something to do with EAGI providing audio on file
descriptor 3, it confuses things. Stick with using the AGI app to call your
scripts and you
Hi,
Currently I use call files to automate the generation of calls from
our address book and the resulting call file looks like this..
Channel: SIP/201
WaitTime: 30
Application: Dial
Data: CAPI/4567:5556789
CallerID: Auto Dial 1000
This method works but it not logging the calls to the CDR and
When dialling in and dialling my extension, when answered I get
Read_channel ## vpb/1-3: Setting record mode, bridge = 0
WARNING[20499]: File chan_sip.c, Line (sip_write): Asked to
transmit frame type 8, while native formats is 4 (read/write = 4/4)
== Spawn extension (default, 1004, 1)
Hi List..
I m getting this mesg while trying to dial an extension, both SIP UAs are registered with asterisk, m trying to dial extension 1015 from UA [EMAIL PROTECTED] to extension 1016 of UA [EMAIL PROTECTED]
In extensions.conf I added
exten = 1015,1, Dial(SIP/7,20,tr)
Any hint?
JF
Hi all, FYI,
I had a similar problem where the new TDM card would show up in
/proc/pci but would not load the module.
Checked out the latest CVS zaptel, made / installed and loaded straight
away. Used Gigabyte GA-8S648 board.
Stuart.
On Fri, 2003-10-03 at 19:49, Mark Spencer wrote:
Is it
how do you go about replacing the sound files in *
with your own ??
Regards Mick
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Good afternoon,
I'm trying to use MeetMe in an AGI script written in Perl, as follows:
print EXEC MeetMe 2000|p \n;
$res = checkresult();
The problem that I have is that when a user press '#' in order to exit
from the conference, everybody goes out. This is randomized because
sometimes
Hi,
could somebody name the minimum configuration files asterisk needs to
run with a SIP phone?
what do i need apart from asterisk.conf and extensions.conf?
tia
--
Mit freundlichen Gren
Conrad Braun
Pentaprise GmbH
Im Pinderpark 5
D-90513 Zirndorf
http://www.pentaprise.de
Good afternoon,
I'm trying to use MeetMe in an AGI script written in Perl, as follows:
print EXEC MeetMe 2000|p \n;
$res = checkresult();
The problem that I have is that when a user press '#' in order to exit
from the conference, everybody goes out. This is randomized because
sometimes
Conrad Braun wrote:
Hi,
could somebody name the minimum configuration files asterisk needs to
run with a SIP phone?
what do i need apart from asterisk.conf and extensions.conf?
tia
Probably sip.conf.. :)
___
Asterisk-Users mailing list
[EMAIL
I have had a similar problem with a BT circuit, it turns out my circiut is not PRI,
but DAS, which is some sort of BT enhanced (or modified) PRI, I believe the
signalling is a bit different. My PRI premise gear has occasional lockups with it.
note this is not an asterisk setup, but a LCR
WipeOut wrote:
Conrad Braun wrote:
Hi,
could somebody name the minimum configuration files asterisk needs to
run with a SIP phone?
what do i need apart from asterisk.conf and extensions.conf?
tia
Probably sip.conf.. :)
___
Asterisk-Users mailing
Conrad Braun wrote:
ok, that's obvious. simply forgot to mention it ;)
but do I need any of the other files at all?
ps. sorry for posting an empty reply just seconds earlier...
Why do you want to remove some of the conf files?
Just leave them all there.. its not like they use up a lot of space
It appears that the T1 Digium cards can split voice and data, but I
would not want data traffic going through the * server...
Yes, they can do this. You can turn your * server into both a PBX and
router.
Do you have any documentation on how to set this up?
Also I was talking with James
I've read and experienced the echo problems with the X100P. Is Digium
going to fix the problem or refund our money? I want to see this work
because myself and other small companies out there use analog lines. I
would trade up to T1 but that requires me to have at least 9 lines. If
I did
depends how large. If you are playing with the idea of setting up something
like vonage but in a not so bandwidth loaded country you will need the
horsepower for codec stuff.
On Monday 13 October 2003 7:11 am, Chris Albertson wrote:
Do you r really need more CPU power for Asterisk? I'd think
tar
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Monday, 13 October 2003 9:54 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] replacing sound files
[EMAIL PROTECTED] wrote:
how do you go about replacing
Hi!
I am trying to do a SIP/H.323 gateway. I want that the SIP proxy server (I
suppose that this is asterisk isn't it?) has all the information about the
user's registration. So, when a request arrives at the gatekeeper from the
H.323 network, this one tries to make multicast to all the others
I am just starting to use asterisk as well as VoIP in general, and it's
a bit confusing finding out what goes where... in my eyes it seems to be
a lot easier to start with a bare minimum, thereby eliminating as many
causes for error as possible. when I feel comfortable, I can always
expand on
On Monday, October 13, 2003 7:21 AM, Andrew Kohlsmith
[SMTP:[EMAIL PROTECTED] wrote:
Also I was talking with James (different James, hahaha) about using a
T400P
to take in a PRI from the telco and provide a PRI (or CAS T1) to an
old
access server, routing modem calls to the access server by
Michael Bielicki wrote:
depends how large. If you are playing with the idea of setting up something
like vonage but in a not so bandwidth loaded country you will need the
horsepower for codec stuff.
On Monday 13 October 2003 7:11 am, Chris Albertson wrote:
Do you r really need more CPU
Does anyone know if there is a dialplan in place that would allow
me to dial out via iaxtel (with a 700 number) and back into my
fwd number?
I've tested fine in the opposite direction, but would like to verify
the fwd incoming call success.
Rich
___
Florian Overkamp schrieb:
Hey, if I press Flash asterisk gets the 'hf' event but does nothing.
What gives ? :-)
We can compare our ATA-configs, because transfering works fine with MGCP
(SIP doesnt).
By the way, I'd think maybe it's not actually transferring but rather
'bridging' through the
Yes, my Asterisk is behind a NAT but I
forward all ports (100-56000) to my Linux box.
Chris HARIGA
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Uriel Carrasquilla
Sent: Monday, October 13, 2003 12:18 AM
To:
[EMAIL PROTECTED]
Subject:
Hi!
I need to open both ports 1720 and 1719. How can I do that?
Thanks.
Regards,
Mireia
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
On 13/10/03 14:05, Conrad Braun wrote:
Why do you want to remove some of the conf files? Just leave them
all there.. its not like they use up a lot of space or anything..
:)
I am just starting to use asterisk as well as VoIP in general, and
it's a bit confusing finding out what goes where... in
You will need :
extensions.conf
indications.conf
logger.conf
manager.conf
rtp.conf
sip.conf
modules.conf ; with a crap load of stuff turned off:
noload = chan_modem.so
noload = chan_modem_aopen.so
noload = chan_modem_bestdata.so
noload = chan_modem_i4l.so
noload = chan_phone.so
noload =
Conrad Braun wrote:
I am just starting to use asterisk as well as VoIP in general, and
it's a bit confusing finding out what goes where... in my eyes it
seems to be a lot easier to start with a bare minimum, thereby
eliminating as many causes for error as possible. when I feel
comfortable, I
Alastair Maw wrote:
On 13/10/03 14:05, Conrad Braun wrote:
Why do you want to remove some of the conf files? Just leave them
all there.. its not like they use up a lot of space or anything..
:)
I am just starting to use asterisk as well as VoIP in general, and
it's a bit confusing finding out
Does anyone know if there is a dialplan in place that would allow
me to dial out via iaxtel (with a 700 number) and back into my
fwd number?
I've tested fine in the opposite direction, but would like to verify
the fwd incoming call success.
Rich
700-99-X where X is the 5 digit
Does anyone know if there is a dialplan in place that would allow
me to dial out via iaxtel (with a 700 number) and back into my
fwd number?
I've tested fine in the opposite direction, but would like to verify
the fwd incoming call success.
Rich
700-99-X where X is the
Hi!
I have configured my gatekeeper to call asterisk everytime that the phone number
begins with 064... When the gatekeeper contacts asterisk, it does it using the
1719 port, but it is closed.
How can I open this port? Or the solution is to redirect the messages arriving
at 1719 to 1720?
Translation: Asked to transmit frame type G.711 A-law, while native
formats is G.711 u-law (read/write =G.711 u-law/G.711 u-law)
Looks to me like you need a disallow=all in your sip.conf and allow=
lines for the codecs you want to allow, then make sure that the IP
phones you are using support at
Hi All.
Followed the information from the link bellow and can now see the card.
But.
When I run modprobe zaptel I get the message that the zaptel.o was
compiled for kernel version 2.4.20-4GB while this kernel version is
2.4.20-4GB-athlon. And fails.
When I run modprobe wcfxo I get the message
Remove the space before Dial
On Mon, 2003-10-13 at 05:27, John Foster wrote:
Hi List..
I m getting this mesg while trying to dial an extension, both SIP UAs
are registered with asterisk, m trying to dial extension 1015 from UA
[EMAIL PROTECTED] to extension 1016 of UA [EMAIL PROTECTED]
Try http://www.fnords.org/~eric/asterisk/ It contains simplified config
files as well as other information.
On Mon, 2003-10-13 at 06:34, Conrad Braun wrote:
Hi,
could somebody name the minimum configuration files asterisk needs to
run with a SIP phone?
what do i need apart from
Here is an example call (works) :
-- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack
-- Called g1/0707038340
-- Zap/1-1 is ringing
!! Unknown IE 76 (Unknown Information Element)
-- Zap/1-1 answered SIP/25-e804
What does that !! Unknown IE 76 (Unknown Information
Will M$ ever stop!!.. Whats the bet their telecoms products will use
non-standard protocols..
I really wouldn't like to run a telecom system on Windoze in the first
place..
Full Story..
http://news.zdnet.co.uk/communications/0,39020336,39117099,00.htm
Hi Marcel,
IE 76 is COLP (Connected Line ID Presentation).
Your telco is so kind to tell you to which number your calls has
been connected. Noting to worry about...
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30
It means that this IE is not implemented in the libpri or is not very
standarized.
regards
Martin
On Mon, 13 Oct 2003, Marcel Prisi wrote:
Here is an example call (works) :
-- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack
-- Called g1/0707038340
-- Zap/1-1 is
put a comma after "Dial"
- Original Message -
From:
John
Foster
To: [EMAIL PROTECTED]
Sent: Monday, October 13, 2003 5:27
AM
Subject: [Asterisk-Users] Extension
Dialing problem with SIP
Hi List..
I m getting this mesg while trying to dial an
Thanks Rich,
I am re-installing the base SuSE Linux system again and will try to install
everything without doing any updates. I can't remember any updates being
done, but these automated installs for numpties like me could do anything
and I wouldn't know.
I will let you know how it goes.
Hi Martin,
it's not implemented in libpri but very well standarized (ETS 300 097).
regards,
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email: [EMAIL PROTECTED]
My fault then :)
I was thinking only in terms of Q931 spec ...
Martin
On 13 Oct 2003, Klaus-Peter Junghanns wrote:
Hi Martin,
it's not implemented in libpri but very well standarized (ETS 300 097).
regards,
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse
This is bull... I can't believe that...
Must be a solution...
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: Monday, October 13, 2003 9:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] No sound with SIP Phones on the
This is bull... I can't believe that...
Must be a solution...
sip is very tricky to get working behind firewalls. sip clients work quite
well with nat, just make sure nat=yes is in the sip profile in sip.conf
my solution has always been to put an asterisk box behind the firewall and
make all
Chris Hariga wrote:
This is bull... I can't believe that...
Must be a solution...
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: Monday, October 13, 2003 9:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] No sound with
Klaus-Peter Junghanns wrote:
Hi Martin,
libpri misses all the fun stuff :-(
hold, retrieve, suspend, ect, cd, conf, 3pty ..
but i am going to change that :-)
regards
kapejod
It misses all the timers, too. :-)
Regards,
Steve
___
Asterisk-Users
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0
Hi all,
inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is
Debian woody. * is the newest cvs co.
I have written a little callgen script which make outgoing calls through my
*:
#! /bin/sh
set -e
n=$1# Nummer
anz=$2 # Anzhal der Versuche
anz2=$3
Are you using the recommended pwlib and openh323 tarballs?
bkw
On Mon, 13 Oct 2003, CW_ASN wrote:
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used
Hello,
Here in our office we are testing
Asterisk.
My collage Igor created to Asterisk PrePaid
applicationwith Postgresql.
It is not in Perl.
We would like to release it to the group
as soon as it will work ok.
It will have authentication, different rates for
users, different rates for
Title: Message
I'd
like to see it. What language is it in? I'm sure everyone in the group could
benefit in some form
--James Cornman [EMAIL PROTECTED]Completely
Reliable Network Conceptshttp://www.crnc.net(v) 973-784-0031(f)
973-784-0038
-Original Message-From:
[EMAIL
In what language is it written in? It
would be interesting to at least look at it and maybe convert it to use MySQL
instead
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Bartosz Jozwiak
Sent: Monday, October 13, 2003
3:49 PM
To: ASTERISK
Hello I am still having problems with IAXTELL and FWD configuration.
I get the following when I dial 17009965342 which is set as an example
to dial to FWD people. 1+700+99+ 5 digit number. I have placed X
where my passwords are.
my IAX.conf has
register = abatista:[EMAIL
From: Brian West [EMAIL PROTECTED]
register = abatista:[EMAIL PROTECTED]/114 doesn't work in iax.conf
also you are sending the full 917009965342
you should only send ${EXTEN:1} strip that 9 off.
OK done I forgot about the stripping the 9 off. Now I can call the numbers, But now
how do I get
Ok
I have tried to post to this list server but have just gotten the automated
reply saying the moderator has to approve it to the list first which was my
mistake for sending from the wrong email account.
So
if the moderator finally approves my questions and you see the same post again
Bartosz Jozwiak wrote:
Hello,
Here in our office we are testing Asterisk.
My collage Igor created to Asterisk PrePaid application with Postgresql.
It is not in Perl.
We would like to release it to the group as soon as it will work ok.
It will have authentication, different rates for users,
UnixODBC. No need to rewrite everything for a simple DB change.
In what language is it written in? It would be interesting to at least
look at it and maybe convert it to use MySQL instead.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
-- Original Message --
From: Rich Adamson [EMAIL PROTECTED]
OK I got it working. Thank you Rich I used your examples and had to add the following
the sip.conf file. It did not work until I had the :5060 on it!
register = 65342:[EMAIL PROTECTED]:5060
Yes, I donwload tgz's from nufone (http://www.nufone.net/downloads/). All
sources was compiled as Jeremy recommeds, and I didn't have troubles with
that. Oh, I'm using RH9.
This is my h323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=alaw
allow=gsm
dtmfmode=rfc2833
Hello,
I'm considering paying Digium to do a modification to Asterisk so that calls
can be parked on specific user-defined numbers(transfer to 701 and it's
parked on 701, transfer to 702 and it's parked on 702) instead of the way
Asterisk currently does call parking(transfer to 700 and then it
It has been written in the same language as other applications like
Dial,Queue,Record and so on.
I hope that our company will say YES say we can release it.
-- Bart
-
Bartosz Jozwiak wrote:
Hello,
Here in our office we are testing Asterisk.
My collage Igor
Is there an underlying reason you want to do this? Because if a call is
already parked on 701 and you transfer another call to 701 to park it,
both callers would be connected.
I am sure there is a better way to implement what you want.
-Original Message-
From: [EMAIL PROTECTED]
On Mon, 13 Oct 2003 14:01:00 -0400, Andrew Kohlsmith wrote:
I really wouldn't like to run a telecom system on Windoze in the first
place..
One of the Meridian agent systems uses OS/2 on their system... :-)
mmm, thanks for reminding me, i still have one system running OS/2. I
hadn't looked
Bartosz Jozwiak wrote:
Hello,
Here in our office we are testing Asterisk.
My collage Igor created to Asterisk PrePaid application with Postgresql.
It is not in Perl.
We would like to release it to the group as soon as it will work ok.
It will have authentication, different rates for
Is there an underlying reason you want to do this? Because if a call is
already parked on 701 and you transfer another call to 701 to park it,
both callers would be connected.
Actually I have to agree with Matt; I would like to be able to specify where
it's parked and get a busy if I try to
I'm curently looking into using SER to front end SIP calls for
Asterisk.
Basicaly all SIP users would register with SER not Asterisk and then
Asterisk and SER exchange registrations.
SER is a very capable SIP router, much more sophisticated than Asterisk
as it can look inside packets and route
Most PBX do park the way your old KSU system did.
As a matter of fact Asterisk is the only PBX I have ever seen
that parks the way it does.
If given a choice my uses would use the normal way. And I would
be happy not to here the question can you speed up her talking? LOL
Andrew Kohlsmith wrote:
That is how many old PBX phone systems work and it is that way our users are
used to working with the phone system. Another issue with the way Asterisk
callparking currently works is that there is only one call-park orbit, you
cannot use a different set of numbers for a different call park
I'm curently looking into using SER to front end SIP calls for
Asterisk.
Basicaly all SIP users would register with SER not Asterisk and then
Asterisk and SER exchange registrations.
SER is a very capable SIP router, much more sophisticated than Asterisk
as it can look inside packets and route
Hi Thomas-
I didn't look closely at your shell script, but I wrote something similar in
Perl (and used shell to start each instance of it). I had a few problems
too with a similar setup (although no machine lockups)
* You are running quite a slow machine to run this script on many lines at
On 13-10 17:11, John Todd wrote:
[...]
SER is an excellent option as a front end to Asterisk. It is a
true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been
the primary focus of Asterisk development. In fact, Asterisk's SIP
implementation is very limited (though it is
I just tested fromstring and emailbody with voicemail2 and a farily new
code and it's working. I don't know what you're doing wrong ... but
something for sure.
regards
Martin
On Mon, 13 Oct 2003, John Todd wrote:
I would recommend then doing grep fromstring
John:
I have
been around voice over data packets for quite a few years and I am still to see
the perfect system that works identical to circuit switching 100% of the
time. My opinion is that there is a lot more to the story than just
parameters. Packet loses, double compressions, faulty
Dunca:
I am not sure I understand your statemnet.
SIP devices (UA) on the other side of the Internet behnid a NAT communicate
to * on the public Internet. Then this Asterisk connects to other Asterisks
(via IAX) that can be behind Firwalls (or NATS). am I understanding
correctly?
Uriel
Chris:
I am glad to see someone else asking the same question I have been asking
myself.
As soon as I get my public IP address, I will install SER on the public side
and Asterisk behind a NAT (with dynamic IP) to see if I can get around
problems I have when my SIP (UA) behind their own NAT on the
John:
are you aware of any documentation on how to configre SER to be a front-end
to Asterisk?
I suspect it is very inexpensive to put a SER server in a hosting facility
to forward traffic to multiple Asterisks based on Least Cost Routing.
My problem is that my experience is with Asterisk and not
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: Monday, October 13, 2003 8:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)
I'm curently looking into using SER to front end SIP
I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the
source code, mailing lists and other resources. Here's the scenario and the approach
I have been pursuing. I'm having some problems with the AGI calls and I hope someone
can give me some clarification.
I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the
source code, mailing lists and other resources. Here's the scenario and the approach
I have been pursuing. I'm having some problems with the AGI calls and I hope someone
can give me some clarification.
Hello,
Is there any way to pass an H323 ID (resembles a sip [EMAIL PROTECTED]) to an
h323 gateway? Thank you in advance for your suggestions!
Regards,
Christopher J. Wolff, VP, CIO
Broadband Laboratories
http://www.bblabs.com
___
Asterisk-Users
However, the timezone is still not straight in the message body.
${VM_DATE} doesn't seem to use the timezone matching routines defined
by the user's tz= setting.
Well it's the task for those who add features to have a global-system
thinking. The emailbody was added way before the timezones ...
Has anyone gotten 3 way calling to work? There seems to be
no way to swap to the other call and sometimes the unit will generate the call
waiting tone ever second. It also seems that if you try to flash the call and
then hang up you have to pick up the phone, flash back to the first call
On Monday 13 October 2003 22:18, Martin Pycko wrote:
However, the timezone is still not straight in the message body.
${VM_DATE} doesn't seem to use the timezone matching routines
defined by the user's tz= setting.
Well it's the task for those who add features to have a global-system
On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote:
John:
are you aware of any documentation on how to configre SER to be a front-end
to Asterisk?
Hi Uriel,
At TeleSIP we run a cluster of several geographically distributed SER Servers
that hande all our SIP Routing. SER is a robust,
Hi List,
After going through mailing list and manual of asterisk, I still could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params
Log of real session:
[EMAIL PROTECTED] root]# telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.0
action: login
username: joe
secret: bob
Response: Success
Message: Authentication accepted
action: originate
exten: 200
context:
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