-
[general]
autofallthrough=yes
context=default
[default]
;exten = _.,1,Dial(SIP/[EMAIL PROTECTED],70,t)
exten = _.,1,AGI(mta_auth.agi,${EXTEN})
exten = _.,2,Hangup
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Software Engineer
[EMAIL PROTECTED]
Heritage Communications Corporation
Melbourne, FL USA 32935
?
Thanks for all of your help,
AZM
The Labs
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autocreatepeer. Thanks -
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I have Obsequium running and have developed a way to parse the .PLS files
that it returns. Is there a way in Asterisk to play rtp:// streams as MOH?
Thanks.
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Software Engineer
[EMAIL PROTECTED]
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Melbourne, FL USA 32935
: HCC Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 209.114.219.98:5060
-- AGI Script HCC_TEST.agi completed, returning 0
Destroying call '[EMAIL PROTECTED]'
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command. I
am not interested in realtime because I am running stable 1.0.3 only and also
it does not provide the qualify functionality I so desire.
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http
NAT translation
and forward off packets to the Asterisk server accordingly.
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?
Cheers.
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is finished I would like to have the balance
shown in the display by sending a sip message to the phone(if
possible otherwise not important).
This would require adding code to the AGI, if it's even possible.
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Software Engineer
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Rogerio
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]: loader.c:423 load_modules: Loading module
chan_oh323.so failed!
So, I am wondering what is wrong and whether the packages I have built are
compatible. Any help on this is greatly appreciated.
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and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
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On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I
the 0.6.2a version.
Michael.
Brian Wilkins wrote:
Hi,
I having a problem compiling the wrapper for oh323. I am running
Debian, kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6
and the openh323 version I have is 1.13.5. I execute the following
commands first before
.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you
were able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with
asterisk-oh323 0.6.2. I get the following errors when trying to compile
the oh323 wrapper for asterisk:
-- snippet of errors
current
Asterisk CVS code.
Michael.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you
were able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with
asterisk-oh323 0.6.2. I get the following errors when trying to compile
the oh323 wrapper
Hi,
I am using the asterisk-oh323 wrapper and I am looking to allow
registration of h323 endpoints and allow Asterisk to act as a gateway. The
idea is simple: H323 endpoints would register with Asterisk. They each would
have their own internal extension (like SIP). If a H323 endpoint dials
Thanks for the example. But my question is how does Asterisk know where to
send data if there doesn't seem to be a facility for people to register
their IP to an extension?
On Yaum al-Arbi'a 12 Jumaada al-Awal 1425 04:41 am, administrator tootai
wrote:
Brian Wilkins a écrit :
Hi,
I am
Hi,
I get this error when trying to dial an outbound extension from a sip
phone:
-- snip --
-- Executing Dial(SIP/2003-02d1, OH323/[EMAIL PROTECTED]|20) in new stack
-- H.323 call to [EMAIL PROTECTED] with codec ALAW
-- Called [EMAIL PROTECTED]
0:33.283 H225
Hi,
We are having an issue here. It seems that whenever we initialize Asterisk
on our network, the router that the Asterisk server is connected to (Cisco
7200) crashes and loses it configuration. This has happended five times and
each time we have tested it, it is always when Asterisk
IOS version 12.xx
As far as a traceback, that's going to be difficult now since we've removed it
from our switch and brought it back here to the office for testing. When we
test it tomorrow or later in the week, I'll see if it crashes again in a test
setting and try to get a traceback to the
.
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Brian Wilkins [EMAIL PROTECTED]
To: Asterisk-users [EMAIL PROTECTED]
Sent: Wednesday, June 30, 2004 11:05 AM
Subject: [Asterisk-Users] Asterisk Causing Cisco 7200 Router to Crash?
Hi,
We are having an issue
It's also involved in VoIP as well.
On Yaum al-Arbi'a 12 Jumaada al-Awal 1425 05:34 pm, Scott Laird wrote:
On Jun 30, 2004, at 2:31 PM, Brian Wilkins wrote:
IOS version 12.xx
As far as a traceback, that's going to be difficult now since we've
removed it
from our switch and brought
A traceback is not possible. The best thing I can show everyone is the reboot
message. The logs got obliterated when the Asterisk server started up and the
best we can imagine, sent an invalid code to the router. We are going to
set up a small test subnet here and bounce around on the router to
Doubt it, it has been behind a firewall with the router(s) the whole time.
Could it be that your * server is compromised and is pounding on the
72xx? Depending on the version of IOS that you're running, this is
fairly trivial thing to do.
John
--
Heritage Communications Corporation
If your fiber card is no longer functioning, there is more to this
issue than some malformed packets... is your router on a UPS? Is
the cable run to the router over 100m/300'?
Nick
The cable is shorter than 300' and it's not a power overload/underload
problem. We have load balanced our
You can only use g729 in pass-thru mode without paying for the licensing fees.
G729 is probably the best codec around. If you plan on having any sort of
thriving business based on VoIP, g729 would be the way to go. I don't suggest
PCMU or PCMA for production. The ATA will pass a list of
channel should i consider $10 for H323 channel $10 for
the SIP channel (for instance), or is it $10 per number of concurrent calls
wanted regardless of the protocols used?
Thanks,
Kido
- Original Message -
From: Brian Wilkins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
G729 sounds better than a cell phone to me. There is no noticable difference,
the way we use it here, between Asterisk and a regular phone call.
On Monday 15 November 2004 04:43 pm, Julio Arruda wrote:
Brian Wilkins wrote:
You can only use g729 in pass-thru mode without paying
visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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::easy::cleanup($curl);
}
sub body_callb {
my($chunk,$handle)[EMAIL PROTECTED];
${handle} .= $chunk;
$rawdata .= $chunk;
print STDERR $chunk;
return length($chunk);
}
sub header_callb {
return length($_[0]);
}
-- end script --
--
Brian Wilkins
SPA-3000
I have dtmfmode=rfc2833 in my sip.conf
I've tried to SPA-PSTN Gain but it makes no difference.
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Brian Wilkins
Software Engineer
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Melbourne, FL USA 32935
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.
On Thursday 18 November 2004 07:56 pm, Joseph wrote:
On Thu, 2004-11-18 at 13:52 +, Brian Wilkins wrote:
The same exact thing happens to me. For me, it is because the calls are
going out via Confer and that has problems sending DTMF tones. I'd
recheck the route that you sending calls out to.
How
);
}
sub body_callb {
my($chunk,$handle)[EMAIL PROTECTED];
${handle} .= $chunk;
$rawdata .= $chunk;
print STDERR $chunk;
return length($chunk);
}
sub header_callb {
return length($_[0]);
}
-- end code --
On Thursday 18 November 2004 11:51 am, Brian Wilkins
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Software
or update options visit:
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Software Engineer
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Melbourne, FL USA 32935
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right? I don't think the Cisco phones support STUN.
-Matthew
- Original Message -
From: Brian Wilkins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, November 23, 2004 4:25 AM
Subject: Re: [Asterisk-Users] SER is a better
[EMAIL PROTECTED]
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Software Engineer
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Heritage Communications Corporation
Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net
Is it possible to execute a script upon successful registration and
authentication of a SIP device in Asterisk? For instance, have a script log
all successful registrations in a database or authenticate users instead of
using the secret=password in the sip.conf file? Thanks -
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+cisco+FXO
On Tuesday 30 November 2004 02:46 pm, Sebastian Nocetti wrote:
I think you CAN'T DO VOIP-VOIP into CISCO Equipment, it have to be
POTS-VOIP or viceversa.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Brian Wilkins
Enviado el: Martes, 30 de
/mailman/listinfo/asterisk-users
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[EMAIL PROTECTED]
http
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devices usually
pass a list of supported codecs to the SIP provider along with the SIP
request. Also, if you have g729 licenses and all else fails: Asterisk should
be able to transcode between various codecs anyways.
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Brian Wilkins
Software Engineer
[EMAIL PROTECTED]
Heritage Communications
scale IP PBX.
Read here on how to convert your MyISAM tables to InnoDB :
http://dev.mysql.com/doc/mysql/en/Converting_tables_to_InnoDB.html
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Melbourne, FL USA 32935
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http
/mailman/listinfo/asterisk-users
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[EMAIL PROTECTED]
http
PROTECTED] on a i686
running Linux
*CLI
Any help would be appreciated. Thanks!
Clay Reiche
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Heritage Communications Corporation
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http://www.hcc.net
| | NULL||
++---+--+-+-++
36 rows in set (0.01 sec)
--
Brian Wilkins
Software Engineer
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Heritage Communications Corporation
Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net
is
17-12-2004 05:55:22.
On Friday 17 December 2004 03:57 pm, Matthew Boehm wrote:
Post this as a bug Brian.
-Matthew
- Original Message -
From: Brian Wilkins [EMAIL PROTECTED]
To: Asterisk-users [EMAIL PROTECTED]
Sent: Tuesday, December 14, 2004 3:51 AM
Subject: [Asterisk
All,
Enjoy these free prompts as an addition to your sounds collection. I hope
you find them useful. You can find them attached to this message.
account-balance-is.gsm
Description: Binary data
lunch.gsm
Description: Binary data
to-hear-your-account-balance.gsm
Description: Binary
of curiosity, why?
-tih
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[EMAIL
I think you mean SIP Gateway. You can forward the SIP off to a SIP Provider by
specifiying it in your sip.conf file as :
[mysipprovider-out]
type=peer
secret=password
username=2345
host=something.hcc.net
fromuser=2345
nat=no
then in your extensions.conf file:
i.e.
exten =
I believe retrieving in real-time is being worked on and should be done soon.
Developers are almost finished working on RealTime.
include = sip_additional.conf in [general]
On Tuesday 12 October 2004 05:26 pm, harry gaillac wrote:
hello Matthew,
I was wrong -:) but retrieving all sip info
The perl script will overwrite the existing conf file. I've had bad
experiences with constant reloading. Maybe you want to schedule your updates
through a crontab.
On Wednesday 13 October 2004 03:28 pm, harry gaillac wrote:
ok but if i add or remove variables from database.
Does the perl
I've had bad experiences with reloads of thousands of extensions several times
throughout the day. I've spoken with others at AstriCon about the same issue,
fyi
On Wednesday 13 October 2004 04:35 pm, Brian West wrote:
The perl script will overwrite the existing conf file. I've had bad
] On Behalf Of Brian Wilkins
Sent: Wednesday, October 13, 2004 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP peers in MySQL Database
I've had bad experiences with reloads of thousands of extensions several
times
throughout the day. I've
We do the same thing here. The customer has a say a starting balance of
$100.00 and they are allowed to use it up. If it goes into the negative, then
they are still allowed to place calls. When it comes time to invoice them, we
invoice them for the amount and they pay. When they pay, we
Hi,
I have a machine that does SMP (Symmetric Multi-Processing) and I was
wondering if it would be a problem if I used a kernel that used SMP with
Asterisk? Would it crash? Thanks -
--
--
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Melbourne, FL USA 32935
http://www.hcc.net
Thanks all for the input - much appreciated.
On Friday 15 October 2004 08:18 pm, [EMAIL PROTECTED] wrote:
On Oct 15, 2004, at 11:20 AM, Brian Wilkins wrote:
Hi,
I have a machine that does SMP (Symmetric Multi-Processing) and I
was
wondering if it would be a problem if I used a kernel
MeetMe though..we need to talk.
Matthew
- Original Message -
From: Brian Wilkins [EMAIL PROTECTED]
To: Asterisk-users [EMAIL PROTECTED]
Sent: Friday, October 15, 2004 11:20 AM
Subject: [Asterisk-Users] Asterisk and SMP
Hi,
I have a machine that does SMP (Symmetric Multi
Hi,
We have a system that takes an e164 number and does a translation to
convert that e164 number to the PIN that is tied to a certain account. We
configure customer's devices to use those PINs so they don't ever have to use
a PIN. They just plug the device into the internet, and it works.
I get these warnings when I reload my config through the console:
Nov 5 04:31:10 WARNING[1301281712]: pbx.c:1938 ast_pbx_run: Channel
'SIP/6822170331-5364' sent into invalid extension '321235689' in context
'default', but no invalid handler
Nov 5 04:31:10 WARNING[1309670320]: pbx.c:1938
You could pass the pound sign as a PLAR (Private Line Automatic Ringdown)
code, with say a PIN number after that. The PLAR code is also called
Automatic Dial when Off-Hook.
On Sunday 07 November 2004 12:08 pm, Mike Roberts wrote:
I have this.
exten = 8,1,ANSWER
exten = 8,2,DigitTimeout,5
I had the same problem with XLite a couple days ago. I couldn't find a
workaround, but every so often the XLite client would pass the * and # and
sometimes it would send it as a %23. It's possible to send * and # through
XLite to Asterisk, but it occasionally works.
On Wednesday 10 November
404 not found can mean many things, are you using a supporting codec?
On Wednesday 10 November 2004 05:25 pm, Ashling O'Driscoll wrote:
Hi,
Hope somebody can help. I have two xlite clients that register with
asterisk. They are called 2000 and 2001.
1)When 2000 rings 2001 a '404 not found'
Hi,
I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular
phone. Asterisk configuration is listed below. When I attempt to place a
H.323 call, I receive the following errors:
- Executing Dial(SIP/2000-3029, OH323/[EMAIL PROTECTED]:1720|20)
in new stack
Aug 13 09:13:03
13 August 2004 08:10 pm, administrator tootai wrote:
Brian Wilkins a écrit :
Hi,
I am using the Grandstream HandyTone 486 as a SIP Adapter with a
regular phone. Asterisk configuration is listed below. When I attempt to
place a H.323 call, I receive the following errors:
- Executing
I had that problem, but apt-get install did the trick.
On Tuesday 31 August 2004 02:53 pm, Huddleston, Robert wrote:
I've been having troubles compiling in the openh323 on both redhat and
debian... one of the biggest problems I had w/ Debian is it couldn't find
alot of libraries like termcap
It sounds like Skinny couldn't load- doesn't sound fatal.
On Thursday 09 September 2004 08:01 pm, [EMAIL PROTECTED] wrote:
Greetings,
I've just installed Asterisk RC2 on a new RedHat 9 box (fully patched).
The build and install went fine, but after starting Asterisk, I get the
following
Quick question:
Are the hotel rooms registered automatically upon receipt of payment for
AstriCon plus hotel room costs ? Or do we have to register manually by
calling the hotel? Thanks -
On Monday 30 August 2004 10:07 pm, Steven Sokol wrote:
Just a brief reminder to everyone who wishes to
/listinfo/asterisk-users
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Here is the information on doing SIP to H323 :
http://lists.digium.com/pipermail/asterisk-users/2004-July/056425.html
On Wednesday 15 September 2004 08:43 pm, Carlos Maynard wrote:
Hello
Asterisk is compiled and running perfectly...
But when i try to compile Channel_h323... that's
If it's what Andrew is talking about, then add the hostname to /etc/hosts.
On Thursday 16 September 2004 05:27 pm, Andrew Thompson wrote:
Rodolfo Grave wrote:
Hi.
I cant make SIP calls from asterisk.
When I start asterisk, I get the following message: What does it means??
Asterisk is
You could have it go to a seperate context, or setup different extensions.
Here's a good link : http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu
- Brian
On Thursday 16 September 2004 07:09 pm, Bartosz Wegrzyn wrote:
Hi,
How can I create a submenus in extensions.conf.
For example:
1
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