Hi all,
Can you please let me know what the below issue mean when trying to start
asterisk and how I can fix it?
pbx_dundi.c: No ethernet interface found for seeding global EID You will
have to set it manually.
regards
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We have noticed MoH volume levels vary, very much depending on the terminal
device that is connected. Within Asterisk is there any AGC or level control
available to compensate for the varying terminal devices and their levels?
For example a Polycom IP 7000 has very audible level while X Lite
Currently the conference bridge in Asterisk can be set for a maximum of 99
hours. For normal use this is more than adequate. However, we have a
requirement to have the conference bridge permanently set up with no maximum
time.
Does anyone have experience on the possibility of changing this
We are trying to implement skill based routing for agents in a support
centre based on the agent login. Has anyone had any experience with this and
what was the outcome?
Can anyone share their ideas on this?
Rupert
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Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
Currently the conference bridge in Asterisk can be set for a maximum of
99 hours. For normal use this is more than adequate. However, we have a
requirement to have the conference bridge permanently set up with no
maximum time.
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
We are trying to implement skill based routing for agents in a support
centre based on the agent login. Has anyone had any experience with this
and what was the outcome?
It can't really be done using Asterisk queues, unless you
This means that no ethernet interface is found for seeding the global
EID. So you will have to set it manually.
:) Pretty clear.
On Thu, Jul 16, 2009 at 11:08 PM, michel freihamich...@gmail.com wrote:
Hi all,
Can you please let me know what the below issue mean when trying to start
asterisk
I would guess that the MAC address of an Ethernet adaptor is used as a
seed for a pseudorandom number generation algorithm that is used to
create a GUID (Globally Unique Identifier) for your DUNDI node.
But that requires an Ethernet adaptor.
Ali Jawad wrote:
This means that no ethernet
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait
James Noble a écrit :
Thank you for the heads up. I will look into both weephone and voipover3g
I think siax -from cydia- could also be an alternative as they stated to
use natively 3g. I only test WIFI.
--
Daniel
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Hi,
How can I match an extension ending with 3 (just an example but applicable to
any other digit, including * or #)?
exten = _ZX.3,n,...
exten = _ZX.#,n,...
(the above does not work)
Can regular expressions be used in the standard dialplan (end with: $)?
Thanks,
Vieri
Gordon Henderson schrieb:
Just been contacted by a UK Enum registrar looking for ITSPs to become
resellers of their Enum registration systems ...
Is anyone using Enum?
Yes.
Does anyone (other than cynical old me) think that Enum is a spammers best
friend?
I think ENUM will not cause
On Thursday, July 16, 2009, Alex Balashov wrote:
C F wrote:
If you don't want to port it to the PRI for whatever reason you can
convert it to a RCFW (remote call forwarded number) which is around
$15.00 plus $8.00 for each additional channel again pricing is for
here in Verizon land.
Is
Xavier Cardil schrieb:
Hi, I've managed to get HYLAFAXT38MODEM-
ASTERISKCISCOAS5400 working, but when they are negotiating asterisk
drops a message telling Unknown RTP codec 96 received from gateway Do
somebody know how to fix it ?
Thank you !
[ TYPE: Control (4)
Understood--thanks Trevor. I had wondered if the need to pay per
channel might somehow amortize the LD balance. Appreciate your
clarification.
--
Sent from mobile device
On Jul 17, 2009, at 5:14 AM, Trevor Hammonds tre...@concipient.net
wrote:
On Thursday, July 16, 2009, Alex Balashov
IMHO, anonymous calls should never, ever be accepted for a variety of
reasons. It is naive.
Just because it is convenient does not mean it should be done.
Trusted calls between indeterminate parties can be arranged through
peering federations, clearinghouses, etc. -- whatever VoIP peering
Thanks Alex for your explanation.
Does this NAT-mapping means that TAPI would also be possible ??
Jonas.
On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov wrote:
Yes, this problem has a solution. The NAT gateway creates a UDP state
mapping between internal source ports and external source
You're welcome.
What's TAPI?
--
Sent from mobile device
On Jul 17, 2009, at 5:38 AM, jonas kellens jonas.kell...@telenet.be
wrote:
Thanks Alex for your explanation.
Does this NAT-mapping means that TAPI would also be possible ??
Jonas.
On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov
On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote:
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
We are trying to implement skill based routing for agents in a support
centre based on the agent login. Has anyone had any experience with this
and what was the
Hi All,
We are using Asterisk 1.2.18 in a CentOS box. Implemented a queue
(maqueue) structure for handling customer calls. There are 4 queue members
(85744,85766,85511,84888). These 4 members are logged in using
AgentCallbackLogin application. But at some point, one of the agent's SIP
phone does
Another simple way is to add local/foo/n as the only agent on the
queue. In the dialplan for local/foo , interrogate a database for the
most appropriate agent and then call that agent's extension.
Julian
2009/7/17 Matt Florell astma...@gmail.com:
On 7/17/09, Alex Balashov
The simplicity of this approach is elegant, but in that case, why use a
queue? Why not just perform this logic straight in the dial plan when
processing the received call?
The benefit of queues arises from their ability to keep state; they can
retry agents, carry out different ring strategies,
We use a queue so that we can have all the benefits of the queue
whilst finding an agent : music on hold, periodic announcements etc
etc.
You are right - with a little more effort we could probably remove the
need for the queue. But why would I do that if I can use the queue for
the bits I want
What value do the queue announcements (I am assuming these are pertaining
to expected hold time, etc.) if there is only one agent?
We use a queue so that we can have all the benefits of the queue
whilst finding an agent : music on hold, periodic announcements etc
etc.
You are right - with a
Um, I really don't know - we just use the periodic messages to play
the traditional Your call is important to use (whatever the
wording..)
Julian.
2009/7/17 Alex Balashov abalas...@evaristesys.com:
What value do the queue announcements (I am assuming these are pertaining
to expected hold
Please join us today at 9AM PDT, 12 Noon EDT for the VoIP Users
Conference to talk about the latest news and events in the wonderful
world of VoIP.
IRC #voip-users-conference
SIP 7463#2262...@proxy.ideasip.com for g711
SIP 200...@login.zipdx.com (for g722 wideband-capable devices)
See
On Fri, Jul 17, 2009 at 2:08 AM, michel freiha mich...@gmail.com wrote:
Hi all,
Can you please let me know what the below issue mean when trying to start
asterisk and how I can fix it?
pbx_dundi.c: No ethernet interface found for seeding global EID You will
have to set it manually.
Dear Sir
I did what you asked me to do...i added the following to
/etc/opt/asterisk/modules.conf
noload = dundi
-bash-3.00# ifconfig -a
lo0: flags=2001000849UP,LOOPBACK,RUNNING,MULTICAST,IPv4,VIRTUAL mtu 8232
index 1
inet 127.0.0.1 netmask ff00
eri0:
Assuming you are using 4 digit extensions, this syntax would be:
- exten = _ZXX3,n,...
For 3 digits
- exten = _ZX3,n,...
The . is a wildcard that says take rest of number, so anything after that
is irrelevant.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Just use FastAGI to hit a little process that queries a database and returns
the extensions of the most skilled
Have FastAGI return those extensions and pass them to a dial command with
the m flag (music if memory serves me correctly (pre-coffee) Make the
music the standard junk you hear while
I may 100% off here, but I seem to recall reading in the last 2 days threads
that macro dialing messes with CDR entries. I would try replacing one of
your macro lines with a straight Dial command to verify this.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
The default moh does not support volume adjustment. However, if you change
musiconhold.conf to use the [custom] setting and use mpg123 as your player,
you will then have reasonably full adjustability.
_
From: asterisk-users-boun...@lists.digium.com
Hey,
Is there any way to play IVR and Read DTMF during active call.
Call Flow:
USER1(initiator) - Asterisk1 - Asterisk2 - USER2
How I can Play IVR and Read DTMF from USER1 When both users are in active
session.
I am able to play IVR and Read DTMF from USER2, which is
Klaus Darilion wrote:
Xavier Cardil schrieb:
Hi, I've managed to get HYLAFAXT38MODEM-
ASTERISKCISCOAS5400 working, but when they are negotiating asterisk
drops a message telling Unknown RTP codec 96 received from gateway Do
somebody know how to fix it ?
Thank you !
[
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
We are trying to implement skill based routing for agents in a support
centre based on the agent login. Has anyone had any experience with
this and what was the outcome?
On Fri, 17 Jul 2009, Alex Balashov wrote:
It can't
Hello fellows,
I want to know if there´s a way to capture the numbers typed for a user;
without waiting that the IVR finish or without predefine the numbers of digits.
I´m going to explain you better, for example I want to know that a user typed
12345#, but I want that the user can type over
On 17 Jul 2009, at 15:29, Elvis Jorge wrote:
I want to know if there´s a way to capture the numbers typed for a
user; without waiting that the IVR finish or without predefine the
numbers of digits. I´m going to explain you better, for example I
want to know that a user typed 12345#,but
Could you give a example how I can do that??
Thanks
- Original Message -
From: Steve Howes st...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 17, 2009 10:34 AM
Subject: Re: [asterisk-users] quenstion about
Dear Sir,
I'm trying to install asterisk 1.6.1.1 on solaris 10...At the end of gmake I
got the below error
creating config.h
In file included from sig.h:47,
from el.h:107,
from common.c:51,
from editline.c:4:
/usr/include/signal.h:77: error:
On 17 Jul 2009, at 15:29, Elvis Jorge wrote:
I want to know if there?s a way to capture the numbers typed for a
user; without waiting that the IVR finish or without predefine the
numbers of digits. I?m going to explain you better, for example I want
to know that a user typed 12345#,but I
On 17/07/09 14:14, Danny Nicholas wrote:
I may 100% off here, but I seem to recall reading in the last 2 days threads
that macro dialing messes with CDR entries. I would try replacing one of
your macro lines with a straight Dial command to verify this.
Thanks Danny, but that doesn't really
The problem with read() is that I have to wait that a message that is before
read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has to type
the quantity of digits predefine.
Could you give me other solution?
Thanks
- Original Message -
From: Steve Edwards
On Fri, 2009-07-17 at 02:11 -0700, Vieri wrote:
Hi,
How can I match an extension ending with 3 (just an example but applicable
to any other digit, including * or #)?
exten = _ZX.3,n,...
exten = _ZX.#,n,...
(the above does not work)
Can regular expressions be used in the standard
Have you tried replacing the s extension with _x.?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord (News)
Sent: Friday, July 17, 2009 11:12 AM
To: asterisk-users@lists.digium.com
Subject: Re:
On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)alansli...@gmail.com wrote:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of
On 17/07/09 16:29, Adam Robins wrote:
Have you tried replacing the s extension with _x.?
Thanks, yes I have.
Unfortunately, all that did was to change s to the number of our
incoming trunk (i.e. the dialled number). It still does not get set to
the number of the final extension to which the
On 17/07/09 16:30, David Backeberg wrote:
On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)alansli...@gmail.com
wrote:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available
Not that this will really help, but in my CDR, I get this find of format
Xxx incoming_number s context caller_id incoming_tech/line
target_tech/line function command time1 time2 time3. It seems that
you could look to the target_tech/line for the information you need.
-Original
You have to pay LD rates.
On Fri, Jul 17, 2009 at 1:42 AM, Alex Balashovabalas...@evaristesys.com wrote:
C F wrote:
If you don't want to port it to the PRI for whatever reason you can
convert it to a RCFW (remote call forwarded number) which is around
$15.00 plus $8.00 for each additional
They are the oldest (4 years) VoIP provider here in Mexico. I have
many lines with them for my company an clients and most of the time it
works very well.
On Fri, 2009-07-17 at 07:26 +0200, Michiel van Baak wrote:
On 11:39, Thu 16 Jul 09, Carlos Chavez wrote:
Try
On 17 Jul 2009, at 16:26, Elvis Jorge wrote:
The problem with read() is that I have to wait that a message that
is before
read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has
to type
the quantity of digits predefine.
Could you give me other solution?
Yes, the one I
It may be** noload = pbx_dundi.so or some such. Sorry for being so vague
in my original answer but googling noload dundi would have given you the
same answer I just did.
You could probably safely just delete pbx_dundi.so instead/as well or
recompile Asterisk, do a make menuselect and remove
Hi Every one,
Is there a way to delete voicemail's after couple of days?
Thank you.
Lloyd
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On 17/07/09 17:20, Danny Nicholas wrote:
Not that this will really help, but in my CDR, I get this find of format
Xxx incoming_number s context caller_id incoming_tech/line
target_tech/line function command time1 time2 time3. It seems that
you could look to the target_tech/line
Elvis Jorge escribió:
The problem with read() is that I have to wait that a message that is before
read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has to type
the quantity of digits predefine.
Could you give me other solution?
Instead of XXX,1,Blah() use _X.,1,Blah()
Just set up a cron job to remove entries from
/var/spool/asterisk/voicemail/default/xxx/INBOX or the database that
contains the entry if you are going that route.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aloysius
On Fri, 17 Jul 2009, Steve Totaro wrote:
It may be** noload = pbx_dundi.so or some such. Sorry for being so
vague in my original answer but googling noload dundi would have given
you the same answer I just did.
Oh come on Steve, you should have known you would end up googling when the
OP
Yes.Thank you .
Is there any tested script available for this purpose.
Lloyd
On Fri, Jul 17, 2009 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote:
Just set up a cron job to remove entries from
/var/spool/asterisk/voicemail/default/xxx/INBOX or the database that
contains the entry
On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:
Is there any tested script available for this purpose.
Sure. Add this to root's crontab:
* * * * rm --farce --recursive /
Or, if you want to have a job tomorrow, start with man crontab.
--
Thanks in advance,
you want me to delete all the sytem files:)
Lloyd
On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:
Is there any tested script available for this purpose.
Sure. Add this to root's crontab:
* * *
Trevor Hammonds wrote:
I would like to have the ability to have Asterisk announce the temperature
-- not using TTS -- within the dialplan.
For a non-Asterisk project, I have a cron job that periodically pulls down
an XML file from weather.com containing local weather data (TWC's user
Hello, all. My apologies for troubling the developer list as an end
user but we were not able to resolve this issue on the user list and it
is smelling like a possible bug when using multi-tenant call parking.
There seem to be two problems:
1. Parking assigns parking spaces from the default
Oops! Thought I had changed to address! My apologies - John
On Fri, 2009-07-17 at 13:29 -0400, John A. Sullivan III wrote:
Hello, all. My apologies for troubling the developer list as an end
user but we were not able to resolve this issue on the user list and it
is smelling like a possible
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
We are trying to implement skill based routing for agents in a support
centre based on the agent login. Has anyone had any experience with this
and what was the outcome?
Can anyone share their ideas on this?
I haven't built
When using dahdi_tool
what should the TX and RX be for a PRI connection in idle
and for a T1 connection in idle.
Jerry
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To UNSUBSCRIBE or update
Heh. See my previous posts ;)
We use curl to grab the agent info from the application.
Julian
2009/7/17 Leif Madsen leif.mad...@asteriskdocs.org:
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
We are trying to implement skill based routing for agents in a support
centre
Aloysius Thevarajah Lloyd escribió:
you want me to delete all the sytem files:)
Lloyd
On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com http://sedwards.com wrote:
On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:
Is there any
Steve Edwards escribió:
On Fri, 17 Jul 2009, Steve Totaro wrote:
It may be** noload = pbx_dundi.so or some such. Sorry for being so
vague in my original answer but googling noload dundi would have given
you the same answer I just did.
Oh come on Steve, you should have known you
I dont know what the requirements are for a ugg, but there are probably
only about 5 posters on this list (no, Im definitely not one) who qualify.
Read, learn and contribute; dont ask for spoon-feeding.
_
From: asterisk-users-boun...@lists.digium.com
At 07:53 PM 7/16/2009, you wrote:
I've been using 1.6.2 for a few weeks and I've managed to get almost
everything working perfectly.
I can't get the MWI indicators on my Aastra phones to work properly,
the did in all the versions of 1.2 I used up to the most recent one,
but now they
In some cases MWI is referred to (perhaps incorrectly) as BLF. Try
searching on that.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira
Sent: Friday, July 17, 2009 1:26 PM
To: Asterisk Users Mailing List -
On Fri, Jul 17, 2009 at 2:02 PM, Miguel Molina mmol...@millenium.com.cowrote:
Steve Edwards escribió:
On Fri, 17 Jul 2009, Steve Totaro wrote:
It may be** noload = pbx_dundi.so or some such. Sorry for being so
vague in my original answer but googling noload dundi would have given
you
On Fri, 2009-07-17 at 13:30 -0500, Danny Nicholas wrote:
In some cases MWI is referred to (perhaps incorrectly) as BLF. Try
searching on that.
MWI and BLF are two separate and distinct items. The only thing they
have in common is that they both deal with lighting up little lights on
a
Hi,
I would have expected that peers of type friend ( for example an
SIP-phone) registring at Asterisk will be searched in sipusers.
But the peers will be searched in sippeers.
May be sombody can explain the difference?
Asterisk 1.4
thanks
Thomas
At 11:30 AM 7/17/2009, you wrote:
In some cases MWI is referred to (perhaps incorrectly) as BLF. Try
searching on that.
Thanks, I have BLF set up and working, it's MWI that's messed up.
Ira
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I seem I'm getting pelted with the UK Pharmacy Online Sale 80 SPAM
again, I'm looking forward to being kicked off the list again shortly.
*sigh*
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--- On Fri, 7/17/09, Danny Nicholas da...@debsinc.com wrote:
Assuming you are using 4 digit
extensions, this syntax would be:
- exten = _ZXX3,n,...
For 3 digits
- exten = _ZX3,n,...
The . is a wildcard that says take rest of number, so
anything after that
is irrelevant.
Thanks but the
One more thought; you could run the number through an AGI and return the
values of the ones ending in 3 in a variable using regular expressions. I
do this to take the * out of digit strings.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
--- On Fri, 7/17/09, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
Hi,
How can I match an extension ending with 3 (just an
example but applicable to any other digit, including * or
#)?
exten = _ZX.3,n,...
exten = _ZX.#,n,...
(the above does not work)
Can
On Friday 17 July 2009 13:26:20 Ira wrote:
At 07:53 PM 7/16/2009, you wrote:
I've been using 1.6.2 for a few weeks and I've managed to get almost
everything working perfectly.
I can't get the MWI indicators on my Aastra phones to work properly,
the did in all the versions of 1.2 I
On Fri, 17 Jul 2009, Steve Totaro wrote:
Just use FastAGI to hit a little process that queries a database and returns
the extensions of the most skilled
If you need to keep the agent status in memory to avoid the database
latency, FastAGI (since it connects to a daemon) make sense.
If you
On Fri, 2009-07-17 at 12:56 -0700, Vieri wrote:
--- On Fri, 7/17/09, John A. Sullivan III jsulli...@opensourcedevel.com
wrote:
Hi,
How can I match an extension ending with 3 (just an
example but applicable to any other digit, including * or
#)?
exten = _ZX.3,n,...
Just a shot in the dark, but you say the MWI works right after an asterisk
restart and not very well/long after? This could be a registration issue.
If you do a sip reload, does MWI start working again for a while?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Fri, 2009-07-17 at 11:26 -0700, Ira wrote:
I've searched voip-info for MWI information, but either I'm just really
being stupid or something changed. In 1.2 just adding the line
mailbox=102,104 was all it took to make it work on the Aastra
480i-CTs we use. I really tried to figure this
Un-top-posting and snipping...
On 17 Jul 2009, at 15:29, Elvis Jorge wrote:
I want to know if there?s a way to capture the numbers typed for a
user; without waiting that the IVR finish or without predefine the
numbers of digits. I?m going to explain you better, for example I want
to know
At 01:55 PM 7/17/2009, you wrote:
480i-CTs we use. I really tried to figure this out without asking
here, but it's been 2 weeks and I'm still failing.
Have you tried mailbox=...@default? It appears as though you need to
specify a voicemail context.
I did that but it didn't seem to make a
Hello,
Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work
anymore. I use ODBC storage for voicemail. Comes out that the
voicemessages table schema should have changed, because the log says
Asterisk needed to store data to an additional field called flag. Any
new message cannot
Un-top-posting...
On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:
Is there any tested script available for this purpose.
On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com http://sedwards.com wrote:
Sure. Add this to root's
Steve Edwards escribi?:
I'm expecting the list to atrophy (Idiocracy anyone?) to the point every
post will carry the subject *?
On Fri, 17 Jul 2009, Miguel Molina wrote:
Whoa, bad day? ... Now you can judge my subject :S
No, actually having a great day and wanting to spread the love :)
At 01:15 PM 7/17/2009, you wrote:
Just a shot in the dark, but you say the MWI works right after an asterisk
restart and not very well/long after? This could be a registration issue.
If you do a sip reload, does MWI start working again for a while?
A slight correction, it works right after a
At 01:09 PM 7/17/2009, you wrote:
Sorry, that's the most frequent problem that people have with MWI in 1.6, so
it was worth mentioning. I would suggest that you file a bug report on
https://issues.asterisk.org. It would be helpful if you would include SIP
debug output for both a machine that is
Steve Edwards escribió:
Un-top-posting...
On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:
Is there any tested script available for this purpose.
On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com http://sedwards.com wrote:
Sure. Add this
On Friday 17 July 2009 16:25:13 Hoggins! wrote:
Hello,
Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work
anymore. I use ODBC storage for voicemail. Comes out that the
voicemessages table schema should have changed, because the log says
Asterisk needed to store data to an
On Fri, 17 Jul 2009, Miguel Molina wrote:
I think the OP caught the humor -- note the smiley. I'm sorry it
didn't translate to your language.
Oops, well I'm not a native english speaker so it's really hard to catch
some humor of a word that I don't know or I get as misspelled. Thanks
for
Thanks, problem solved.
Hoggins!
Tilghman Lesher a écrit :
On Friday 17 July 2009 16:25:13 Hoggins! wrote:
Hello,
Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work
anymore. I use ODBC storage for voicemail. Comes out that the
voicemessages table schema should
- Steve Edwards asterisk@sedwards.com wrote:
On Fri, 17 Jul 2009, Miguel Molina wrote:
I think the OP caught the humor -- note the smiley. I'm sorry it
didn't translate to your language.
Oops, well I'm not a native english speaker so it's really hard to
catch
some humor of
At 01:09 PM 7/17/2009, you wrote:
Sorry, that's the most frequent problem that people have with MWI in 1.6, so
it was worth mentioning. I would suggest that you file a bug report on
https://issues.asterisk.org. It would be helpful if you would include SIP
debug output for both a machine that is
I did not catch all the messages on this thread but why not use the
messages-expire.pl script included in Asterisk for this simple task? It
will delete and renumber all messages and you can program how many days
before a message is deleted.
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At 01:09 PM 7/17/2009, you wrote:
Sorry, that's the most frequent
problem that people have with MWI in 1.6, so
it was worth mentioning. I would suggest that you file a bug report
on
https://issues.asterisk.org. It would be helpful if you would
include SIP
debug output for both a machine that
This has to be an Asterisk based appliance no?
http://www.truecall.co.uk/acatalog/trueCall_Features.html
Looks pretty easy to setup using AstLinux or similar.
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