Tilghman Lesher wrote:
On Saturday 08 August 2009 05:26:40 Suzuki Hironobu wrote:
Thomas Kenyon wrote:
uzuki Hironobu wrote:
Hi,
I am a beginner who began to use Asterisk in this July.
Last week,
I made two addons for PostgreSQL (cdr_addon_postgresql.c and
Hi,
I am a beginner who began to use Asterisk in this July.
Last week,
I made two addons for PostgreSQL (cdr_addon_postgresql.c and
res_config_postgresql.c),
because I use not usual MySQL but PostgreSQL.
# Of course, not scratch build but modified version 1.4.8.
But I don't know how to change
uzuki Hironobu wrote:
Hi,
I am a beginner who began to use Asterisk in this July.
Last week,
I made two addons for PostgreSQL (cdr_addon_postgresql.c and
res_config_postgresql.c),
because I use not usual MySQL but PostgreSQL.
Err, cdr_pgsql and res_config_pgsql are part of the main
Thomas Kenyon wrote:
uzuki Hironobu wrote:
Hi,
I am a beginner who began to use Asterisk in this July.
Last week,
I made two addons for PostgreSQL (cdr_addon_postgresql.c and
res_config_postgresql.c),
because I use not usual MySQL but PostgreSQL.
Err, cdr_pgsql and res_config_pgsql are
On Saturday 08 August 2009 05:26:40 Suzuki Hironobu wrote:
Thomas Kenyon wrote:
uzuki Hironobu wrote:
Hi,
I am a beginner who began to use Asterisk in this July.
Last week,
I made two addons for PostgreSQL (cdr_addon_postgresql.c and
res_config_postgresql.c),
because I use not
Hi,
I have asterisk running a single T1 card with a connection to a nortel
CS 1000.
All calls to extensions, local and long distance are working just fine.
My issue is this: The nortel CS 1000 supports connections to and
intercom system
that is just line level audio to speakers.
When my PRI
Hi,
Is the Paging on the Nortel connected to a trunk port or is there some
device connected as a physical extension that answers the call?
Hopefully you have access to the nortel box.
I am going to assume that you are not very familiar with the nortel
command interface. If I am incorrect, I
Hello,
I just installed asterisk 1.6.1.1, asterisk-addons 1.6.1.1 and Asterisk-GUI
2.0 on Ubuntu 9.04 from *source.
*Asterisk-GUI's web interface doesn't appear from this URL
http://localhost:8088/asterisk/static/config/index.html*. *It says
Not Found
The requested URL was not found on this
Dear Asterisk community!
I am having trouble with a project concerning the 183 Session Progress SIP
messages. Asterisk seems to only accept these when there is also a Session
Description (SDP) included in the message.
I also verified this by looking at the code.
However for a project we
Floimair Florian schrieb:
I am having trouble with a project concerning the 183 Session Progress SIP
messages. Asterisk seems to only accept these when there is also a Session
Description (SDP) included in the message.
I also verified this by looking at the code.
Which version of
: [asterisk-users] Question about core CDR system for multilpe
servers
Gustavo A Gonzalez escribió:
Hi all! I’m not sure if it is the correct place but, I’ve five boxes
running
asterisk and each one with his own cdr mysql database. What Im looking for
is to get a core CDR system
Hi all! Im not sure if it is the correct place but, Ive five boxes running
asterisk and each one with his own cdr mysql database. What Im looking for
is to get a core CDR system that holds information stored on each asterisk
server. Have you any suggestion/process to accomplish that?. Thanks!!!
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gustavo A
Gonzalez
Sent: Thursday, June 04, 2009 10:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about core CDR system for multilpe
: [asterisk-users] Question about core CDR system for multilpe
servers
Hi all! I?m not sure if it is the correct place but, I?ve five boxes running
asterisk and each one with his own cdr mysql database. What Im looking for
is to get a core CDR system that holds information stored on each asterisk
server
, 2009 10:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about core CDR system for multilpe
servers
Hi all! I?m not sure if it is the correct place but, I?ve five boxes running
asterisk and each one with his own cdr mysql database. What Im looking for
is to get
Gustavo A Gonzalez escribió:
Hi all! I’m not sure if it is the correct place but, I’ve five boxes running
asterisk and each one with his own cdr mysql database. What Im looking for
is to get a core CDR system that holds information stored on each asterisk
server. Have you any
...@lists.digium.com] On Behalf Of Miguel Molina
Sent: Thursday, June 04, 2009 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about core CDR system for multilpe
servers
Gustavo A Gonzalez escribió:
Hi all! Im not sure
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gustavo A
Gonzalez Sent: Thursday, June 04, 2009 10:23 AM
Hi all! I?m not sure if it is the correct place but, I?ve five boxes
running asterisk and each one with his own cdr mysql database. What Im
looking for
Hi
My setup is
Internet - firewall - asteriskbox
- spa3102a
- spa3102b
the spa's can talk to the firewall directly. The firewall does NAT.
The current asterisk flow for outgoing calls is
phone = spa3102 = asterisk = vsp
and vis versa for inbound
I need to obtain one variable in the dialplan containing the IP address that
Asterisk is using, I mean, the originating IP for any calls coming out of
Asterisk via SIP. Is this possible?
F.Alves
___
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Call AGI script (Bash, Perl, whatever) that parses the value out of
'ifconfig' (or somewhere in /proc, routing table, whatever) and sets
it as a channel variable before returning.
I need to obtain one variable in the dialplan containing the IP address
that
Asterisk is using, I mean, the
Subject : Re: [asterisk-users] Question
---
You may have better luck with a better Subject.
On Tue, 19 May 2009, Venefax wrote:
I need to obtain one variable in the dialplan containing the IP address
that Asterisk is using, I mean, the originating IP
Hi,
After following the messages to install flite, I can find the following files.
/usr/lib/asterisk/modules/app_flite.so
/etc/asterisk/flite.conf
That's mean flite is installed successfully. Then I restart asterisk
but nothing found for that module.
sip*CLI core show application flite
Your
Hello,
I am using an SPA3102, all is working with asterisk 1.4, voice mail,
outbound calling etc, and it even passes the cid name/num to my analog
phone. However, when someone is calling me, I hear the beeps but the
caller-id information is not showing up on my phone, is this an SPA3102
Hello Justin,
You can try with a softphone first.
Good luck.
On Thu, Apr 30, 2009 at 6:37 PM, Justin Piszcz jpis...@lucidpixels.comwrote:
Hello,
I am using an SPA3102, all is working with asterisk 1.4, voice mail,
outbound calling etc, and it even passes the cid name/num to my analog
2009/4/30 Justin Piszcz jpis...@lucidpixels.com
Hello,
I am using an SPA3102, all is working with asterisk 1.4, voice mail,
outbound calling etc, and it even passes the cid name/num to my analog
phone. However, when someone is calling me, I hear the beeps but the
caller-id information is
Hey All -
I've got an interesting problem, here is what I'm trying to accomplish:
Six agents, two queues, three skill levels
Queue A (queue B is the same)
- Level 1
-- Agent 1
-- Agent 2
- Level 2
-- Agent 3
-- Agent 4
- Level 3
-- Agent 5
-- Agent 6
I'd like a call to come in to Queue
: Thursday, March 19, 2009 2:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question regarding call queue and penalty's
Hey All -
I've got an interesting problem, here is what I'm trying to accomplish:
Six agents, two queues, three skill levels
-users] Question regarding call queue and penalty's
Hey All -
I've got an interesting problem, here is what I'm trying to accomplish:
Six agents, two queues, three skill levels
Queue A (queue B is the same)
- Level 1
-- Agent 1
-- Agent 2
- Level 2
-- Agent 3
-- Agent 4
asterisk-users@lists.digium.com
Date: Thu, 19 Mar 2009 16:19:03 -0400
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Question regarding call queue and penalty's
Ahh - so use three queues and not one queue with three penalties
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question regarding call queue and penalty's
Ahh - so use three queues and not one queue with three penalties?
On Thu, Mar 19, 2009 at 4:04 PM, Danny Nicholas da...@debsinc.com wrote:
Wouldn't this work?
Exten = s,1
Hi,
I built a FARM OF ASTERISKs split into 3
geographically dispersed sites (for high
aggregate bandwidth concerns). There were 60
machines in total. All of them Dual Xeon 3.0
with 2GB. [ This turned out to be way more CPU
that I needed.] Each machine had 18~24 separate
conference
2009/3/6 BERGANZ François franc...@acropolistelecom.net
hello,
I will do a server to do a lots of conferences (MeetMe).
I want to know that if I dont use a digum card, the limit of simultaneous
calls is harder without a card than with a card ?if, yes, how harder is the
limit?
the transcoding card isnt a good source for timing. the card only make
interruptions if it is working.
if the meetme dont requeire transcoding the card will not generate any
timing.
David
2009/3/6 Grygoriy Dobrovolskyy megaho...@gmail.com
2009/3/6 BERGANZ François
Just a silly question that I'm not sure.
Ringinuse is working with IAX in 1.6??? like in sip??
Thanks!
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options
Sebastian wrote:
Just a silly question that I’m not sure.
Ringinuse is working with IAX in 1.6??? like in sip??
I assume you're referring to the queues.conf option, correct? An easy way to
check is to issue a queue show command when an IAX2 queue member receives a
call. If his status is
hello,
I will do a server to do a lots of conferences (MeetMe).
I want to know that if I dont use a digum card, the limit of simultaneous
calls is harder without a card than with a card ?if, yes, how harder is the
limit?
thank you
Cordialement,
BERGANZ François
P Pensez à
Hi all,
I have a TDM2400 full of FXS cards and a TDM800 full of FXO cards.
If I have a fax machine on the FXS port dialing out through asterisk
on the TDM800 FXO, should I be expecting any problems?
Or should this just work as expected? (ie, flawlessly with the
asterisk box essentially
2009/3/4 Mikel Lindsaar raasd...@gmail.com
Hi all,
I have a TDM2400 full of FXS cards and a TDM800 full of FXO cards.
If I have a fax machine on the FXS port dialing out through asterisk
on the TDM800 FXO, should I be expecting any problems?
I think you should get problems for faxing.
If
On Thu, 26 Feb 2009, Haim Dimer wrote:
Hello,
Some of my users have phones lacking a DND button. I need to provide
an extension they can dial that will put them in DND, i.e. tell the
server not to send them any calls until they get off the DND.
I've researched it for almost 3 days now and
Thank you Gordon and Alexander. With your help, I got it working like
so:
[app-dnd-on]
exten = *78,1,Answer
exten = *78,n,NoOp(${CALLERID(num)} is going on DND ACTIVE)
exten = *78,n,Set(DB(DND/${CALLERID(num)})=On)
exten = *78,n,Playback(do-not-disturbactivated)
exten = *78,n,Hangup
Hello,
Some of my users have phones lacking a DND button. I need to provide
an extension they can dial that will put them in DND, i.e. tell the
server not to send them any calls until they get off the DND.
I've researched it for almost 3 days now and tried a range of
configurations. I'm
...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Haim Dimer
Sent: Thursday, February 26, 2009 6:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about Do Not Disturb
Hello,
Some of my users have phones lacking a DND button. I need to provide
Hey List,
Anyone know the correct way to override an announcement on a queue by queue
basis?
My goal is to have one of my queues say press one to blah.. and no
position announcements I have the jump from queue context working (the
press 1) I just need the correct message played to the user
Christopher Aloi wrote:
Hey List,
Anyone know the correct way to override an announcement on a queue by
queue basis?
My goal is to have one of my queues say press one to blah.. and no
position announcements I have the jump from queue context working (the
press 1) I just need the
Mark Michelson wrote:
Christopher Aloi wrote:
Hey List,
Anyone know the correct way to override an announcement on a queue by
queue basis?
My goal is to have one of my queues say press one to blah.. and no
position announcements I have the jump from queue context working (the
press
Here's the version -
Asterisk SVN-branch-1.4-r143404
Just static queues.
Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/ dir
for these queue announce files? So my custom file should live in that dir
right?
Thanks for the help :)
On Tue, Feb 17, 2009 at 1:05 PM,
Christopher Aloi wrote:
Here's the version -
Asterisk SVN-branch-1.4-r143404
Just static queues.
Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/
dir for these queue announce files? So my custom file should live in
that dir right?
Thanks for the help :)
Yah - Found my problem, I can't spell -
periodic-*annouce* = SD-PLS-HOLD
periodic-announce-frequency=10
: )
On Tue, Feb 17, 2009 at 1:19 PM, Christopher Aloi chris.a...@gmail.comwrote:
Here's the version -
Asterisk SVN-branch-1.4-r143404
Just static queues.
Is it true that Asterisk
Christopher Aloi wrote:
Yah - Found my problem, I can't spell -
periodic-*annouce* = SD-PLS-HOLD
periodic-announce-frequency=10
: )
Oh, Ha! That'll do it every time.
Mark Michelson
___
-- Bandwidth and Colocation Provided by
, 2009 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question regarding custom announcements
inqueues.conf
Here's the version -
Asterisk SVN-branch-1.4-r143404
Just static queues.
Is it true that Asterisk looks in the default /var/lib/asterisk
I have outgoing call files working. I am trying to get the manager to
originate a call.
My outgoing call file that works looks like:
Channel: SIP/devcentos5x64_to_panel/mediaport
SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav
SetVar: agi_pa_recno=1725
Context: smvoice-dialout
Jerry Geis wrote:
I have outgoing call files working. I am trying to get the manager to
originate a call.
My outgoing call file that works looks like:
Channel: SIP/devcentos5x64_to_panel/mediaport
SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav
SetVar: agi_pa_recno=1725
Context:
On Wed, Feb 4, 2009 at 7:40 PM, Jerry Geis ge...@pagestation.com wrote:
Seems like the first call to Channel is being MADE successfully.
Then it goes to do Context and Exten: I get failed...
[smvoice-dialout]
exten = smvoice_single_mediaport,1,agi(smvoice)
exten =
Jerry Geis wrote:
Jerry Geis wrote:
I have outgoing call files working. I am trying to get the manager to
originate a call.
My outgoing call file that works looks like:
Channel: SIP/devcentos5x64_to_panel/mediaport
SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav
SetVar:
Is there an ATA type device out there that has low level audio out for
connecting speakers?
My asterisk server is in one building, I wish to have speakers in
another building
and connect them up to a low level audio device that I can call into and
speak.
Can I connect speakers into the FXS or
-users@lists.digium.com
Subject: [asterisk-users] question on connecting speakers
Is there an ATA type device out there that has low level audio out for
connecting speakers?
My asterisk server is in one building, I wish to have speakers in
another building
and connect them up to a low
On Mon, 22 Dec 2008, Jerry Geis wrote:
Is there an ATA type device out there that has low level audio out for
connecting speakers?
My asterisk server is in one building, I wish to have speakers in
another building
and connect them up to a low level audio device that I can call into and
We sell BT200's premodified for this exact purpose. We install an RCA/Phono
jack on the back so you can connect the phone directly to your overhead paging
equipment. They also have a toggle switch for moving between the chassis
speakerphone and the external jack. We're currently using these in
hi
there is only one king of queue, just queue
if you want callback you shoulndt use agentcallbacklogin because is
deprecated.
you should use queueaddmember() or something like that.
David
2008/12/6 Mike [EMAIL PROTECTED]
Hello,
I'm trying to setup a very simple queue with 5 SIP
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on queue terms
hi
there is only one king of queue, just queue
if you want callback you shoulndt use agentcallbacklogin because is
deprecated.
you should use queueaddmember() or something like
Mike wrote:
Thanks. I know agentcallbacklogin is deprecated, but I am not even
sure if I need anything special, I can`t find a clear answer. All the
queues example I find are ones where the agent have to login. I
simply need to have a queue that rings 5 SIP phones according to the
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on queue terms
Mike wrote:
Thanks. I know agentcallbacklogin is deprecated, but I am not even
sure if I need anything special, I can`t find a clear answer. All the
queues example I find are ones where the agent
Lytle
Sent: Sunday, December 07, 2008 12:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on queue terms
Mike wrote:
Thanks. I know agentcallbacklogin is deprecated, but I am not even
sure if I need anything special, I can`t find
Hello,
I'm trying to setup a very simple queue with 5 SIP phones. I do NOT want
the agents to have to be on the phone to get calls, but I want those 5 SIP
phones to ring (according to the strategy chosen in queue.conf) to dispatch
calls.
Is this a call back queue, or is a callback queue a
Hi.
Probably you should use SS7.
It depends on your hardware.
On Wed, Nov 19, 2008 at 8:44 AM, mark morreny [EMAIL PROTECTED] wrote:
Hi Andrew,
Thank you for your info. I am actually looking for connecting mobile base
station with asterisk via E1.
Any idea on where I should start
On Nov 18, 2008, at 7:30 PM, mark morreny wrote:
Hi,
Is it possible to connect Asterisk with a mobile base station to
handle call switching? What kind of protocol will I need to use to
convert to sip?
Any pointer or info will be greatly appreciated.
Best Regards,
Mark
I've got a
Hi,
Is it possible to connect Asterisk with a mobile base station to handle call
switching? What kind of protocol will I need to use to convert to sip?
Any pointer or info will be greatly appreciated.
Best Regards,
Mark
___
-- Bandwidth and
On Tue, Nov 18, 2008 at 22:30, mark morreny [EMAIL PROTECTED] wrote:
Hi,
Is it possible to connect Asterisk with a mobile base station to handle call
switching? What kind of protocol will I need to use to convert to sip?
Any pointer or info will be greatly appreciated.
There are various
Hi Andrew,
Thank you for your info. I am actually looking for connecting mobile base
station with asterisk via E1.
Any idea on where I should start looking?
Thanks,
Mark
On Wed, Nov 19, 2008 at 1:03 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote:
On Tue, Nov 18, 2008 at 22:30, mark morreny
Unfortunately RealTime isn't going to be an option - it's another level
of configuration I want to avoid, but more importantly since I'm
planning on being able to run these scripts on an Astlinux install,
there won't always be a MySQL database available. If worst comes to
worst, and the extra
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rob Hillis wrote:
Unfortunately RealTime isn't going to be an option - it's another level
of configuration I want to avoid, but more importantly since I'm
planning on being able to run these scripts on an Astlinux install,
there won't always be
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rob Hillis wrote:
Unfortunately RealTime isn't going to be an option - it's another level
of configuration I want to avoid, but more importantly since I'm
planning on being able to run these scripts on an Astlinux
Historically Asterisk's config file parser ignored unknown keywords.
This is useful for exactly the things you are trying to do. I hope 1.6
did not remove this feature.
Rob Hillis wrote:
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rob Hillis wrote:
Hi guys,
I'm about to embark on a small (undoubtedly to get much larger) project
to write a set of scripts to handle provisioning of phones - Snom to
begin with, possibly with others (most likely Polycom and Linksys) to
follow later. Since I want this script to handle *all* aspects of phone
It should ignore the keywords, but you will get lots of errors in the CLI.
My guess is that if you put it all in a DB (and use realtime) you can
probably do whatever you want.
PaulH
Rob Hillis wrote:
Hi guys,
I'm about to embark on a small (undoubtedly to get much larger) project
to
2008/10/13 Tilghman Lesher [EMAIL PROTECTED]
snip
Pray tell, how do you echo cancel in both directions? Wouldn't that
necessitate cancelling echo before it occurs on the line (sort of a white
noise/pink noise kind of operation)? Seems like modelling a projectile
such
that when it reaches
On Tuesday 14 October 2008 02:10:39 Olivier wrote:
2008/10/13 Tilghman Lesher [EMAIL PROTECTED]
snip
Pray tell, how do you echo cancel in both directions? Wouldn't that
necessitate cancelling echo before it occurs on the line (sort of a white
noise/pink noise kind of operation)?
2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED]
snip
Where did you hear that media gateways filter one-way only?
Hi ,
Reading over this reply, it seems to me that having EC working in one
direction is not a so well known fact.
Could we say :
1. EC working in one direction is the
Just a note that deserves to be reminded is
http://lists.digium.com/pipermail/asterisk-dev/2006-January/017774.html
If Generally speaking there is only one direction of echo cancellation
needed is true, EC works in one way ...
___
-- Bandwidth and
This one is also a must-read
http://lists.digium.com/pipermail/asterisk-dev/2007-May/027536.html
except that is the following scheme, I'm wondering if arrows and RX/TX
legends are coherent ...
What is written :
TX * --- *
TDM X Y
RX + --- -
What I would write :
RX
On Sunday 12 October 2008 04:15:02 Olivier wrote:
2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED]
Handsets use a 4-wire connection. Handsets with the the volume turned
up could cause a form of echo as the microphone picks up the ear piece
audio (I call this acoustic echo). Everything
2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED]
Handsets use a 4-wire connection. Handsets with the the volume turned
up could cause a form of echo as the microphone picks up the ear piece
audio (I call this acoustic echo). Everything I said applies to 2-wire
caused echo. Other types
2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED]
All calls with a 2-wire analog piece have echo. You cannot perceive the
echo because it happens so fast on non-VoIP connections. On VoIP calls
you have significant extra latency while causes you you to perceive the
echo.
Do you mean
Olivier wrote:
2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED]
All calls with a 2-wire analog piece have echo. You cannot perceive the
echo because it happens so fast on non-VoIP connections. On VoIP calls
you have significant extra latency while causes you you to perceive the
echo.
2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED]
Olivier wrote:
2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED]
All calls with a 2-wire analog piece have echo. You cannot perceive the
echo because it happens so fast on non-VoIP connections. On VoIP calls
you have significant
Handsets use a 4-wire connection. Handsets with the the volume turned
up could cause a form of echo as the microphone picks up the ear piece
audio (I call this acoustic echo). Everything I said applies to 2-wire
caused echo. Other types of echo is fairly uncommon and cannot be
solved by
Hi,
I'm using the following setup :
Alice IPPhone --LAN- Media gateway PSTN --- Phone
Bob
For certain calls, users complains about echo : they can ear their own voice
in their handset, though media gateway echo cancel is turned on.
I'm wondering how this echo
All calls with a 2-wire analog piece have echo. You cannot perceive the
echo because it happens so fast on non-VoIP connections. On VoIP calls
you have significant extra latency while causes you you to perceive the
echo. Echo must be removed before the call is converted to VoIP -- in
your
On Thu, Oct 9, 2008 at 6:38 AM, C. Savinovich
[EMAIL PROTECTED] wrote:
I am tinkering with a new router, a Linksys wrt610n dual-band, etc. But
the when I connect it, the softphones(x-lite) on the computers don't even
register. After a couple of hours of hassle, I found out that if I dmz the
I am tinkering with a new router, a Linksys wrt610n dual-band, etc. But
the when I connect it, the softphones(x-lite) on the computers don't even
register. After a couple of hours of hassle, I found out that if I dmz the
router to the computer I am using, the softphone starts to work.
Hi,
I have a simple desire to be able to screen people before being onnected to
them. I`ve seen plenty of examples on the web and I`ve figured it out.
There is only one case in where it doesnt act as I want it to: if I hang up
the phone, I don`t want the caller to be disconnected but (for the
Mike wrote:
So, I guess my question is: how do I set a variable that ISN`T lost
when the call initiated using the Dial g option is hung up ?
You can use the internal database for that:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+db
Doug
--
Ben Franklin quote:
Those
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on screening calls / Question about
the Dial g option
Mike wrote:
So, I guess my question is: how do I set a variable that ISN`T lost
when the call initiated using the Dial g option is hung up ?
You can use
Mike wrote:
Doug,
Thanks for the quick answer. How does that help me though, since this is a
per channel variable and not a global variable?
Make sure your key in the database is specific to only that call. Time,
date, caller-id number or even a combination of all.
Can't you save your
On Tue, Oct 7, 2008 at 4:36 PM, Mike [EMAIL PROTECTED] wrote:
Hi,
I have a simple desire to be able to screen people before being onnected to
them. I`ve seen plenty of examples on the web and I`ve figured it out.
There is only one case in where it doesn't act as I want it to: if I hang up
Hello there.
I have a problem that I can't solve. I am developing an application
with Java and Asterisk. In addition, I am using Windows Vista, AsteriskWin32
PBX, asterisk-java-0.3.jar and XLite. I startup the DefaultAgiServer without
problems and I have a java application running for
Ext. 221
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Santiago
Panchi
Sent: Tuesday, September 30, 2008 10:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about Asterisk and Java
] *On Behalf Of *Santiago Panchi
*Sent:* Tuesday, September 30, 2008 10:14 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Question about Asterisk and Java
Hello there.
I have a problem that I can't solve. I am developing an
application with Java and Asterisk
Venefax wrote:
I need to dial a DTMF string with the Dial function using the D(“DTMF”)
function. What is the character for a delay? I mean, normally in other
technologies we use the comma to mean “wait 200 ms “. Is there an
equivalent in Asterisk? If it is the comma indeed, how many ms
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