I already had spfsoftfail set to on, but have added the domain to
blockstricgSPFRE now.
I think my confusion is - shouldn't ASSP be catching the obviously forged
source domain not matching usps.com ?
And since I already had spfsoftfail on, shouldn't assp have prepended the fail
warning to
domain
Why is ASSP not throwing out this message based on the obviously faked
sender domain/ip
possibly because you use an outdated version ? I think there was fix for
this in the last months.
Thomas
Von:Michelle Dupuis mdup...@ocg.ca
An: assp-test@lists.sourceforge.net
assp-test
The scam of the day seems to be fake billing notices from USPS. I have
usps.com added to the strict SPF list. Still they are getting through. I'm
slowly getting ASSP setup right, but I'm still missing something.
I ran the header through the ASSP analyzer, and it didn't find a reason to
My users are getting flooded with fake linkedin messages. I ran the message
and header through the analyzer and got the info below. Can someone advise on
what I need to adjust to trap these?
Some notes, netdorm.com is a mail host I use to buffer incoming mail (so I
trust it) - but there is
(* apologizes - I sent same message to assp user list, but I'm using v2 so
should have been sent here *)
I'm setting up ASSP in an environment with an SMTP gateway at the network edge.
So, all mail must flow through that SMTP gateway (relay) before hitting ASSP.
This messes up a lot of ASSP
I'm setting up ASSP in an environment with an SMTP gateway at the network edge.
So, all mail must flow through that SMTP gateway (relay) before hitting ASSP.
This messes up a lot of ASSP features like SPF, since everything is arriving at
ASSP from an approved SMTP host, but prior hosts may
We have a multitenant Asterisk 1.4 installation for multiple small business,
and we need to report how many calls a single business has active at one time.
Is there a way to VIEW how many calls are up in a single context? (Or some
other way to accomplish the same)?
Thanks
--
Wow - nice! A few quick questions:
1. How long can the recording be for translation?
2. Any limitation on how much text the return (transcribed) variable can hold?
3. Any commercial / terms of use limitations?
From: asterisk-users-boun...@lists.digium.com
1. I checked the log and I don't see any registration attempt, so I *assume*
they simply send an invite, and so they are in the external/outside context of
my dialplan. So they are trying to reach extensions which don't exist. If
they succesfully registered they would be on the internal
...@lists.digium.com] On Behalf Of Andrew Furey
[andrew.fu...@gmail.com]
Sent: Wednesday, December 28, 2011 11:37 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Interesting attack tonight fail2ban them
On 29 December 2011 12:07, Michelle Dupuis mdup...@ocg.ca wrote:
I thought that it might be worth
Here is more of a SIP debug log:
As you can see Asterisk retries four times but I assume the softphone is not
responding?
---
Really destroying SIP dialog
'637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4'mailto:'637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4'
Method: OPTIONS
Reliably
The BB is using wifi, on the same subnet as the asterisk server so no need for
NAT.
There is no keep alive option on the softphone (very simplistic settings)
Thanks
--
_
-- Bandwidth and Colocation Provided by
I happened to be in the cli tonight as some (208.122.57.58) initiated a simple
attack - just trying to make long distance calls from outside context.
Although harmless, this went on for several minutes as the idiot just used up
my bandwidth with SIP messages. Here's and example:
[2011-12-28
On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
I happened to be in the cli tonight as some (208.122.57.58) initiated a simple
attack - just trying to make long distance calls from outside context.
Although harmless, this went on for several minutes
I have a softphone I'm trying on a blackberry, that registers on my Asterisk,
can make outgoing calls, but can't receive calls.
There is very little traffic with this phone (see debug below - as the phone
registers), and sip show peers confirms it is unreachable.
Any suggestions? Is this just
There is a script on www.generationd.com designed for Asterisk. It will
convert the Wav49 to mp3, add call info into MP3 tags, add a company logo, etc.
and then email the message.
It's a one line change to add to asterisk - very handy. (We use it for Android
phones, nice to see call info
] On Behalf Of jon pounder
[j...@inline.net]
Sent: Friday, November 25, 2011 8:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] android won't play wav49: how to change format
On 11/25/2011 06:39 PM, Michelle Dupuis wrote:
There is a script on www.generationd.com designed for Asterisk
Although you say SIMPLE...not all virtualization hosts allow software
installation. On VMware the host has become an appliance you can't really mess
with...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On
VMware is moving all server products to their ESXi engine. (The old VMware
server and ESX products are moving to legacy status - with these you could
actually do stuff on the kernel). ESXi is no longer a kernel you can mess
with, can't install drivers, etc. ESXi is being treated as an
I'm switching my MSDOS usb key to FREEDOS now, and my fancy menu system (buit
in config.sys) doesn't work under freedos (just scrolls right past during
bootup).
Is the menu feature not available in the freedos version of config.sys? Or
does it work differently?
Thanks
, one
after the other)...
Thanks
From: Michelle Dupuis [mdup...@ocg.ca]
Sent: Thursday, October 13, 2011 1:16 PM
To: freedos-user@lists.sourceforge.net
Subject: [Freedos-user] Menu in config.sys not working
I'm switching my MSDOS usb key to FREEDOS now, and my
and if I can add a question 4 to the list:
4. Can I load device drivers from autoexec.bat? (essentially moving my menus
from config.sys to autoexec.bat)
Thanks
From: Michelle Dupuis [mdup...@ocg.ca]
Sent: Thursday, October 13, 2011 1:28 PM
To: freedos-user
Hi all - I'm new to freedos, but pretty good at DOS linux. Hopefully someone
can help...
I have a USB key with lots of OS versions on it for repair. I decided to tree
FREEDOS on a parition, and although I got it booting, on bootup I get a
screenfull of Warning: using suspect partition
If one server is supposed to carry the full load of the other during failure,
then you have to size each server to handle 100% load - so load balancing is
pointless.
Checkout haast at www.generationd.comhttp://www.generationd.com and read the
docs on how it does failover...certainly good for
Has anyone written a C wrapper to ease development with the AMI? I found a
couple of c++ ones, but not C.
Thanks!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
We are building an app that will initiate outbound calls using .call files, and
each call can be a different duration (eg: 1min to 5min). These calls will go
through an Asterisk service with other calls/apps running.
I need to control the MAX number of channels in use so I don't overload this
We ran into this a few years ago. Polycoms and Grandstreams worked fine with
#xxx extensions, but Aastra's would not. Could not dial extensions beginning
with #
We chased Aastra tech support for 2 weeks. They acknowledge the bug, and we
were told they would fix this in their next firmware
If you check the archives you might find the original messages on this topic
from a few years ago...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
[davies...@gmail.com]
Sent: Wednesday,
Cool topic!
Our company (generationD) developed some CID scripts for free use, and we would
be interested in building and hosting this service.
On the spec side, how do we avoid users claiming numbers belonging to others?
(Could be an admin nightmare)
Do we allow number ranges?
Do we require
I'm looking for recommendations for standalond PRI to SIP converters. (Needs
to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy features...
Thanks!
--
_
--
We have an MTA on the edge through which all mail flows (and ASSP is behind it).
Is there a way for Assp to strip off the last MTA IP/Hostname for bad host
checks, etc?
(I posted this a month ago to the -test list but got no response...sorry to
bump)
I upgraded from 1.2.x to version 2 of ASSP. And WOW are there a lot more
settings (not wow in a good way). I'm trying to tune ASSP to behave as well as
my old ver 1.x but I'm missing info. In particular, many messages have hardle
any x-assp-* in the header (see example below). Why? Some
I think the OP's point was that open source should mean:
Free to modify
Free to contribute code
Free to use.
Leaving the first two but taking away the free to use really takes the F out
of FOSS. There have been other posts discussing Digium's license requirements,
code ownership, etc. I
Yes - the USB connection carries the data. Keep in mind that the HA aspect
of this product just means you can connect to two asterisk servers. There is
not data replication, detection of asterisk failure, etc. (without buying more
xorcom products). Be sure to do your homework. But they do
...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 30, 2011, at 8:31 PM, Kaushal Shriyan wrote:
On Sun, May 1, 2011 at 8:48 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
Yes that's it - one PRI line in, 2 out (one to the PRI card in each server).
If you have lots of PRI lines, you may
Use simple RJ45 (8 wire) A-B switched controllable by serial port, and use
HAAST to throw the A-B switch to reroute the PRI.
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Adolphe Cher-aime
or point me
to any document of website.
--
Sent from my iPhone
On Apr 30, 2011, at 12:09 PM, Michelle Dupuis mdup...@ocg.ca wrote:
Use simple RJ45 (8 wire) A-B switched controllable by serial port,
and use HAAST to throw the A-B switch to reroute the PRI
...@lists.digium.com] On Behalf Of Kaushal Shriyan
[kaushalshri...@gmail.com]
Sent: Saturday, April 30, 2011 11:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] HA Asterisk
On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
There are lots out there, but here's
For the High Availability part check out the HAAST add-on for Asterisk at
www.generationd.com
It detects a variety of failures, shuts down the failing system, starts
asterisk on the peer, moves the IP over, etc. Runs with every Asterisk variant
and every Linux distro. No special hardware
I'm still getting emails containing links to sites in russia. I have this
regex in my BombRe file:
http\:\/\/[A-Za-z\.]+?\.(?:ru|ly|nl|ec|cc|vc|ua|in|am)[\s\/\n\r\\]
Which according to my (web based) regex testers SHOULD catch the email message
below. So, I put it into the mail tester in ASSP
)\/? - or
http\:\/\/[A-Za-z\.\/]+?\.(?:ru|ly|nl)(?:\s|\/|\n|\r) - are better
read: http://perldoc.perl.org/perlretut.html
Thomas
Von:Michelle Dupuis mdup...@ocg.ca
An: assp-test@lists.sourceforge.net
assp-test@lists.sourceforge.net
Datum: 19.04.2011 04:11
Betreff:[Assp-test] Bombre
I had a discussion going yesterday about the right regex to catch a certain
URL. Here's the expression
http\:\/\/[A-Za-z\.\/]+?\.(?:ru|ly|nl|ec|cc)(?:\s|\/|\n|\r)
This works fine. Now I wanted to change the expression to catch a slash (/) as
well as a backslash, so I changed the expression
I get a lot of spam which I'm trying to identify by a regex to match the URL in
the body. I have the regex below entered into my bombre file, and the sample
message below is NOT being caught by ASSP 2.0
Any idea why? The regex tester online suggest this should match...but assp
isn't matching
Nope...IPv4. (IPv6 not even installed)..
From: Grayhat [gray...@gmx.net]
Sent: Saturday, April 16, 2011 1:06 PM
To: ASSP development mailing list
Subject: Re: [Assp-test] FW: Name server errors - bug?
That is a strange DNS = Name Server p�
It should
DNS is running fine and host is setup fine. So I hard coded the IP's into ASSP
and now it seems happy.
This is 64 bit activeperl. Many of the packages I needed were available
through PPM so I grabbed source from CPAN and compile them in. Not a perl
expert, so maybe that's an issue...
I just switched to ver 2 of ASSP...feeling a bit bleeding edge. I'm running
ASSP 2.0.1(2.0.19) on ActivePerl (win x64).
Although my system's DNS entries are right, the assp log fills with these
errors. Any idea how to fix?
Apr-15-11 16:42:17 [Worker_1] Warning: Name Server p�: does
found
The ASSP ERROR: no answering DNS-SERVER found is what I see when
there is no access to external DNS (root-servers).
Peter
Peter Bowey Computer Solutions
- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: assp-test@lists.sourceforge.net
Sent: Saturday, April 16
on Linux. I too run an MTA (Postfix) as a 'front end' -before- ASSP.
I have everything from the 'outside' sent through a gateway / forewall.
I run my own DNS Server, as I suspect you do..
Pete
Peter Bowey Computer Solutions
- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: ASSP
Message
-
From: Michelle Dupuis
To: ASSP development mailing list
Sent: Saturday, April 16, 2011 9:42 AM
Subject: Re: [Assp-test] FW:
Name server errors - bug?
Yes I run my own DNS servers too. I left
something in the DNSServers field so I will erase those. Although I'm
not a perl programmer
by the other arguments to new().- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: ASSP development mailing list assp-test@lists.sourceforge.net
Sent: Saturday, April 16, 2011 9:42 AM
Subject: Re: [Assp-test] FW: Name server errors - bug?
Yes I run my own DNS servers too. I left
I have a situation where an Asterisk server is NATted, sitting behind a PIX.
One public IP is used for one purpose, now a second public IP is required for
another.
Is there a way to have Asterisk use more than one public IP when behind NAT?
(I already use the externalIP setting)...
If not,
: Re: [asterisk-users] Multiple public address to one Asterisk
serverbehind NAT?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Tuesday, February 22, 2011 3:34 PM
To: Asterisk Users
I found some great pieces of script on the internet that I've combined to allow
Asterisk to send voicemails as an MP3 file, and encode the sender name and
number as well as message number as tags into the MP3 file. I even include a
cover art image which has our company logo and PBX symbol in
Ok - I've put the script up on the www.generationd.com web site. Just go to
the Downloads | Asterisk section to pull it down.
I would like to keep control of this script so please send me changes (don't
repost elsewhere) and I'll keep the latest version up for everyone. I'll add a
link to
Take a look at High Availability ASTerisk (HAAST) from www.generationd.com
Their software sits between the OS and asterisk, and can failover servers,
switch IP addresses, control external interfaces, etc.
It can run on different hardware (make a cluster from different/cheap boxes),
it allows
, 2011 2:44 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Dialplan to bridge 2 legs?
On Sun, 23 Jan 2011, Michelle Dupuis wrote:
Is it possible to have a call file enter the dialplan, and then initiate
2 outbound calls and then bridge them?
A call file can specify a channel
to bridge 2 legs?
Un-top-posting...
On Sun, 23 Jan 2011, Michelle Dupuis wrote:
Is it possible to have a call file enter the dialplan, and then
initiate 2 outbound calls and then bridge them?
On Sun, 23 Jan 2011, Steve Edwards wrote:
A call file can specify a channel and a context/exten/priority
We have an application that plays a variety of sound files on one leg of a call
(generated by a call file). We've been told that the party listening to the
audio files intermittantly hears robotic sounding audio (on/off during the
same call).
Anyone have ideas on cause? These calls are on an
I've searched through the wiki but I can't find what I need...I'm trying to
figure out what the max call duation is. I found references to show
application AbsoluteTimeout but that isn't in 1.6 (not even prepending core
to the front). A core help show didn't help...
--
Message-
From:
asterisk-biz-boun...@lists.digium.commailto:asterisk-biz-boun...@lists.digium.com
[mailto:asterisk-biz-mailto:asterisk-biz-
boun...@lists.digium.commailto:boun...@lists.digium.com] On Behalf Of
Michelle Dupuis
Sent: November-21-10 10:54 AM
To: Asterisk Business List
Subject
GenerationD is please to announce a Lite version of its High Availability
ASTersik product (HAAST). The new lite version is designed for small
installations (5 channels max) that want all of the failover and high
availability features of larger installations, allowing small companies to use
a
Is the a CLI command that shows all channels in use at one time? (Whether IAX,
SIP, SCCP, etc)?
As well, when I SIP SHOW CHANNELS I see phones registering showing as
channels in use. Is there a way to filter this output?
Thanks!
MD
--
We have a small office installation running over a cable modem. (8M down, 500k
up confirmed with numerous speed test sites)
When a single call is up, call quality is fine. When a second call is up,
outbound audio is immediately choppy. We're using ulaw, and confirmed that
traffic with 2
Jitterbuffer affects inbound audio only, not outbound (the other side hears the
choppiness) so I don't think that will help/
Trunking only reduces overhead after 4+ calls, so that shouldn't help either.
(Since this occurs at 2 calls)
I can't wireshark the other end since the other end is my
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen
[benny+use...@amorsen.dk]
Sent: Monday, September 27, 2010 10:35 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk Redundancy
Michelle Dupuis
: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Vahan Yerkanian
[va...@arminco.com]
Sent: Monday, September 27, 2010 1:02 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk Redundancy
On 9/27/10 8:57 PM, Michelle Dupuis wrote:
HAAST
Check out HAAST (High Availability ASTerisk) at
www.generationd.comhttp://www.generationd.com (also on the voip wiki)
You get the cluster/heartbeat replication without needing to add openSER or
full HAlinux. A simpler approach - easier to config and manage
MD
Your (local phone) dialplan is not getting pushed out to the handset. Increase
the version number in your config to force it out to the handset...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of
I realize this is getting a bit outside myth...but hopefully someone can offer
some ideas...
I'm using the latest NVIDIA drivers on Fedora 12, with Nvidia 8600GT. Although
the dual DVI outputs work great, the driver just won't detect anything
connected to the component video connector.
Is
Are there any best practices for using a SAN with Asterisk? In the past we've
kept config files local, but voicemail on a SAN. Aree there any issues with
latency putting voice prompts, configs, etc. on a SAN?
Anyone have some best practices to share?
MD
--
I need to grab the voicemail WAV file once the voicemail command is done. Is
there a hook to be notified that voicemail is done, and get the name of the
recorded file?
Thanks
MD
--
_
-- Bandwidth and Colocation Provided by
Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase, rerecord, etc.
I can probably do it through dialplan but it feels like I'm reinventing the
wheel.
...@lists.digium.com] On Behalf Of Leif Madsen
[leif.mad...@asteriskdocs.org]
Sent: Tuesday, July 27, 2010 9:49 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
On 10-07-27 08:39 PM, Michelle Dupuis wrote:
Is there a prebuild module/dialplan which gives me a nice
-boun...@lists.digium.com] On Behalf Of Sherwood McGowan
[sherwood.mcgo...@gmail.com]
Sent: Tuesday, July 27, 2010 8:47 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
There's an app_record, and I believe app_dictate
On 7/27/2010 7:39 PM, Michelle
Of Leif Madsen
[leif.mad...@asteriskdocs.org]
Sent: Tuesday, July 27, 2010 9:22 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Grab voicemail WAV file when done
On 10-07-27 08:38 PM, Michelle Dupuis wrote:
I need to grab the voicemail WAV file once the voicemail command is done
...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
[paul.belan...@polybeacon.com]
Sent: Tuesday, July 27, 2010 10:10 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
On Tue, Jul 27, 2010 at 10:05 PM, Michelle Dupuis
:55:07PM -0400, Michelle Dupuis wrote:
And after cleaning everything 323/pwlib/ptlib related, and reinstalling ptlib
and h323plus, I can't even get asterisk to compile chan_h323 anymore.
Perhaps something old was left over.
My .configure run shows:
checking /usr/src/openh323plus/h323plus
I see that objective systems has updated their ooh323 stack, but it is not
compatible with the latest chan_ooh323 wrapper available on their site.
Has anyone update the chan_ooh323 wrapper for Asterisk 1.6.2.x ?
Michelle
--
_
I'm really struggling with an Asterisk 1.6.2.7 install (on centos 5.4)
The pwlib + opal packages don't satisfy Asterisk's configure script (to let
H323 compile), so I removed those and added the latest ptlib + h323plus (from
h323plus.org)
I can compile ptlib and h323, but when I load chan_h323
+pwlib
from centos packages work? (trick asterisk .configure to accept them)?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
[mdup...@ocg.ca]
Sent: Sunday, June 20, 2010 7:45 PM
To: Asterisk
Contact me off list for more info.
Michelle
From: asterisk-biz-boun...@lists.digium.com
[asterisk-biz-boun...@lists.digium.com] On Behalf Of perl ninja
[perlni...@gmail.com]
Sent: Friday, June 18, 2010 8:14 PM
To: Asterisk Business List
Subject:
I'm looking for a small formfactor mobo for an install that needs to handle 25
phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone
know what kinds of call volume that will handle?
MD
--
_
-- Bandwidth
I checked out the sites and can't figure out what this thing is! (Without
delving into the documentation).
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew J. Roth
[mr...@imminc.com]
Sent:
I'm succesfully using IAXmodem for faxing (using hylafax) with Asterisk. I
would like a little more control for outbound calls using IAXmodem, but I'm not
sure how to do it. It looks like dialing out over IAXmodem bypasses the
dialplan altogether...can anyone confirm this?
MD
--
This isn't an Asterisk issue, it's a routing issue. Take a look at iproute2
and routing policies.
Another way to view it is that Asterisk hands the communications over to Linux,
where the network route takes over. (The * bind statement just tells * what IP
to listen on)
If you have 3
I have a Centos 5.4 64 bit installation. I've tried installing asterisk
1.6.2.7 from source, and from RPM, and although overall things work, the
chan_ooh323.so module won't load. Every attempt to load causes Capabilities
failure for OOH323. OOH323 Disabled.
I looked at the source and the
The High Availability HASTerisk (HAAST) product on www.generationd.com is a
software solution that does automatic failover, etc between multiple asterisk
machines. I'm guessing this could be part of an overall solution for you
From:
I can't find this in the wiki/email history..but I'm sure it's based asked
before.
The port range define in rtp.conf - is that for connections initiated by
asterisk? Or the port range asterisk listens on? Or both?
Thanks!
MD
--
I would like to play music to an inbound caller, AFTER asterisk answers the
call, but before the call is bridged by DIAL. Is there a simple way to
achieve this?
MD
--
_
-- Bandwidth and Colocation Provided by
Exten = s,n,wait(10,m)
Exten = s,n,Dial.
This would wait 10 seconds playing MOH before dialing.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Monday, March 22, 2010 3:58 PM
To: 'Asterisk Users
,n,Dial(SIP/callwithus/17025551212,120,A(ginr3))
On Mon, Mar 22, 2010 at 6:41 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I think I forgot some important information...
I'm actually running an AGI script after the answer (and before the dial).
I would like to play MOH while the AGI script
I've got Asterisk 1.6 bridging to an Avaya using H323. The Avaya is
autoanswering calls to music (as expected) and audio seems fine, but I see
this error on bridging:
WARNING[8833]: chan_ooh323.c:1054 ooh323_indicate: Don't know how to
indicate condition 20 on ooh323c_o_2
Is this a warning I
We have a client with Asterisk 1.6 installed via yum (onto Centos). It did
not included the chan_h323 driver apparently, so we installed add-ons by
yum. We then got ooh323.
Is it possible to install the H.323 drivers without compiling from source?
--
We're looking for someone who has experience (more than a couple of
installs) of connecting Asterisk 1.6 to an Avaya using ooh323.
We have logs, error messages, config files, and (soon) screen shots of the
config on the Avaya. We can get Asterisk to see the Avaya's gatekeeper
(CLAN) but
Michelle Dupuis mdup...@ocg.ca
We're looking for someone who has experience (more than a couple of
installs) of connecting Asterisk 1.6 to an Avaya using ooh323.
We have logs, error messages, config files, and (soon) screen shots of the
config on the Avaya. We can get Asterisk to see the Avaya's
. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis wrote:
I'm trying to connect an Asterisk 1.6 to an Avaya with
gatekeeper (CLAN). When chan_ooh323 first loads it tries to establish a
connection with the gk but I it fails. I have the following extract from
the ooh323 log
In case someone wants to see the detailed ooh323 log (which shows the failed
attempt to connect to the gatekeeper). I appreciate any help!!
21:32:06:832 Sent GRQ message
21:32:06:885 GkClient Received RAS Message
21:32:06:885 Received RAS Message = {
21:32:06:885 gatekeeperConfirm = {
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When chan_ooh323 first loads it tries to establish a connection with the gk
but I it fails. I have the following extract from the ooh323 log. Can
anyone give some insight?
Thanks!
MD
23:02:59:045 Sent GRQ message
We are coordinating a connection to a SIP provider who told us they use two
port ranges for RTP, 7000-8000 and 1-2.
We've never encountered that before (and I believe rtp.conf only supports a
single range). We can obviously setup 7000-2 within RTP.conf, but I'm
wondering if there is
without any
problems. I need your ooh323.conf and all relevant CM config (signal-group,
trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis mdup...@ocg.ca wrote:
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When
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