2006/8/2, Andy Kuo [EMAIL PROTECTED]:
Hi,
Can you give a quick example on how to query an EXTERNAL database?
Create a AGI Script. It may take actual variable- and channel values
to create a query for the external DB. Store the result in a variable
to be used in the next dialplan priority.
On Wednesday, August 02, 2006 8:35 PM Vadim Berezniker wrote:
No idea, but DIALEDPEERNAME should contain the same value as
BRIDGEPEER. Try that.
Already did. Both are empty.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
Hi,
an all of my installations (1.2.9.1 bristuffed) the parameters BRIDGEPEER and
DIALEDPEERNAME are empty after a successful dial command. Can someone please
try to confirm this? I am not sure what I could have done to the implementation
to cause this so it might very well be a bug.
Thanks,
Thanks Paul ( ManxPower).
Are you saying though that there is no way to re-configure Asterisk to pick
up immediately or after one ring? Is this an Asterisk shortcoming or a
reality of analog phone lines?
I'm just wondering how many users are willing to have the caller wait an
additional 2 rings
At 11:23 PM 8/2/2006, you wrote:
I'm just wondering how many users are willing to have the caller wait an
additional 2 rings (in addition to the 3 audible rings for the Asterisk
receiving phone). This seems like something that should have some sort of
workaround. No?
You can disable callerid
On 8/2/06, Time Bandit [EMAIL PROTECTED] wrote:
The problem a number of people are not entering the pin fast enough
,they are not given enough time to enter the PIN( I assume this is a
mailbox number)
looking at all the doc is seems everything is configurable, can some
one point me in the
Ira ha scritto:
At 11:23 PM 8/2/2006, you wrote:
I'm just wondering how many users are willing to have the caller wait an
additional 2 rings (in addition to the 3 audible rings for the Asterisk
receiving phone). This seems like something that should have some sort of
workaround. No?
You
Hello
Im trying to decide whether or not I want to use IAX2 trunking on our WRAP
based customer computers.
As it only has a 200mhz processor, I want to make shure that the trunking part
does not affect call quality.
Does anybody know if trunking is more CPU intensive than non trunking?
Hi,Your Consultant has developed it with PHP scripts, so you must check those files in /var/lib/asterisk/agi-binYour application logic is there.Hope it helps,Best regards,Marco Mouta
On 8/3/06, Randy Paries [EMAIL PROTECTED] wrote:
On 8/2/06, Time Bandit [EMAIL PROTECTED] wrote: The problem a
On Thu, 3 Aug 2006 08:58:47 +0800
Hi Gavin,
This is the default of setting of the Asterisk. If you wish to adjust
the timing, please edit the source file of the Asterisk name call
chan_zap.c, And look for the static int matchdigittimeout line to
change the setting. The timing is in
Andrew Kohlsmith wrote:
Aside from using a Norstar ATA connected to an FXS port on Asterisk and
executing a hookflash *1, no. There isn't a really good way to do it, as
that is part of what keeps you hooked into their proprietary crap.
It's the same as trying to tie in SIP phones through
I'm having a problem where the very first words of the Asterisk voicemail
system prompt are distorted into a loud ear-splitting beep. When I dial my
VoiceMailMain extension I get this loud beep followed by the rest of the
initial voicemail system prompt. After that everything works fine. I've
Hi,
From the web (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue),
we have to set up 2 tables (queue and queue_member). I understand
table queue is used to define a queue and its related attribute. How
about the queue_member? I expect there will store queue member
information
Hi.
How can you check if a channel variable has been set in the dialplan?
I have some logic in the dialplan that must do a ExecIF() if a specific
channel variable has been set, otherwise it must do nothing.
Thank you very much.
___
--Bandwidth and
There is presently no .write member in the structure declaration for
this function in channels/app_sip.c:
static struct ast_custom_function sip_header_function = {
.name = SIP_HEADER,
.synopsis = Gets or sets the specified SIP header,
.syntax = SIP_HEADER(name),
Hi
What I can use with ASTERISK to call clients to remind them about their
appointments?
Thanks
Dmitri
smirnoff at aei dot ca
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
2006/8/3, unplug [EMAIL PROTECTED]:
Hi,
From the web (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue),
we have to set up 2 tables (queue and queue_member). I understand
table queue is used to define a queue and its related attribute. How
about the queue_member? I expect there
dmitri smirnoff [EMAIL PROTECTED] wrote:
What I can use with ASTERISK to call clients to remind them about their
appointments?
You can use Callfiles. Look how the wake-up script works:
http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+Wake-Up+Call+PHP
Also have a look at
Andrew:
The key here is try to create a way to integrate Asterisk Voicemail with
existing Meridian PBX and send the MWI to M2616.
I'm investigating the propietary protocol used by Nortel in the integration
between Octel-Dialogic and Meridian for MWI. I believe is good idea to
create an appl
On 3 Aug 2006, at 09:14, Jon Schøpzinsky wrote:
Hello
Im trying to decide whether or not I want to use IAX2 trunking on
our WRAP based customer computers.
As it only has a 200mhz processor, I want to make shure that the
trunking part does not affect call quality.
Does anybody know if
Use the ISNULL function. ISNULL will return a value of 1 if the string it's
fed contains data. Example:
If you set VARNAME=foo, and then grab the value of ${ISNULL(${VARNAME})}, it
will be 0 because VARNAME has data
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
On Wednesday 02 August 2006 19:47, Kevin P. Fleming wrote:
Not quite true; the A104 uses a quad framer, but a different one from the
Digium boards.
Oops; my mistake. :-)
-A.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
On Thursday 03 August 2006 08:03, Rushowr wrote:
Use the ISNULL function. ISNULL will return a value of 1 if the string it's
fed contains data. Example:
If you set VARNAME=foo, and then grab the value of ${ISNULL(${VARNAME})},
it will be 0 because VARNAME has data
Just for clarification...
I have Asterisk setup to call two telephones (via SIP trunk)
and bridge the calls. However, when users are talking to each other, the telephone
audio volume is significantly quieter than if they just called each other
directly (without Asterisk).
Is there a way to configure a volume
I thought I was the only one!!! I actually replaced a phone acting just
like you stated until I realized it was required the extra push as
well...
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
Turner
Sent: Thursday, August 03, 2006 12:48 AM
Hi everyone:
I specting high volume of RAM memory consumption in my Asterisk server, the
version of Asterisk is 1.2.4, It could be a bug in my Asterisk version?
When i reboot the server, the asterisk day per day increase the use of the
RAM memory.
Any help is appreciated.
Chris.
Hi everyone:
My asterisk application (Version 1.2.4) with no reason sunddenly die. Can i
have the way for debug the reason why the asterisk appl die?
Regards.
_
On the road to retirement? Check out MSN Life Events for advice
kritikus Araklidas wrote:
Hi everyone:
My asterisk application (Version 1.2.4) with no reason sunddenly die.
Can i have the way for debug the reason why the asterisk appl die?
Regards.
Hi,
in /etc/asterisk/logger.conf, uncomment this line:
;full = notice,warning,error,debug,verbose
Thanks, I know your right (I tried the second option). Problem is that the
phone doesn`t RING. The light flashes, the as far as an audio ring goes,
it`s completely silent.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Vincent
(C)
Sent:
I've set up a ring all context on my gateway on extensions.conf:
[EMAIL PROTECTED] ~]# grep *7 ast/extensions.conf
exten = *7,1,Dial(SIP/1201SIP/1202SIP/1203,15,tr)
Asterisk shows that it rings the lines but in reality nothing happens...
Any thoughts on this?
2006-08-02 17:07:20
You will get a call waiting beep. One. However you can change the
config file and have multiple beeps. You can also change the beep
'sound'. However you must also be aware that while the phone is
playing the beep(s), you are not hearing the far end of the call.
On Aug 3, 2006, at 9:21 AM,
All,
I have a wip 300 phone, it does not support SIP MESSAGE.
I just found there is opensource code for it.
has anyone changed it to support SIP MESSAGE and can share?
How do we contact linksys and have them add it?
ftp://ftp.linksys.com/opensourcecode/wip300/
Thanks. That's an ok solution. I just thought I could make the Polycom
ring normally (or even better, with decreased volume) when a new call comes
in.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: August 3, 2006 11:00 AM
To:
I am using MP3 files on my Asterisk 1.2.10 server. I have defined
mode=files in musiconhold.cfg but the volume of the files is very low.
Is there any way to increase the volume of files played using the native
file format in Asterisk?
--
Carlos Chavez Prats
Director de Tecnología
Thanks Bradley.
Here's a full sip console trace on the first pbx box.
xxx.yyy.128.18(phone 3254101): Originating phone, registered on pbx1.
xxx.yyy.142.162: pbx1
xxx.yyy.142.163: pbx2
Do you know what you are looking for?
Doug.
[Aug 3 09:40:28]
-- SIP read from xxx.yyy.128.18:5060:
INVITE
I have been trying to get an Asterisk 1.2.10 server using ooh323 from
asterisk-addons to call an Alcatel 4400 pbx. Apart from some stability
problems caused by ooh323 I have not been able to establish a call.
Does anyone have experience with this kind of setup? Is ooh323 stable
enough
Oh... That's real nice. I was considering using SIP instead of IAX to trunk
calls between Asterisk boxes as IAX has some severe limitations in regards to
passing variables. A few people said 'use SIP!' because you can set the SIP
headers. Looks like that isn't an option!
-Original
Hi Everyone,
Has anyone had an issue with the Mediatrix and Asterisk where you need to reboot the mediatrix every morning? When I try to place calls or send calls in I am receiving a busy signal. I look at the sip show channels and all the channels seem to be used by the mediatrix. When I
Ok... it'd be great if someone could explain this to me...
User A on pbx1 wants to dial User B on pbx2. We do a local lookup and don't
find user B on pbx1, so we do a DUNDi lookup of user B, get a result, and place
the call to user B on pbx2 with IAX2.
When pbx2 calls the AGI script that
Im aware of ARRAY and would have used it but it only sets variables - you
cant reference the array by name or its elements as is common in other
programming languages.
Would it be possible to use ASTDB and the while application? For
instance the user inputs 3 numbers to be used for Follow
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users
Mailing List - Non-Commercial Discussion
[mailto:[EMAIL PROTECTED]
Sent: Thu, 03 Aug 2006 12:50:54
-0300
Subject: RE: [asterisk-users] SIP_HEADER() read-only
Oh... That's real
- Original Message -
From: Carlos Chavez
[mailto:[EMAIL PROTECTED]
To: Asterisk
[mailto:[EMAIL PROTECTED]
Sent: Thu, 03 Aug 2006 12:39:48
-0300
Subject: [asterisk-users] MoH native volume
I am using MP3 files on my Asterisk 1.2.10 server. I have defined
mode=files in
- Original Message -
From: Jerry Geis
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Thu, 03 Aug 2006 12:21:13 -0300
Subject: [asterisk-users] wip 300 opensource
code - changes to support SIP MESSAGE
All,
I have a wip 300 phone, it does not support SIP MESSAGE.
- Original Message -
From: J. Oquendo
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thu,
03 Aug 2006 11:20:19 -0300
Subject: [asterisk-users] Ringing all extensions
I've set up a ring all context on my gateway on extensions.conf:
[EMAIL PROTECTED] ~]# grep *7
- Original Message -
From: kritikus Araklidas
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Thu, 03 Aug 2006 10:45:14 -0300
Subject: [asterisk-users] RAM memory high
Comsumption
Hi everyone:
I specting high volume of RAM memory consumption in my Asterisk server,
Hello,
I have some of my users that when they leave their voicemail it cuts
off mid message.
the majority works fine. i am new to this so be gentle.
my voicemail.conf is below.(Asterisk 1.2.7.1)
i have the maxsilence set to 200, so i assume that is not the problem
not sure if i understand
- Original Message -
From: Frank Tarczynski
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thu,
03 Aug 2006 07:16:15 -0300
Subject: [asterisk-users] Garbled initial
voicemail prompt
I'm having a problem where the very first words of the Asterisk voicemail
system
I'm a little confused about something. If IAX2 doesn't support passing
variables like the accountcode for example, between Asterisk servers, why do
all the examples have it defined in iax.conf? If the protocol doesn't support
it, then why have it set for an IAX user agent?
Doug.
On 8/3/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
Your Consultant has developed it with PHP scripts, so you must check those
files in
/var/lib/asterisk/agi-bin
Your application logic is there.
Hope it helps,
Best regards,
Marco Mouta
Marco (thanks for the response)
so if understand
I don't understand what the problem is. If you want to pass a variable
set the variable, but prefix it with __ (2 underscores)
Set(__DNID=${DNID})
Douglas Garstang wrote:
Oh... That's real nice. I was considering using SIP instead of IAX to trunk
calls between Asterisk boxes as IAX has some
Hi,
Do you mean I need to preset the queue member to the table? How can I
configure if I want to do the follow sequence using realtime queue?
1. Agent needs to login the queue when necessary (using dial plan)
2. Agent needs to logout the queue when necessary (using dial plan)
3. Calling party
Hi Angel,
We have two DL360s with a TE410P in each one - we had to disable USB to
get the PCI slot to have an IRQ to itself.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Aug 2, 2006, at 6:38
Yup - burned us a few times, too - on IP501s as well.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Aug 3, 2006, at 6:42 AM, Bill Gibbs wrote:
I thought I was the only one!!! I actually
-Original Message-
From: Douglas Garstang
Sent: Thursday, August 03, 2006 10:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX Variables
I'm a little confused about something. If IAX2 doesn't
support passing variables like the
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Thu, 03 Aug 2006 13:29:58 -0300
Subject: [asterisk-users] IAX Variables
I'm a little confused about something. If IAX2
The problem is that I am trying to pass call state variables BETWEEN Asterisk
servers. IAX doesn't seem to support this. One option was to add SIP headers on
the first system, trunk the call with SIP instead of IAX, and then get them at
the other end.
-Original Message-
From: Eric
List,
Sometimes, when doing an ENUM on US toll-free numbers, we get the following
response when dialing to tf.voipmich.com:
Forbidden - wrong password on authentication for INVITE to 'Torrenga
Engineering sip:[EMAIL PROTECTED]
Can anyone explain what this means, or how to avoid it? It does not
I've enabled call
waiting on all my lines. I've setup a queue, and added agents, either as static,
or dynamically using (queue#)*.
When I call the
queue with 2 extensions, only one rings thru until the other answers. In fact,
if I look at the CLI, it doesn't even process the 2nd caller
I'm basically trying to figure out why you aren't authenticating properly.
When you posted the [peer] sections of your sip.conf, were the secret= lines
literal, or did you replace the actual secret with password?
- Brad
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
Brad,
Here's the invite that the first Asterisk box sends to the second Asterisk box.
I see no reference to dundisip1 or dundisip2 in there... I'm not even sure that
SIP can support using different From: and authid's?
[Aug 3 12:45:49] Reliably Transmitting (no NAT) to xxx.yyy.142.163:5060:
Basically, I want to be able to detect that there is voicemail waiting at the
CO on an FXO port and somehow flash the message waiting light on an H.323
phone (or any other type of phone)
I assume that by the lack of response, there is no way to pick up on the
vocemail waiting signalling from
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bob Bosiljevac
Sent: Thursday, August 03, 2006 2:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Detecting voicemail from CO on
,
CALLFILENAME=20060803-144028-1154630428.24777) in new stack
-- Executing GotoIf(Zap/1-1, 0?15:99) in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp(Zap/1-1, NO RECORDING NEEDED) in new stack
-- Executing Macro(Zap/1-1, dial|15|tr|212) in new stack
-- Executing GotoIf
Date: Thu, 3 Aug 2006 14:52:53 -0400 (EDT)From: Bob Bosiljevac
[EMAIL PROTECTED]Subject: Re: [asterisk-users] Detecting voicemail from CO on FXO portand passing to H.323 phone. Possible?To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comMessage-ID: [EMAIL
Brad...
Here's the INVITE that the second asterisk box receives from
the first Asterisk box, after the second asterisk box sends a
Proxy auth message to the first. The first sends the dundisip
userid, but for some reason the second asterisk box is
matching it against the From: 3254101
Greath!, Thank's Anthony.
---
Hi Angel,
We have two DL360s with a TE410P in each one - we had to disable USB to
get the PCI slot to have an IRQ to itself.
Regards,
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
What do you want for the TE410P
Give me a call!
or e-mail me
Thanks!MarkVoice
InternationalAnaheim, Ca 92868714-279-0204 Ext 102WEBSITES
www.voiceinternational.com www.digiumboards.com www.nmsboards.com www.quintumhardware.com www.quintumdirect.com www.dialogicdealer.com
Title: RE: [asterisk-users] Re: DUNDi with SIP
I forget if this does what you want, but try adding a fromuser setting to yoyr peer entries.
- Brad
-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]]
Sent: Thu Aug 03 15:58:50 2006
To: Asterisk Users Mailing List -
Hi,
I`m trying to record a conference, and I`ve been using .wav format to get
decent audio quality. The conference goes fine, but when I listen to the
recording after, I hear horrible echo (which I couldnt hear on the conf
call itself).
Whats causing this?
Mike
hi all i have a newbrand phone Linksys spa941 and i realize that my
asterisk have ECHO. :(
in the zaptel file i have this parameters
echocancel=128
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
i need to add any other parameter to cancel the echo?
thanks. any tips?
Title: RE: [asterisk-users] Re: DUNDi with SIP
I
don't think that will do it. SInce I started with Asterisk a year ago, I've
wrestled the ENTIRE time with the sip conf file. It really makes no sense to me
after doing this 8-5 for 10 months now.
The
first system is sending the correct user
Title: RE: [asterisk-users] Re: DUNDi with SIP
*scream* I just got it to work with
this...
180netsip =
global_dundi_local,0,SIP,dundisip1/${NUMBER},nopartial180netsip =
global_dundi_local,0,SIP,dundisip2/${NUMBER},nopartial
Greetings, I've created a perl CGI that will generate an asterisk call file in /var/spool/asterisk/outgoing/I'm stuck however getting the application 'Flite' working in the call file. By way of example, the CGI generates
the
I'm trying to detect when a T1 goes to Yellow or Red alarm. I noticed
these events will be displayed on the CLI.
What I'd like to do is cause an email to be sent when from a script on
these events, but somehow I would need to
capture the CLI outputs to detect messages
Message are:
wct4xxp:
Have you looked to see if they're being logged to /var/log/asterisk/full ? That would be much easier to detect. --joeyOn 8/3/06, Bart Fisher
[EMAIL PROTECTED] wrote:I'm trying to detect when a T1 goes to Yellow or Red alarm.I noticed
these events will be displayed on the CLI.What I'd like to do
Paul I don't know why you are overcomplicating things. Just like
asterisk can do the MWI for FXS it should be able to pick up the MWI
form the Telco on an FXO, and then based on that asterisk should be
able to turn on the MWI on any ohter device within asterisk. It would
work something like this:
I'm assuming he doesn't mean stuttered dialtone, In the US beside for
stuttered dialtone, there is also a short ring followed by some ADSI
signaling that a lot of the $10 phones at wall mart know how to read
and turn on a lamp on the phone to indicate new VM. The same happens
when turning off the
Mike you can do all that, just paly around with the xml files, you
will want to change the tone it uses.
On 8/3/06, Mike [EMAIL PROTECTED] wrote:
Thanks. That's an ok solution. I just thought I could make the Polycom
ring normally (or even better, with decreased volume) when a new call comes
Yes, even with the r because the second it goes off hook it is
answered. This is an FXO port.
On 8/2/06, Jorge Mendoza [EMAIL PROTECTED] wrote:
Even if he has r in the dial plan?
Jorge
C F wrote:
Then you have something wrong some other place, if you are using an
FXO card then asterisk is
Oh, good idea - the messages do appear there - I'll check it out
Thanks
Joey McDonald wrote:
Have you looked to see if they're being logged to
/var/log/asterisk/full ? That would be much easier to detect.
--joey
On 8/3/06, *Bart Fisher* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
I gave that a try but had no luck. I keep getting all circuits busy.
Perhaps there is another way.
I think it is having trouble when transfering zap to zap.
but no matter what i do i can't get it.
I made a sip number to try from, but its not working
[ext-local-custom]
;exten =
I actually have it semi-working. My trunks were set up improperly.
Now i can do it, but only if i specify a specific zap channel.
exten = 299,1,Macro(dialout,2,1914304,,)
the 2 takes me to zap 1. I tried to replace that with "s" but no luck..
any idea how to do this where it will pick
Date: Thu, 3 Aug 2006 16:46:24 -0400From: C F
[EMAIL PROTECTED]Subject: Re: Re: [asterisk-users] Detecting voicemail from CO on FXOportTo: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comMessage-ID:[EMAIL PROTECTED]Content-Type: text/plain;
CF,
Adding www after Dial doesnt solve the
trouble.
I think we are talking the same but I dont
express correctly.
Did you saw my dialplan? I dont think I would
have to add r.
Yes, I have installed a 4 FXO Card, with fxsks
signalling. What I mean is I understand FXO doesnt give
I have a Polycom phone that was setup without
provisioning through an FTP server. It has a number of contacts that where
input via the phone. I would like to add this phone to a small network that was
provisioned through an FTP server and keep the contacts already on the phone.
How do I
Joshua Colp wrote:
- Original Message -
From: Frank Tarczynski
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thu,
03 Aug 2006 07:16:15 -0300
Subject: [asterisk-users] Garbled initial
voicemail prompt
I'm having a problem where the very first words of the
Can you please show your dialplan? with and withou the www?
Please only the dialplan that doesnt work. Also include
/etc/asterisk/zapata.conf
On 8/3/06, Pablo Mora [EMAIL PROTECTED] wrote:
CF,
Adding www after Dial doesn't solve the trouble.
I think we are talking the same but I don't
I have this in my dialplan:
exten =
s,1,set(LPROMPTDIR=$[${LANGUAGE}?${LANGUAGE}/${PROMPTDIR}::${PROMPTDIR}])
exten =
s,n,set(LPROMPTDIR=$[${LANGUAGE}?${LANGUAGE}/${PROMPTDIR}::${PROMPTDIR}])
The first gets eval-ed like this:
-- Executing Set(Zap/3-1, LPROMPTDIR=tw/china-ivr) in new stack
You're right- that is a simplified way of putting it (a subdialect, as it
were). Point is, to my knowledge (I'll happily stand down if it is) that
support isn't presently there, and there hasn't been a huge rush of people
The support is only a few (realy I don't know how many lines of code,
Hi,
What do I need to add to the dialplan BEFORE a caller enter a queue so
that the recorded call (generated by queue monitor) is encoded to mp3?
I'm defining the monitor save destination with:
exten =
1,n,Set(MONITOR_FILENAME=/new/monitor/queues/${TIMESTAMP}-${CALLERID(nam
I've posted this to the users forum, but
Iguess the list has more traffic, so I'm reposting it
here.
-
Hi, some 13 years ago I use to develop
IVR systems based on Dialogic cards. Now after many years away from CTI, I'm
taking my first steps with Asterisk. As I expected many
Ok
Here goes dialplan
[general]
static=yes
writeprotect=yes
[incoming]
exten = s,1,Answer
exten = s,2,Background(pbx)
exten = s,3,Set(TIMEOUT(response)=5)
exten = 1001,1,Dial,SIP/1001|20
exten = 1001,2,Hangup
exten = 1001,102,Congestion,3
exten =
Further UK prompts have been added to the www.tel.net site.
There's now a complete list of UK (England, Scotland, Wales) Counties,
Towns and London Boroughs, as well as the standard 1.2 base and
additional Asterisk sounds.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 /
On 8/3/06, Stephen Murphy [EMAIL PROTECTED] wrote:
I have a Polycom phone that was setup without
provisioning through an FTP server. It has a number of contacts that where
input via the phone. I would like to add this phone to a small network that was
provisioned through an FTP server
chmod
u-w mac-directory.xml
The
phone will quietly not be able to write to the contacts
directory.
Doug.
-Original Message-From: Stephen Murphy
[mailto:[EMAIL PROTECTED]Sent: Thursday, August 03, 2006
4:09 PMTo: asterisk-users@lists.digium.comSubject:
[asterisk-users]
Change:
callprogress=yes
To:
callprogress=no
Also when dialing over the Zap FXO ports make sure to add the ww
before the DTMF digits so that your extension.conf reads like this:
exten = 9,1,Dial(Zap/g1/ww9 )
On 8/3/06, Pablo Mora [EMAIL PROTECTED] wrote:
Ok
Here goes dialplan
Douglas Garstang wrote:
The phone will quietly not be able to write to the contacts directory.
However, it seems the directory on the phone is maintained. I still
can't work out how to get the Polycoms to replace any locally added
directory items with a master list from the provisioning
With regard to Jims question: the
phone will upload the contacts to a file named MACADDRESS-directory.xml when
the phone boots. Copy that file to -directory.xml and all the
new phones booting up will start with that config.
With regards to Stephens question:
Joey McDonald wrote:
Have you looked to see if they're being logged to
/var/log/asterisk/full ? That would be much easier to detect.
--joey
Hi Joey. What is /var/log/asterisk/full ? I've never heard of it, and don't
see it in my /var/log/asterisk/ .
Cheers,
-- Nick
e: [EMAIL
exten = _9.,1,Dial(Zap/1/w${EXTEN:1},20)
Should the w be before the number dialied?
Should it the w come AFTER the number dialed?
[EMAIL PROTECTED]
blah...
From: Dante Passalacqua [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
1 - 100 of 114 matches
Mail list logo
