what is wrong with this please:
;exten => 4,1,WaitExten(3)
exten => 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten => 4,n,WaitExten(3)
exten => 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten => 2,n,Waitexten(3)
exten => 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten => 3,n,WaitExten(3)
exten => 9,
On Fri, Jan 29, 2010 at 12:08:19AM +0100, Laurent CARON wrote:
> Hi,
>
> I'm currently trying to get a BN4S0e (which is basically a BN4S0 with a
> PCIe connector) working with dahdi.
>
> The module is loading properly but the card is not detected by the module.
>
> Is support on dahdi planned f
thanks for the response.
I tried to simplify and am now tuning the following, but it is not
responding with anything.
something wrong with timing?
here is what I have:
exten => 1625,1,Answer()
exten => 1625,n,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten => 1625,n,WaitExten(3)
exten => 9625,1,Answer(
Thanks very much everybody who contributed their thoughts. I would try
to get some DID's so that each physical location can be identified by
911 call Center.
Regards
Shahnawaz
On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote:
> Leif Neland wrote:
>
>> 2: Often callers are answered with an
On Fri, Jan 29, 2010 at 3:09 PM, Hristo Benev wrote:
> If I use call file with spool
> Fax is send but if I use manager
> I get
> Any suggestions?
Well, one obvious solution is to just use call file. Problem solved.
Try changing your call manager setup to use a Local channel instead,
and set up
I have several models running; IP335, IP430, IP450 & IP60. I'm running
v3.2.1.
I don't have convenient access to the configs at the moment. But if you
download the newer firmware from Polycom their stock configs will give
you all that you need.
http://downloads.polycom.com/voice/voip/sip_sw_relea
thanks..
2010/1/30 Kevin P. Fleming
> Mariano Lecuona wrote:
>
> > All I just want to be able to detect the fax signail while doing an
> > outbout call taking advance of the out_dialout feature of asterisk. So
> > for to have a clear image on how i am doing it.
>
> The faxdetect functionality in
Mariano Lecuona wrote:
> All I just want to be able to detect the fax signail while doing an
> outbout call taking advance of the out_dialout feature of asterisk. So
> for to have a clear image on how i am doing it.
The faxdetect functionality in Asterisk is not intended to detect
answering FAX m
Hi,
All I just want to be able to detect the fax signail while doing an outbout
call taking advance of the out_dialout feature of asterisk. So for to have a
clear image on how i am doing it.
I have my .call that I move the /var/spool/asterisk/ouotgoing like:
(numbers were changed to preserve priv
There's no issue here. There are parameters in the Polycom config files
for the various gain settings. The headset, handset and speakerphone
volume and mic gains are all separate. They can be tweaked as you like
via the config files.
Michael
--Original Message Text---
From: hin lee
Date: Sat, 30
Yes, the external calls are going over DAHDI. The problem is on the Polycom
phones b/c if I pick up the handset, the other end can hear me fine. The
problem is when using the hands-free (speakerphone) instead the handset.
Here are some similar posting of the same issue.
http://www.trixbox.o
I am also having this issue with the MOH. Would be nice to find a solution!
From: Steve Edwards
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Fri, January 29, 2010 3:43:12 PM
Subject: Re: [asterisk-users] Help for MOH - sounding scratchy/
Try something like:
exten => 1,1,WaitExten(3)
exten => 1,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten => 1,n,WaitExten(3)
exten => 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten => 2,n,WaitExten(3)
exten => 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten => 3,n,WaitExten(3)
exten => 9,1,SayNumber($
total up for current call.
then read back the number
2010/1/30 Håkon Nessjøen :
> For all calls combined, or for the current call?
>
> On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron
> wrote:
>>
>> I want to create a script for IVR that compiles responses, aggregates
>> them to a total number.
>
Olle E. Johansson wrote:
> Here's something interesting:
>
> 21.4.25 487 Request Terminated
> The request was terminated by a BYE or CANCEL request. This response is never
> returned for a CANCEL
> request itself.
>
> This is only used in combination with Cancel's, but in this case the call was
For all calls combined, or for the current call?
On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron wrote:
> I want to create a script for IVR that compiles responses, aggregates
> them to a total number.
> Then, run an equation based on the result.
>
> Press 1 for X (X is a positive number 500)
> Pr
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the resul
Thereis an option with log queue stored into database. You can do an
cross join from cdr and queue log using callid. With this solution you
can track your call over your asterisk system.
I wrote about this in old post and submit an complete solution.
Regards,
On Sun, Jan 24, 2010 at 1:14 PM, Wil
Friends,
Before the Christmas holidays, I did send this letter and did not get a lot of
response, but some. Since then, I've been able to get interest from a few
parties that are willing to fund parts of this work, including Digium, the main
sponsor of Asterisk. I will also apply for additional
Gentlemen,
I did borrow an Aastra RFP 32 for some tests that i wanted to do.
Everything seems to be working except CLID. Setup as below:
DECT handset - GAP - Aastra RFP-32 - SIP - Asterisk - SIP Phone
When SIP Phone calls DECT handset, the display on the DECT handset only
shows the number of SIP
How can I seperate outgoing calls to the same ITSP but with a different
DID ??
I have defined a peer in sip.conf for the incoming calls that are routed
to a certain [context]. Then in the dialplan this [context] send the
incoming calls to the correct context based on the callerID that was
send wit
Hi
If I have an incoming call coming down a SIP trunk to a particular
internal SIP extension- I can answer the extension fine, all works
well
However, if I change extension.conf from dialling the internal
extension to forward the call to an external cell phone (up the same
trunk as the incoming l
29 jan 2010 kl. 17.20 skrev Kristian Kielhofner:
> On Fri, Jan 29, 2010 at 10:31 AM, Kevin P. Fleming
> wrote:
>>
>> Well, that's the problem, and it's the reason why 603 is so commonly
>> used. This is a situation where the current request has failed, but
>> there is no indication that repeat
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