On 08/27/06 13:23 Rushowr said the following:
Set(TIMEOUT(absolute)=seconds)
Change seconds to the number of seconds you want to allow a call to last
alternatively, look at the L() option to Dial.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED]
On 08/26/06 23:52 Crazy Boy said the following:
Hi friends,
I did music on hold. How can we implement music on call transfer? I am
unable to find any tutorial about setting up music on call transfer,
i'm not exactly sure what you're intending to do, but MoH is already active
and played
Hello,Could you tell how i can use it in PERL AGI script?currently i am using in my AGI with this format, but some time call is not disconnecting customers talking without money.$dialstr = "SIP/terminator/15745405022|350|tTL(653044:7000:5000)";$AGI-exec('Dial', $dialstr);regards,
Get your
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First big question is are you checking beforehand how long the limit should
be by calculating ((BALANCE / RATE) / 1000)
If you're not, that would be why it doesn't disconnect the customer within a
time period that wouldn't result in a negative
Mario wrote:
We have used both IP501, IP601 (Polycom), Snom 320 and Snom 360. All of
them are good phones with very good quality of voice and full of features.
However, SNOM phones have a feature (missing from Polycom) that most of
our customers really require: with SNOM phones you have leds
Hi,i am using the same calculating ((BALANCE / RATE) / 1000) method to return tTL.and i am sure my GAI is working well. but could u tell me how i can set Verbose() sepecial for my dialstring?Regards,
Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.
from within asterisk, just run the following
command:
show application Verbose
That'll fill you in. Your other solid option is to search
the wiki
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
AbdulSent: Sunday, August 27, 2006 4:05 AMTo:
Dear all
I am using AMP 1.0.10 . my voicemail system working perfectly, but now
i like to sent user PIN ( Password of the extension) number with that
mail .
how could i read the user passwd value (PIN) so that i could append
the mail format
thanks
Salaque
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said:
I want that each call from PSTN goes to Asterisk to the context for
this line. Within this context can be a menu or a dial command, ...
As more I read, as more I get confused, ... and each try is not working!
I don't know
any one try that with g723 codec?
thanks
Salaque
On 8/27/06, Rosli Sukri [EMAIL PROTECTED] wrote:
i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem
to get it to work with ms netmeeting
On 8/26/06, atik khan [EMAIL PROTECTED] wrote:
Hi,
i used to work ooh323 with
Hi,
I have a problem about musiconhold/playback command in the dial
plan. Actually, I have the following dial plan for inbound call.
[from-gateway]
exten = 1234,1,Answer()
exten = 1234,2,MusicOnHold(c1)
exten = 1234,3,Hangup()
When I using mobile to make call 1234, it works and I can hear
Hi all,
I'm running Asterisk SVN-trunk-r40489 on which one I have a Sipura 1001
connected. I face a problem when sending digits to voicemail password:
each one is sended twice (eg 35 give 3355) I have the same behaviour if
I have to enter the mailbox number before.
I have no problem to
On 27 Aug 2006, at 04:03, Dovid Bender wrote:
I was not going to get it based on what people said about the E61
and the NAT issues. Is this false ? I was thinking of getting it
for when I travel to Israel. There seems to be a lot of open wifi
connections all over the country there. Also
Hello
I would like to NOT record a CDR for internal calls, but the C option
(suppose to work like NoCDR() ) is just not working for me. My dial line is
exten = _70XX,1,Dial(SIP/${EXTEN}|20|Ctr)
Could someone give me a short example of using NoCDR correctly.
Thanks
Master
Hi. If I register asterisk with another server as an extension to
that server -- say -- using iax2 how can I dial an extension on that
second server?
I tried the following exten = 8200,1,Dial(iax2/201/8200,,r) but got
no route to destination even though the other server saw my
registration.
On Saturday August 26 2006 11:15 am, Matt Riddell (IT) wrote:
John Millican wrote:
Hello all,
I am trying to test if the length of a dialed number is greater than 7.
When i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
You can get that on the polycom if you want to fork over another $200.00 +
for the side car. Or if you are using a 601 you can use the first line for
all your calls and then the next 5 for it.
- Original Message -
From: Mario [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On 27 Aug 2006, at 04:03, Dovid Bender wrote:
I was not going to get it based on what people said about the E61
and the NAT issues. Is this false ? I was thinking of getting it
for when I travel to Israel. There seems to be a lot of open wifi
connections all over the country there. Also
HiI change the kernel to 2.6.14.7, but unfortunately the problem still exist.The
messages empty HDLC frame or bad CRC received appear only when there
is not traffic on card (0 active calls). It never happens during a
call. Strange?!?
Any other tips are gladly expected :).CheersAndrew
- RR [EMAIL PROTECTED] wrote:
Any ideas if it's possible to either record greetings/names in a
different format than GSM OR be able to convert these voicemail
subscriber greetings in my database to some other format?
They will be recorded in the same formats that you record voicemail
NAT is a problem at the moment, I can only connect to my Asterisk server on the same network. Wifi work nicely and you can get up groups of access points so that when you move it roams to the next active point you're on.
I hear Nokia are aware of the NAT issue and are going to update.RegardsRobOn
When I do echocancel=yes it stops working. I have to have it at no in order
for it to work.
- Original Message -
From: Dovid Bender [EMAIL PROTECTED]
To: Dovid Bender [EMAIL PROTECTED]
Sent: Friday, August 25, 2006 12:23 PM
Subject: Fw: [asterisk-users] [RESOLVED] One way audion on
Hi,
I'm having problems with calling the ${CDR(billsec)} ${CDR(duration)}
variables in an AGI.
Note that I'm using Asterisk-1.2.10 and Realtime extensions + Realtime sip
users/peers.
John
mail2web - Check your email from the
Can you give us some more info? Like agi debug output?
On 8/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I'm having problems with calling the ${CDR(billsec)} ${CDR(duration)}
variables in an AGI.
Note that I'm using Asterisk-1.2.10 and Realtime extensions + Realtime sip
Hi,
I'm running Asterisk 1.2.10 bristuffed.
Asterisk is registring perfectly against my provider (musimi.dk), and
incoming calls comes in and are routed fine to either internal ZAP
(ISDN BRI) and/or SIP.
But
I can't dial out via SIP (musimi)
sip.conf:
[musimi]
type=friend
host=musimi.dk
AGI
===
$res = $AGI-exec(Hangup);
$foo = ${CDR(billsec)};
myVerbose($foo); #print on CLI
$foo = ${CDR(duration)};
myVerbose($foo);
$foo = ${CDR(answer)};
myVerbose($foo);
$foo = ${CDR(start)};
myVerbose($foo);
Hi All,I was hoping someone could help me with a problem I'm having determining a users number. Is there any way in the dialplan or with an AGI to detect what a users number is for use in a meetme conference?
I am using the MeetMeAdmin function from within the dialplan.I
would like one of my
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[EMAIL PROTECTED] wrote:
AGI
===
$res = $AGI-exec(Hangup);
$foo = ${CDR(billsec)};
myVerbose($foo); #print on CLI
$foo = ${CDR(duration)};
myVerbose($foo);
$foo = ${CDR(answer)};
keeping track of the confno is easy since I created it,
but I don't know how to determine the user number of the last person that
joined the conference.
Is there a way to store this in a variable before they join the conference?
Or perhaps a way to detect the last user to join the conferences
First I nor sure if you can use the 7970 with SIP. LoadID looks for me the
bootimage does not match with the applicationimage. Mabe you have to erase the
flash (I'm not sure)
here is my config for Skinny Channel
venus:/srv/tftpboot # cat XMLDefault.cnf.xml
Default
callManagerGroup
members
Hey List!What are your thoughts on redundancy?? Is it best to have the two Asterisk boxes share a /etc/asterisk directroy; so if one falls out of service the other takes over or is it best to have each node pull from a shared DB??
Cheers!-- --Christopher T Aloi--
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Personally I've used the shared database method previously, I've even setup
a mysql cluster and had each asterisk host be a query node.
SKM
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
On 17:31, Sun 27 Aug 06, Christopher Aloi wrote:
Hey List!
What are your thoughts on redundancy??
Is it best to have the two Asterisk boxes share a /etc/asterisk directroy;
so if one falls out of service the other takes over or is it best to have
each node pull from a shared DB??
What we
Is it a bug or is it me?
For the longest time I have been using the feature within voicemail to
call back a number by caller ID.
Never had a problem with it at all.
I just updated to the latest (stable) asterisk from asterisk.org
Option 3 (advanced) then 2 then 1
caller number 7347292615
and
Is there a maximum number of SIP devices that can be registered to an extension?-brandon-- Brandon GalbraithEmail: [EMAIL PROTECTED]
AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost
___
Stop trying to con lonely nerds in to answering you questions with
subjects like that Steve!
Anyway, check the bug tracker, I think someone posted on this list
about a week ago with the exact same problem.
On 8/27/06, Steve Gladden [EMAIL PROTECTED] wrote:
Is it a bug or is it me?
For the
On Sat, Aug 26, 2006 at 02:02:51PM -0700, Martin Joseph wrote:
On 2006-08-22 01:59:09 -0700, Tomislav Parčina [EMAIL PROTECTED] said:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5
over Grandstream HT488 ATA.
snip
Personally I found the FXO port on the
Shouldn't the line:
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
be:
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
note the .dk in the second one...
Also I don't see a register line in your sip.conf. In the [general]
section I would have expected something like:
register=a
Woops, sorry the first part of my response is wrong:
Shouldn't the line:
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
be:
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
note the .dk in the second one...
What I said here is incorrect, looks to me you have it right.
You may still
Hello,
I apologise if this has been covered on this list in the past but I have
been searching
for a couple of weeks for a solution and have not yet come across one.
I have set up a basic click to call service on my trixbox. The service
operates by creating
a call file and then once in the
You can parse the Variable BEFORE sending to the conf.
Ie:
Exten = _8700X,1,Set(${DB(conf${EXTEN}/lastin)=${CHANNEL})
It will always be the last one in.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Sunday,
- Original Message -
From: Henrik Woffinden [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, August 27, 2006 11:50 AM
Subject: [asterisk-users] Cannot dial out through SIP provider
Hi,
I'm running Asterisk 1.2.10 bristuffed.
Asterisk is registring perfectly
snip
2) What are the best sources (cost effective) to get prompts recorded.
/snip
I would go with allison. She is the one that did all the voice files that
you currently have on asterisk. So if you use her for your prompts you will
have the same voice thru out ur PBX. A client of mine just
Hello.
I'm trying to evaluate my path to several voip providers, so I set
qualify=400 in iax.conf. But, I'm not seeing any
REACHABLE/UNREACHABLE or LAG messages in the logs. Is there a logging
option to set so these will show up? Also, how often does asterisk do
a qualify check.
Thanks,
Hi Matt,
Thanks for the information.
Correct syntax for calling the CDR function in the AGI would be a great
help.
I have tried $foo = $AGI-exec(${CDR(xxx)});
and
$foo = $AGI-${CDR(xxx)};
none of the above works.
then i tried this:
$AGI-verbose(CDR(billsec));
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