Hello,
When an incoming call is not not answered within a 15 secons timeframe, a
customer of mine asked me to forward it to a group of several callees.
I did this using Dial(SIP/101SIP102SIP/103).
The trouble is that sometimes, two callees would both answer almost at the
same time.
For the first
yes - but what would REALLY BE GOOD is if func_odbc
allowed Muli-stepped SQL. Since that is the ONLY way to execute a
TRANSACTION
How they thought it was a Good Idea to hamstring func_odbc like they
did is beyond me.
Tilghman Lesher wrote:
On Monday 02 June 2008 05:48, Atis Lezdins wrote:
No - I just would like to suggest that if you provide a solution in a
more clear English manner, more people can benefit from you knowledge
Which I assume is why you posted it in the first place.
Jay R. Ashworth wrote:
On Thu, May 29, 2008 at 04:24:57AM -0400, Al Baker wrote:
Quote
you mean the CO gave an All-circuts-are-busy tone ???
If not, what does AST_CONGESTION mean
Philipp Kempgen wrote:
Sanjay Rajdev schrieb:
I tried to call a number on the ZAP channel through manager, I got an
Unknown reason for failure, with the following Originate Response.
Event:
Hi All,
I am getting following error when i start AsterFax:
Please help me to solve this issue:
[EMAIL PROTECTED] asterfax]# ./asterfax.sh
log4j: Threshold =null.
log4j: Retreiving an instance of org.apache.log4j.Logger.
log4j: Setting [au.com.noojee.asterfax] additivity to
On Tuesday 03 June 2008 23:22, Atis Lezdins wrote:
chan_agent with AgentCallbackLogin was working but not completely
stable for my dialplan which was quite heavy when I was on 1.2,
however you may try that out. Or just upgrade to 1.4 (or even 1.6 and
try state_interface)
Iam using API Action
You really should discuss this at the Asterfax forums:
http://forums.asteriskit.com.au/
later,
PaulH
On Wed, 2008-06-04 at 14:17 +0530, Sukhbir Singh wrote:
Hi All,
I am getting following error when i start AsterFax:
Please help me to solve this issue:
[EMAIL
What about using RealTime LDAP in 1.6? That woudl be much faster than a RDBMS.
2008/6/3 Sherwood McGowan [EMAIL PROTECTED]:
Mindaugas Kezys wrote:
Thank you for your opinion.
Then my question would follow: how to build human-friendly system which will
use GUI and lets user use that system
On Wed, Jun 04, 2008 at 10:45:13AM +0100, Gavin Henry wrote:
What about using RealTime LDAP in 1.6? That woudl be much faster than a RDBMS.
If performance is such a major issue, why not use explicit queries?
realtime has overhead even in extensions/proiorities where it is not used.
--
Hi All;
Why busydetect=yes caused this autmatic hangup happens
without any reson (while responding to entering the
digits in the IVR) I do not know! And what is the
solution I do not know.
I used busydetect=yes and busycount=5 in zapata.conf
to help me in hangup when detect the busy tone, but I
2008/6/4 Tzafrir Cohen [EMAIL PROTECTED]:
On Wed, Jun 04, 2008 at 10:45:13AM +0100, Gavin Henry wrote:
What about using RealTime LDAP in 1.6? That woudl be much faster than a
RDBMS.
If performance is such a major issue, why not use explicit queries?
realtime has overhead even in
Hello all,
this might be a crazy question
can I connect 2 FXS plugs to the same analog phone ?
my reason: I'm expecting that, with this setup, the phone could operate
transparently through the redundant FXS if the main FXS would fail... of
if asterisk is stopped on one of the servers...
Hi Benoit,
Anyone already did that (changing jabber status/ status message of many
accounts)
or know if it's even remotly possible ??
We discussed that during the last XSF devcon in Brussels. Actually
Asterisk (or any other XMPP client) cannot change the Jabber status on
behalf of another
Reading this thread with interest..
Curiously enough I asked something similar a while back - got a few
replies and at the time decided to stick to pure dialplan - but my
application was a general purpose PBX type of thing, and I didn't want to
use realtime nor an SQL database... ('embedded'
Hi Matt,
On Wed, Jun 4, 2008 at 1:05 AM, Matthew Gibson [EMAIL PROTECTED] wrote:
I'd be interested to know more about the status abilities as well, we've
tried to test jabberstatus application, but it doesn't seem to function as
we expect, it should be returning 0,1,2,3,4,5 based on users
I did not see anything wrong with his English, but maybe your
understanding of the subject is lacking. It is like talking to my
mother about routing or SIP.
People here will often not spoon feed you the answers but give you a clue.
Google results from his clue (thread apply all bt)
Less than two minutes of googling:
http://www.asterisk.org/doxygen/1.2/AstDebug.html
Please try to figure out your own problem before sending it to a list
with thousands and thousands of recipients.
Thanks,
Steve Totaro
On Wed, Jun 4, 2008 at 8:17 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
I
When I send a call out the MAX I get the following
-- Got SIP response 484 Address Incomplete back from 172.16.10.230
Any ideas on how to make 911 appear as a ten digit number to the device so
that it will pass the number out to the PSTN ?
This is not a max tnt problem, the tnt
Thanks for the answer
My case is the following
queues.conf
[6010]
;fullname = techsupport
strategy = rrmemory
timeout = 10
context = ntech
wrapuptime =
autofill = yes
autopause = no
maxlen =
joinempty = no
leavewhenempty = no
reportholdtime = no
musicclass =
member = SIP/6000
extentions.conf
On Wednesday 04 June 2008 02:07:13 Al Baker wrote:
Tilghman Lesher wrote:
On Monday 02 June 2008 05:48, Atis Lezdins wrote:
You can use func_realtime in dialplan, that will be much faster as it
doesn't create separate process (as AGI does), and uses internal
asterisk connection pool, so
On 4 Jun 2008, at 11:43, Joao Ferreira gmail wrote:
can I connect 2 FXS plugs to the same analog phone ?
No. Fire and death.
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On Wed, Jun 4, 2008 at 9:04 AM, Steven Howes [EMAIL PROTECTED] wrote:
On 4 Jun 2008, at 11:43, Joao Ferreira gmail wrote:
can I connect 2 FXS plugs to the same analog phone ?
No. Fire and death.
Unless you use a 2-lines analog phone :)
___
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On 6/3/08, Joe Carroll [EMAIL PROTECTED] wrote:
Quick question for the folks using MAX TNTs for aggregators..
When I send a call out the MAX I get the following….
-- Got SIP response 484 Address Incomplete back from 172.16.10.230
Any ideas on how to make 911 appear as a ten
Hi!
I just upgraded my Asterisk server from 1.4.6 to 1.4.21 and now I experience
some strange behaviour.
1) The Asterisk CLI (asterisk -r) stops responding after some minutes. I
cant CTRL-C or exit the CLI anymore and no activity is shown. Just like if
the connection is interrupted.
2) When I
I noticed safe_asterisk is nolonger used from the init.d script (on
ubuntu) for asterisk-1.6.0-beta9. I'm curious if there is another
init.d script out there, or even the best way to call safe_asterisk.
Or is safe_asterisk nolonger the script of choice for starting,
restart asterisk.
One of the
Simon Hyde wrote:
Hi,
G.722 is heavily used by Broadcasters worldwide for wideband voice
communications over ISDN. I'd like to be able to receive these G.722 over
ISDN
calls into an Asterisk exchange (with mostly a view to routing the calls to a
Voicemail box where material can be
Which flavor of G.722 has been implemented in Asterisk? And starting
with what release version?
Thanks,
Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
[EMAIL PROTECTED]
The first place you may want to look is in the SYSLOG of the TNT,
allowing you to see things such as the ISDN error code along with the
SIP code. You can try to catch that on the terminal of the TNT, but it
may make more sense to pipe your syslogs out to an external box, if you
aren't doing
Michael Graves wrote:
Which flavor of G.722 has been implemented in Asterisk? And starting
with what release version?
The only flavour with a defined RTP format is the full 64kbps one.
Steve
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Is there some location that outlines the major differences between
Asterisk version 1.4 and version 1.6? I've read through change logs and
several technical discussions, but technical details don't really give
me the big picture. Basically, is 1.6 more stable than 1.4? Is it more
efficient?
2008/6/4 Brent Davidson [EMAIL PROTECTED]:
[snip]
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
[snip]
Just a small aside...
You go to the trouble of building/using Oslec, and then use hardware
EC? Very odd. Does Oslec understand
Have you tuned rxgain txgain in Zapata.conf? shameless-self-plug
http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/
/plug
Also, have you used fxotune to tune each FXO interface?
I believe echo cancellation happens at the Zaptel / DAHDI level, so using
Brent Davidson wrote:
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
Why on earth are you running two layers of echo cancellation - hardware
and software? To be honest, I think this is asking for trouble - I've
seen two occasions
Yes, I was using a name instead of an IP address. And if memory
servesI *think* it is using TCPprefered...but I could be wrong.
Kevin
Mike wrote:
I have been running into a few issues with Asterisk/polycom and I am
running out of ideas. This problem has been ongoing for the last couple
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Brent Davidson wrote:
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
Why on earth are you running two layers of echo cancellation - hardware
and software? To be
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Brent Davidson wrote:
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
Why on earth are you running two layers of echo cancellation -
Matt Watson wrote:
Have you tuned rxgain txgain in Zapata.conf? shameless-self-plug
http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/
/plug
Also, have you used fxotune to tune each FXO interface?
I believe echo cancellation happens at the Zaptel /
Something that I can put on our internal company website to replace
our hardware IP phones.
I see many web 2.0 startups offering browser based clients for their
own service, but I can't seem to find anything that I can use with my
own PBX. Do I suck at searching google or has the future not
I have a need to disable the Send reply feature in asterisk voicemail
(1.2) because we have an environment where multiple servers use the same
real-time database for voicemail but the voicemail files are stored on
the individual server that the user registers with. When a user on
server A
Hi,
I want to reduce the dead time before the queue is calling the next agent. I
see there 5 seconds delay.
It is possible to reduce this time, or what is Asterisk doing within this
timeframe.
best regards
Thomas
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Have you seen these client?
http://www.mozillavoip.com/
http://tringme.com/
http://www.twoiplink.com/
http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsFirefox
(Dated, since project changed names)
BTW, you can also trying to roll your own using old OpenWengo...
Is anyone running Lumenvox on Gentoo? My asterisk install has been running
like a champ for a few years now and I really hate the thoughts of changing
distros just for Lumenvox.
Here is my issue:
The engine needs the libs from boost. I emerged boost and noticed that
there were four libs that
EdPimentl [EMAIL PROTECTED] wrote:
Have you seen these client?
http://www.mozillavoip.com/
http://tringme.com/
http://www.twoiplink.com/
http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsFirefox
I was hoping that there was an open, free, full featured sip
Steve Underwood wrote:
Michael Graves wrote:
Which flavor of G.722 has been implemented in Asterisk? And starting
with what release version?
The only flavour with a defined RTP format is the full 64kbps one.
Steve
I was going to say strawberry, but to try to answer his other question,
Just an update. I tried updating to the newest Rhino Release firmware
1.15 and newest stable driver version 2.2.6. It works OK with
zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against
zaptel 1.4.10.1 Asterisk does not see any zap channels. I'm currently
running one branch
On 4 Jun 2008, at 21:00, Hilary Miller wrote:
EdPimentl [EMAIL PROTECTED] wrote:
Have you seen these client?
http://www.mozillavoip.com/
http://tringme.com/
http://www.twoiplink.com/
http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsFirefox
I was
On Wed, Jun 4, 2008 at 4:13 PM, Tim Panton [EMAIL PROTECTED] wrote:
You won't (yet) find a Flash implementation that talks direct to
your Asterisk because Flash doesn't support UDP (yet) and
it doesn't include a VoIP protocol (yet).
So all the Flash softphones out there have to use a
Greetings.. i'm facing a slight problem i hope.. the management in my call
center requires using the chanspy 555 to monitor newly hired agents.. and there
seems a problem where the monitoring extension gets stuck and can't soft hang
upit .. anyone got a solution for that? it just gets stuck
Make sure you enable all the USE flags, and then perhaps try
emerge boost
again
I've had times where leaving out a badly named USE flag meant that
critical things didn't end up getting built. A particularly egregious
must enable all USE flags case is if you try
emerge ejabberd
Without all the
The Asterisk development team has released Asterisk-Addons version 1.2.7,
1.4.9,
and 1.6.0-beta4 to address a major security vulnerability in the ooh323 channel
driver. The releases may be downloaded from http://downloads.digium.com/.
AST-2008-009 details a remote crash vulnerability in the
Asterisk Project Security Advisory - AST-2008-009
++
| Product | Asterisk-Addons |
Hi to all
if someone of you is interested on it, i've changed the code of app_asr.c
With these patch you can use the ASR application to play DTMF tones,
so you can have your own AGI application that uses the ASR and manages
the DTMF tones without change the dialplan.
EXAMPLE
exten =
Asterisk Project Security Advisory - AST-2008-009
++
| Product | Asterisk-Addons |
See below, we replaced the area code and prefix of with NPANXX for concerns
Interestingly enough, on the syslog messages from the TNT we are seeing Called
= 911, Q850 Cause = 28, SIP Response = 484
Extension Changed NPANXX7604 new state InUse for Notify User NPANXX7555
-- Executing
Does anyone have any experience getting Avaya phones working with
Asterisk? (I.E. 9650) BLF etc?
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To UNSUBSCRIBE or update options visit:
you can download a FREE browser softphone and or clcik to call at
http://1ezphone.com/downloads Let me know if you have any porblems and I
can help you
- Original Message -
From: Hilary Miller
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
You can download a FREE browser softphone and or cliick to call that
supports UDP athttp://1ezphone.com/download It works well with Asterisk I
use it everyday
- Original Message -
From: Hilary Miller
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Yes we do everyday here at Google
- Original Message -
From: Mark Best
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Avaya IP Phones with *
Date: Wed, 4 Jun 2008 15:24:16 -0700
Does anyone have any experience getting Avaya phones
None of them have features like hold, transfer, voice mail, dtmf,
conferenceas far as I know none of them has caller ID Only 1ezphone.com
has all that and the buttons are programmable for CRM features.
- Original Message -
From: Tim Panton
To: Asterisk Users Mailing List -
Cause 28 indicates Invalid number format.
Joe Carroll wrote:
See below, we replaced the area code and prefix of with NPANXX for
concerns
Interestingly enough, on the syslog messages from the TNT we are seeing
Called = 911, Q850 Cause = 28, SIP Response = 484
Extension Changed
Busy Lamp features? How is the sound quality compared to
Polycom/Cisco/Snom etc? Recommend this kind of phone?
(FYI: Doing phone research - while trying to be 'backwards'-compatible
with an Avaya IP G450/S8700 system.)
From: [EMAIL PROTECTED]
[mailto:[EMAIL
sorry its http://1ezphone.com/download
- Original Message -
From: Bob G
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Browser based VoIP client? -
http://1ezphone.com/downloads
Date: Wed, 4 Jun 2008 17:46:08 -0500
you can
you can reduce the 5 seconds to any number you wish.. but from a personal
experience .. if you put the retry to zero.. nothing will change.. i suggest to
use 1 as your minimum aiting number
Tarek Sawah
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Wed, 4
Jun 2008
We use Polycom 650s on our asterisk and Avaya G700s along with Avaya IP
phones .The sound is good on all of them.
- Original Message -
From: Mark Best
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Avaya IP Phones with *
Date: Wed, 4
On Wed, 2008-06-04 at 18:01 -0500, Bob G wrote:
sorry its http://1ezphone.com/download
Anyone ran wireshark on the box running this app? Who's to say this
binary swf is to be trusted? Is the source available somewhere?
Cheers,
Patrick
___
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I have my SIP provider and Astra 480i's set to ulaw, but unless my
Snom M3's aren't set to alaw they sound very bad as they pop and drop out?
Why is this?
Thanks!
jlc
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On June 4, 2008 06:20:57 pm Joe Carroll wrote:
Interestingly enough, on the syslog messages from the TNT we are seeing
Called = 911, Q850 Cause = 28, SIP Response = 484
That really looks like the switch that the TNT is talking to is rejecting the
number, not the TNT...
-A.
JR Richardson wrote:
You mentioned this started happening 3 months ago, what happened then?
Network changes, equipment changes, traffic increased, new users
(downloading allot during the day, surfing porn), wireless
interference?
The initial problem started when our DS3 was throwing
On Wed, Jun 4, 2008 at 5:52 PM, Bob G [EMAIL PROTECTED] wrote:
None of them have features like hold, transfer, voice mail, dtmf, conference
as far as I know none of them has caller ID
Only 1ezphone.com has all that and the buttons are programmable for CRM
features.
Hrm:
- no apparent
Hello,
I've run fxotune at different times but continue to get what seem to be
strange numbers in /etc/fxotune.conf. It ends up with:
5=7,255,251,251,2,255,255,1,255
6=7,255,251,251,2,255,255,1,255
7=7,255,251,251,2,255,255,1,255
8=9,2,250,253,4,252,0,255,255
On Wednesday 04 June 2008 22:02:19 John Morey wrote:
Hello,
I've run fxotune at different times but continue to get what seem to be
strange numbers in /etc/fxotune.conf. It ends up with:
5=7,255,251,251,2,255,255,1,255
6=7,255,251,251,2,255,255,1,255
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