Hi all,
Is it possible to store voicemail in a Mysql database without using ODBC?
I've got RTA sip and voicemail working; I just want to store the messages in
the db now. Configuring ODBC seems like a lot of work if I don't have to.
TIA.
--
Take care and have fun,
Mike Diehl.
--
5 maj 2011 kl. 05.28 skrev Flavio Goncalves:
My 2 cents. All these problems seem to be lack of focus. Digium,
please stop doing everything to everyone. Too many versions, too many
features, too many code, too many bugs. Following the Pareto's
principle, 80% of the users use only 20% of the
5 maj 2011 kl. 06.33 skrev Olivier:
2011/5/5 Flavio Goncalves fla...@asteriskguide.com
snip
but stuffing Asterisk with
many new features on each version does not seem to be contributing to
the stability of the code or the migration to newer versions.
yes but it seems to me that code
Hello,
is there yet any feedback on this ??
What is causing this and how to overcome this ?
Using Asterisk 1.6.17.2, but had it on the past also with version
1.6.2.10, 1.6.2.11 and 1.6.2.16.1
I had it again today :
[May 5 10:13:51] DEBUG[16223] audiohook.c: Failed to get 160 samples
from
Hello,
and this is what happened before :
/[May 5 10:08:38] DEBUG[16215] channel.c: Hanging up channel
'SIP/expsom10-0442'
[May 5 10:08:38] DEBUG[16215] chan_sip.c: This call was answered
elsewhere[May 5 10:08:38] DEBUG[16215] chan_sip.c: ### It's the
cause code, buddy. The cause
Hello list,
can it be that this has something to do with MixMonitor :
[May 5 10:49:41] DEBUG[19790] func_audiohookinherit.c: Set audiohook
MixMonitor to be inheritable
[May 5 10:49:41] DEBUG[19869] audiohook.c: Read factory 0x95dc590 and
write factory 0x95dcfc8 both fail to provide 160
On 04/05/11 18:17, || dave cantera Mobile wrote:
paul, doug,
I had several AMD athlons 64bit... no problems running centos, suse.
they seem solid on 1.4.xx... had a few intel celerons and P4s. they were
good as well. guess I was Lucky back then!
thanks for supporting the list!
daveC
don't get
On 05/05/11 05:41, Cary Fitch wrote:
Flavio E. Goncalves
www.asteriskguide.com http://www.asteriskguide.com
Compare to which version of Windows… Patches are a never ending process
Cary Fitch
I think this attitude is half the problem. Asterisk is not a desktop
computer
Hello,
I have some problems in placing calls through IAX... It does not work :)
in the asterisk console I can't see nothing about dialplan enter or
so, IAX debbugging seems to be unuseful...
this is my configuration:
[612]
type=friend
secret=123456
notransfer=yes
disallow=all
allow=gsm
On 05/05/11 00:02, Ira wrote:
Not that it applies but I recently installed a Snom M3 and it seems to
behave like you want. When I walk out of range and then back in the call
is usually still there. I've not tested past that so it might hang up
after an unknown timeout.
Ira
The difference
On 05/05/11 04:37, Richard Kenner wrote:
I recently tried to update my Aastra 57i to version 3.2 and ran into
a problem. It won't properly register and says contact mismatch.
I added sip contact matching: 2 to aastra.cfg, but that didn't help.
When I look at the SIP trace, but I see is the
Yes, I'm talking about mid-call.
I do have rtptimeout and qualify set, both to 30 seconds, which should be
plenty of time.
I set them both because if a phone moves out of range, and never comes back,
asterisk was keeping the channel open way to long.
On Wed, May 4, 2011 at 7:50 PM, Matt Riddell
On 5/05/11 10:46 PM, Shawn L wrote:
Yes, I'm talking about mid-call.
I do have rtptimeout and qualify set, both to 30 seconds, which should
be plenty of time.
I set them both because if a phone moves out of range, and never comes
back, asterisk was keeping the channel open way to long.
Yeah,
5 maj 2011 kl. 12.04 skrev Paul Hayes:
On 05/05/11 05:41, Cary Fitch wrote:
Flavio E. Goncalves
www.asteriskguide.com http://www.asteriskguide.com
Compare to which version of Windows… Patches are a never ending process
Cary Fitch
I think this attitude is half the
What happens when base stations are moved such as at any time, handsets are
either within range of one (or two or more) access point ? Is the call ended
when the handset moves from one access point to another ?
Are those access points 80211r compliant ?
--
I would never choose to use it. Our system is built on top of it (before I
ever got here) and it would be too great a task to change it not to mention
management would not go for a change.
On Wed, May 4, 2011 at 6:09 PM, Matt Riddell li...@venturevoip.com wrote:
On 5/05/11 3:02 AM, vip killa
Hi All;
When the endpoint register on Asterisk or initiate a call, so they exchange the
sip username and password. What is the possibility that this will be capture by
the hacker and how to avoid this problem?
Regards
Bilal
--
5 maj 2011 kl. 14.08 skrev bilal ghayyad:
Hi All;
When the endpoint register on Asterisk or initiate a call, so they exchange
the sip username and password. What is the possibility that this will be
capture by the hacker and how to avoid this problem?
We never exchange passwords in
Little to none...SIP is set up so that the packet contains identifiable data
(the username) but the authentication is performed with a digest of the
username password [domain] and [CalliD] (I think I got that right)
On Thu, May 5, 2011 at 7:08 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi
Bilal,
On 05/05/2011 08:08 AM, bilal ghayyad wrote:
When the endpoint register on Asterisk or initiate a call, so they
exchange the sip username and password. What is the possibility that
this will be capture by the hacker and how to avoid this problem?
Strictly speaking, there is no
Thanks Alex for clearing up the bit about the NONCE, that's what I was
trying to remember when I said CallID :)
Good explanation by the way! :)
On Thu, May 5, 2011 at 7:17 AM, Alex Balashov abalas...@evaristesys.comwrote:
Bilal,
On 05/05/2011 08:08 AM, bilal ghayyad wrote:
When the
5 maj 2011 kl. 14.17 skrev Alex Balashov:
Bilal,
On 05/05/2011 08:08 AM, bilal ghayyad wrote:
When the endpoint register on Asterisk or initiate a call, so they
exchange the sip username and password. What is the possibility that
this will be capture by the hacker and how to avoid this
Not sure if you are issuing DHCP at the access point or from a central control
From a central control should allow seamless roaming within different APs,
assuming easy auth to the AP, the only issue you get is when the handset
dithers between choosing signals from one or the other, and thats
On 05/05/2011 08:21 AM, Olle E. Johansson wrote:
Because they HAVE TO. In the 401/407 reply, there's a challenge
(nonce) which is an important part of the MD5 Digest Auth scheme.
I meant more to contrast with how some UACs will attempt to re-cycle old
Authorization credentials in
On Thursday 05 May 2011, bilal ghayyad wrote:
Hi All;
When the endpoint register on Asterisk or initiate a call, so they exchange
the sip username and password. What is the possibility that this will be
capture by the hacker and how to avoid this problem?
If the two devices are connected by
Asterisk does indeed send an Options before the OK but my 57i doesn't
seem to mind.
That's odd. It does for me.
Perhaps you need to upgrade firmware on the Aastra phone?
The problem occured when I DID upgrade it! Precisely to the one
you mentioned.
Or turning off qualify for this peer
Hi,
I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now
when a caller is placed into Queue and gets connected with Member, I want to
record the call. It does record the call when I use MixMonitor() before
placing the caller into Queue, but not when MixMonitor() is used in
Hello list,
what does this mean :
[May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered
elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the
cause code, buddy. The cause code!!!
[May 5 14:58:12] DEBUG[8770] chan_sip.c: This call was answered
elsewhere[May 5
On 05/05/11 13:41, Richard Kenner wrote:
Asterisk does indeed send an Options before the OK but my 57i doesn't
seem to mind.
That's odd. It does for me.
Perhaps you need to upgrade firmware on the Aastra phone?
The problem occured when I DID upgrade it! Precisely to the one
you
On 05/05/11 14:04, Jonas Kellens wrote:
Hello list,
what does this mean :
[May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered
elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause
code, buddy. The cause code!!!
[May 5 14:58:12] DEBUG[8770] chan_sip.c: This
5 maj 2011 kl. 15.11 skrev Paul Hayes:
On 05/05/11 14:04, Jonas Kellens wrote:
Hello list,
what does this mean :
[May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered
elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause
code, buddy. The cause code!!!
Hi,
Is there a reliable way to auto-dial SIP phones (specifically Polycom) with
some sort of TAPI driver in Windows? I am aware of SIPTAPI, which makes the
user's phone ring, and when picked up dials the desired number, but I (and
more to the point, many of my customers) find this annoying.
On 05/05/2011 03:11 PM, Paul Hayes wrote:
On 05/05/11 14:04, Jonas Kellens wrote:
Hello list,
what does this mean :
[May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered
elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause
code, buddy. The cause code!!!
[May 5
In that case it suggests it is some setting you have applied to the
phones that is causing it. Can you post the local.cfg server.cfg
files from the phone (removing the passwords from there first)?
Sure: local.cfg is checksums, server information, and:
contrast level: 3
ringer volume: 8
On 05/05/11 14:16, Olle E. Johansson wrote:
We've had that for quite some time. There's an option to Dial() and one for
Queue() to enable it. Check the documentation.
/O
yes my only problem with the 'c' option for the Dial command is that it
still seems to add the Reason header if the
In that case it suggests it is some setting you have applied to the
phones that is causing it.
I just called Aastra tech support. I'm always VERY impressed that the
first person who picks up the phone is very technical. He said that they've
had reports of this issue. The problem goes away
Hi All,
Just wondering is it safe to use asterisk 1.8 latest branch on production ?
http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision 317100
-S
--
_
-- Bandwidth and
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Thursday, May 05, 2011 9:14 AM
To: asterisk-users
Subject: [asterisk-users] Asterisk 1.8 latest branch safe for production ?
Hi All,
Just wondering is it
Thanks Danny - We are replacing 1.2.x and with just basic features
There are some bug in 1.8.3.3 thats why i thought go with latest branch. cant
wait for 1.8.4.x release
-S
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Thu, 5 May 2011 09:17:14 -0500
Subject: Re:
On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell sruff...@digium.com
wrote:
I know this thread is dead but: I do not believe this should go into the DAHDI
kernel modules.
I agree. It's just too bad Dahdi is unable to report how an outgoing
call is doing: Still ringing, busy, answered.
--
5 maj 2011 kl. 16.35 skrev Gilles:
On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell sruff...@digium.com
wrote:
I know this thread is dead but: I do not believe this should go into the
DAHDI
kernel modules.
I agree. It's just too bad Dahdi is unable to report how an outgoing
call is
On 11-05-04 06:01 PM, Matt Riddell wrote:
On 3/05/11 4:01 AM, Hans Witvliet wrote:
Just a thought
If Digium / the community realy want an objective way of deciding
whether can/should migrate to any other version, you realy need a
feature-matrix (pethaps starting from version 1.2.*)
And for
Where's the dislike button?
On May 4, 2011 7:03 AM, vip killa vipki...@gmail.com wrote:
screw that i just got hylafax to work with IAXMODEM...i refuse to pay
digium a dime... supposed to be open-source right?
On Wed, May 4, 2011 at 9:52 AM, Danny Nicholas da...@debsinc.com wrote:
Here is the thing, there is nothing stopping 'the community' today from doing
this. In fact, we already have a testsuite [1] in place, running each
subversion commit and producing results for the last year. But this is only
one type of testing; automated, we also have unit tests built
Hi ALL,
Does IAX2 support call completion feature ? I meant between SIP --- IAX2 ? or
IAX2---IAX2
-S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hi,
If my memory serves me right, up to Asterisk 1.6, Queue app internals kept
the application from working some other apps such as PickUp.
I wonder if such things are possible (and if possible, still keep useful
Queue Logs ie logs in which picked up or transfered calls are shown as
such):
1- a
Hello Mike,
It is possible for Polycom phones to auto-answer an incoming call with
speakerphone. I don’t have the details available right now, but it requires
changing the phone’s configuration and sending a custom sip header with the
INVITE. Great care should be taken when implementing this,
On Thu, May 5, 2011 at 7:46 AM, Asterisk Man theasterisk...@gmail.comwrote:
Hi,
I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now
when a caller is placed into Queue and gets connected with Member, I want to
record the call. It does record the call when I use
2011/5/5 Olivier oza_4...@yahoo.fr
Hi,
If my memory serves me right, up to Asterisk 1.6, Queue app internals kept
the application from working some other apps such as PickUp.
I wonder if such things are possible (and if possible, still keep useful
Queue Logs ie logs in which picked up or
I have been using sendfax and recievefax with 1.8.x.x version I have a
patch that Kevin Fleming wrote to allow the forced shutoff of T.38 F
option. This was considered a new feature so it is not in new releases of
1.8.x and I have not been able to get a patch working for the current
releases.
Pan,
Thank you, that makes sense. I’ll investigate further.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. Christensen
Sent: Thursday, May 05, 2011 11:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial
On Wed, May 4, 2011 at 9:01 AM, vip killa vipki...@gmail.com wrote:
screw that i just got hylafax to work with IAXMODEM...i refuse to pay
digium a dime... supposed to be open-source right?
I don't believe you really understand what Open Source means...it does not
mean FREE. It means that
On Thu, May 5, 2011 at 10:07 AM, Olle E. Johansson o...@edvina.net wrote:
So how can we fix this? How can we get more people involded? What makes
projects like FedoraTesting[3] and DebianTesting[4] popular? How can the
Asterisk project reproduce their success?
Give them something that
I don't believe you really understand what Open Source means...it
does not mean FREE.
Actually, it DOES mean free, especially since Asterisk is under the
GPL. But, as RMS often says, that's 'free' as in 'free speech', not
'free beer'. That problem doesn't exist in French, where there are
two
It isn't any better than the so called t.38 support in Asterisk that
only drops calls. Gee I wonder why, maybe so they can sell their fax
product?
On Wednesday, May 4, 2011, Steve Edwards asterisk@sedwards.com wrote:
On Wed, 4 May 2011, vip killa wrote:
screw that i just got hylafax
Hi,
Does anyone know if Asterisk is a good tool to be used for a large quantity
of g711 and g729 transcoding?
What is the best alternative for that?
--
Woody Dickson
woodydick...@gmail.com woody.dick...@gmail.com
US and Worldwide Termination
Contact me for the following
On 05/05/2011 12:36 PM, Woody Dickson wrote:
Does anyone know if Asterisk is a good tool to be used for a large
quantity of g711 and g729 transcoding?
Well, what is large? Transcoding is a fairly CPU-bound process.
In principle, the answer to your question is probably more yes than no,
At 07:56 AM 5/5/2011, you wrote:
So how can we fix this? How can we get more people involded? What
makes projects like FedoraTesting[3] and DebianTesting[4]
popular? How can the Asterisk project reproduce their success?
Well, it's not a lot of people willing to run beta software on their
From: Ira i...@extrasensory.com
Sent: Thursday, May 05, 2011 12:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4
behind?
At 07:56 AM
Hi All,
I am using
http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/
to implement missed call feature. and i modify script to grab email address
from voicemail.conf
But i am not able to see DEST extension in this script ? what would be the
Set a variable ${_CALLED_EXT} to ${EXTEN} before you hang up the call, then
reference that variable in your h exten.
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 11:59 AM, satish patel satish...@hotmail.com wrote:
Hi All,
I am using
You want me to do this in macro-stdexten ? I have following dialplan. I have
used h extension in original context because you can't you h inside macro
right ?
[macro-stdexten]
exten = s,1,Dial(${ARG2}iax2/${ARG1},20,t) ; Ring the interface,
20 seconds maximum, call screening
On Thu, 2011-05-05 at 18:16 +0530, Asterisk Man wrote:
Hi,
I have a simple Queue(named 1) and one Member(SIP/1119) logged into
it. Now when a caller is placed into Queue and gets connected with
Member, I want to record the call. It does record the call when I use
MixMonitor() before placing
Also check for CANCEL, since this should be the status if the caller
hangs up before the call is picked up.
But CANCEL is return nothing
[macro-stdexten]
exten = s,1,Dial(${ARG2}iax2/${ARG1},20,t) ; Ring the interface,
20 seconds maximum, call screening option (or use P for
The majority of open source projects out are NOT run by commercial
institutions... they are run by people committed to a better product (not to
making money)... they are maintained by people who have the resources to
host a repository (which does not take a lot of resources) and a community
of
if you saved ${_CALLED_EXT} to the value of ${EXTEN} from within the macro,
you'd get 's'do it while you still have the called number as the EXTEN
On Thu, May 5, 2011 at 12:42 PM, satish patel satish...@hotmail.com wrote:
Also check for CANCEL, since this should be the status if the caller
I had that problem too,
I wastesting with asterisk 1.8.3.2 and come across this:
Call from one extension to another with:
[macro-internal-call];ARG1=extension to call
exten = s,1,Set(TOCALL=${DB(SIP/${ARG1})})
exten = s,2,Dial(SIP/${TOCALL},60,tT)
...
As I had no entry in the asteriskdb, so
Could you please tell me how ( Syntax ) and where in macro ?
I am not expert in dialplan variables. I appreciate your help
Date: Thu, 5 May 2011 12:44:19 -0500
From: sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] missed call notification
if you
After google i found something and i tried following. I set variable before
Dial and its give me proper value in h extension but now question is if
multiple user dial multiple extension then will it overwrite current variable
value ?
exten = s,1,Set(_CALLED_EXT=${ARG2})
exten =
On Thu, May 5, 2011 at 1:43 PM, vip killa vipki...@gmail.com wrote:
The majority of open source projects out are NOT run by commercial
institutions...
Postfix kicks butt. But only because IBM paid for development, for a
long number of years, and because they hired somebody who had a really
good
No, the variables are channel specific except for when they're inherited,
which doesn't affect you here
On Thu, May 5, 2011 at 1:02 PM, satish patel satish...@hotmail.com wrote:
After google i found something and i tried following. I set variable
before Dial and its give me proper value in h
I forget the term, but basically the variables you set on a current active
channel are only accessible on that channel. In this case the variables are
specific to the specific call in progress.
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 1:02 PM, satish patel satish...@hotmail.com wrote:
And Sherwood beats me to the punch again :).
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 1:15 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote:
No, the variables are channel specific except for when they're inherited,
which doesn't affect you here
On Thu, May 5, 2011 at 1:02 PM,
You guys awesome! its working now only i need to modify script and do some
trimming
Thanks a lots again..
-S
From: wcse...@selbytech.com
Date: Thu, 5 May 2011 13:20:59 -0500
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] missed call notification
And Sherwood beats me to
This issue has been resolved in latest branch 1.8 and will be resolved 1.8.5
version.
Thanks for report.
-S
Date: Thu, 5 May 2011 19:47:50 +0200
From: supp...@accm.at
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk-1.8 crash if no extension specified
in
Heheh, well Warren, I'm just a quick draw I guess ;-) Hey, at least you have
dCAP by your name! I've been at this 6-7 years and still haven't gotten off
my butt and taken the tests :D
On Thu, May 5, 2011 at 1:20 PM, Warren Selby wcse...@selbytech.com wrote:
And Sherwood beats me to the punch
Hello list,
I'm looking for a way to have the estimated hold time on a queue prior to
joining it.
someone suggested to me to Queue() first for 1 sec, read
variable QUEUEHOLDTIME, validade it and Queue() again.
But as we're using real time configuration that would mean a event
ENTERQUEUE and a
Hey all!
I'm trying to do a bit of logic here so that a user only has to dial one
code to pause/unpause in a queue (e.g. *0 will (un)pause depending on the
users's state). My logic looks fine to me but every time ${PQMSTATUS} shows
up empty.
Here's the extensions.conf part
exten =
On 05/06/2011 02:09 AM, David Backeberg wrote:
T.38 has a boatload of problems, and most of those problems are
because people who aren't employed by Digium did not read the specs,
or they did read the specs, but felt like they had to violate the
specs to get their code to work with a different
On Thu, May 5, 2011 at 4:07 PM, Paul Belanger pabelan...@digium.com wrote:
On 11-05-05 12:30 PM, Ira wrote:
At 07:56 AM 5/5/2011, you wrote:
So how can we fix this? How can we get more people involded? What
makes projects like FedoraTesting[3] and DebianTesting[4] popular? How
can the
At 01:07 PM 5/5/2011, you wrote:
I am not saying using production servers to test, rather reproducing
your production setups in a test environment. You would then create
test plans or test cases of the features you use in Asterisk. Once
documented, for each and every RC of Asterisk you go
On 6/05/11 8:35 AM, Ira wrote:
At 01:07 PM 5/5/2011, you wrote:
I am not saying using production servers to test, rather reproducing
your production setups in a test environment. You would then create
test plans or test cases of the features you use in Asterisk. Once
documented, for each and
PQMSTATUS is set only after you run the application PauseQueueMember().
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 2:11 PM, Louis Carreiro carreir...@gmail.com wrote:
Hey all!
I'm trying to do a bit of logic here so that a user only has to dial one code
to pause/unpause in a queue
Richard wrote:
No, conference scheduling is not a feature that we have built
directly into ConfBridge, and I'm debating on what it would look
like.
Scheduling isn't built into MeetMe either, but the fact that it
dynamically reads from a database means that you can write external
programs
Hi All,
I would like to write a script to run on peers to monitor my resources such
as whether a card was removed and send a signal to LB so it can resize the
capacity configuration for that peer, but I have no idea which event in
Asterisk should be monitored when a card was remove or added?
Well that makes sense
Is there anyway to determine the user's state ([un]paused) in the dial plan
prior to using the [Un]PauseQueueMember() app?
I'd like to be able to keep it very simple for my users by instructing them
to dial one key code to pause or unpause.
v/r,
Me
On Thu, May 5,
There may be a more elegant way to do it, but what I do is simply call
PauseQueueMember() first, then check the result of PQMSTATUS. I forget the
exact results but I think it's either 'Success' or 'Already Paused', and then I
base my next step in logic based on that result.
Thanks,
--Warren
Hmmm I'll have to check that out tomorrow AM. So you're calling
PauseQueueMember() to actually pause the user first? Something like
PauseQueueMember(,${EXTEN})?
If the PQMSTATUS is Success instead of Already Paused, wouldn't it
always respond with Success therefore not allowing the logic to
Hi all,
I want to play an audio hint to caller when his dial fail rather than the
current sound dodo. The caller use asterisk to do the call so I want to
setup this asterisk to achieve play specific audio when the dial fail
or time out.
How to setup to achieve that?
Thanks!
--
On Thu, May 5, 2011 at 8:20 PM, Louis Carreiro carreir...@gmail.com wrote:
Hmmm I'll have to check that out tomorrow AM. So you're calling
PauseQueueMember() to actually pause the user first? Something like
PauseQueueMember(,${EXTEN})?
If the PQMSTATUS is Success instead of Already
At 03:00 PM 5/5/2011, you wrote:
Yes, but in my world there is one Atom powerd Linux box running
Asterisk, 4 or 5 Windows machines and 2 Macs. If I want to test, it has
to be on my production box and I'm more than happy to run beta software
on that box. My comment is just that the protocol for
On 6/05/11 3:14 PM, Ira wrote:
At 03:00 PM 5/5/2011, you wrote:
Yes, but in my world there is one Atom powerd Linux box running
Asterisk, 4 or 5 Windows machines and 2 Macs. If I want to test, it has
to be on my production box and I'm more than happy to run beta software
on that box. My
Thank you very much for your response and suggestion.
I raised the question because in my project I don't want to record all the
Queue
calls. I just want to record calls connected with some specific members.
--AM
On Thu, May 5, 2011 at 11:10 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On
On Thu, May 5, 2011 at 11:02 PM, Matt Riddell li...@venturevoip.com wrote:
On 6/05/11 3:14 PM, Ira wrote:
At 03:00 PM 5/5/2011, you wrote:
Yes, but in my world there is one Atom powerd Linux box running
Asterisk, 4 or 5 Windows machines and 2 Macs. If I want to test, it has
to be on my
I have an Audiocodes MP-114 and modem dial-out can not establish connection.
Snooping with Wireshark I get a lot of entries like this:
No. TimeSourceDestination Protocol Info
2934 27.221559 10.0.0.11010.0.0.109RTP PT=ITU-T
Further,
Before I start working on full project, I wanted to test the functionalities
to be implemented. So I wrote a small test dialplan to check whether I can
record a Queue call in Macro which gets executed on Member answer. My actual
macro would be like this...
[macro-agntanserd]
exten =
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