Suggestion for you:
exten = _X.,n,MP3Player(/dir/to/your/mp3/directory/)
You could try changing
/usr/bin/wget -q -O - $1 | /usr/bin/madplay -Q -z -o raw:- --mono -R
to something like
/bin/cat $1/`ls $1 | shuf | head -1` | /usr/bin/madplay -Q -z -o raw:- --mono -R
Sorry, didn't notice the version
Hi Sean,
Do you know about t38modem and hylafax?
There are lots of wonderfull options with both of them.
If you need config files with both of them tell me.
See ya
2010/5/2 sean darcy seandar...@gmail.com
I can't get a test T.38 fax between 2 1.6.2 machines, using app
_fax and spandsp pre17
You can call an external script and call CURL from there (either use
AGI, or Asterisk cmd System).
It depends on your task what to use (perl/bash/C...)
On Mon, May 3, 2010 at 7:47 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Dear All,
Last Week i tried and goggling more on how to call
Hello to all.
This has been my first post to the list and I'm a bit flustered by the
situation I describe so sorry if I got anything wrong.
Any help or pointer anyone can give me will be greatly appreciated.
Regards
Enrique
De: Enrique Mora
Enviado el: lunes, 03 de mayo de 2010 9:05
Para:
Suddenly, after restarting our server we are unable to load chan_dahdi
The configuration has been stable for months but for some reason we get these
errors when trying to load chan_dahdi. The Unregister application
DAHDISendKepadFacility application does not appear in any logfiles prior to
On Mon, 3 May 2010, DHAVAL INDRODIYA wrote:
Last Week i tried and goggling more on how to call RESTful webservice
from Dialplan?
i found CURL function but while i tried to use it ,it 's not supported
HTTPS request and we cannot set headers while send a request.
also without HTTPS . i
Hi,
before starting asterisk check your dahdi driver. The outpud of
/etc/init.d/dahdi start, the dahdi module with lsmod, and look at
/proc/dahdi for the pri status.
another resource is: http://www.voip-info.org/wiki/view/Asterisk+PRI
regards
2010/5/3 Enrique Mora em...@context.es
Hello to
Hi everybody,
I have a problem using parking for outgoing call.
A is an local sip phone. A is using the local extension :
[local]
exten = _XXX.,1,Wait(0)
exten = _XXX.,n,Dial(SIP/${EXTEN:0...@trunk_sip_2,0,TK)
exten = _XXX.,n,Dial(DAHDI/4/${EXTEN:0},0,TK)
exten =
Unfortunately no, it did not solve my problem, the sitation is the same.
Any other hint?
Best regards,
Peter Gelencser
2010.05.01. 9:38 keltezéssel, Rudi Oosthuizen írta:
Had a similar problem with a B410p BRI card. Had to enable (or disable)
the 100ohms termination jumper on the card,
Is there any way, i can detect in asterisk that which party hanged up the
call either from A side or B.
Both parties are using SIP protocol. I am using Asterisk 1.4.27
Shariq Khan
0333-3501125
--
_
-- Bandwidth and Colocation
Hello,
I need to limit the RTP ports used by an asterisk in a client,
Actualy the range defined is from 1 to 2 udp ports.
If I only have 10 local sip extension ¿how many ports/range should I
set up in /etc/asterisk/rtp.conf?
Which is the way to calculate the rtp ports needed on an
In my installation, netstat usually indicates 4 ports per extension, so my
assumption is that you would need 40 ports or a range of 1-10039.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy
Sent:
Hello,
I saw that Asterisk don't calcultate fine the ANSWEREDTIME.
I want that when ANSWEREDTIME =~ 5.6 become 6 and if =~10.3 become 10
because, now, if ANSEREDTIME =~ 15.9, it become 15! it isn't correct
How can I have a rounded ANSWEREDTIME ?
Where have I to manipulate the sources?
thank
someone has Asterisk 1.2 (upgrade is not possible), and wants to spy on
specific extensions he can specify while dialing a code, could you please
kindly tell us how to do this.
Thanks
Hi all... I'm sorry for repeating my message.
I have a problem with caller id on my asterisk server with xorcom astribank.
here is my configuration :
centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2
ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco
The values for ANSWEREDTIME are set in apps/app_followme.c and
apps/app_dial.c . The values are set in seconds, so if you're looking to
set nearest minute you'll just need to change the sprintf from %1d (1
decimal point x.x) to %0d (x).
-Original Message-
From:
Hi,
I am diverting an incoming call to a mobile phone and a landline using the
following:-
exten =
020300,3,Dial(SIP/44208...@sipproviderSIP/4470...@sipprovider,120,r)
For billing purposes, i need to be able to work out whether the diverted call
was answered by the mobile or
On Mon, May 3, 2010 at 3:04 PM, Danny Nicholas da...@debsinc.com wrote:
In my installation, netstat usually indicates 4 ports per extension, so my
assumption is that you would need 40 ports or a range of 1-10039.
Sounds reasonable, I was going to suggest 100 would easily do, but an
actual
I am 99% sure you will be able to catch this information in AMI. I
didn't try with call diverts, but it says really alot.
On Mon, May 3, 2010 at 4:41 PM, Dan Journo d...@keshercommunications.com
wrote:
Hi,
I am diverting an incoming call to a mobile phone and a landline using the
I am diverting an incoming call to a mobile phone and a landline using the
following:-
exten =
020300,3,Dial(SIP/44208...@sipproviderSIP/4470...@sipprovider,120,r)
For billing purposes, i need to be able to work out whether the diverted
call was answered by the mobile or
Hi!
In my installation, netstat usually indicates 4 ports per extension,
so my assumption is that you would need 40 ports or a range of
1-10039.
Sounds reasonable, I was going to suggest 100 would easily do, but an
actual measured value is even better :)
Be a bit careful:
All:
My company has an existing ESI IVX E-class system with 45 phones. I can
add one more card, to expand it another 6 phones, but it's $8000, and
then the system will have to be replaced.
I have the Asterisk server up and running, with 2 sip lines from the
local phone service. (Thanks to
Hi!
exten =
020300,3,Dial(SIP/44208...@sipproviderSIP/4470...@sipprovider
,120,r)
For billing purposes, i need to be able to work out whether the diverted
call was answered by the mobile or whether it was answered by the
landline.
How can i log which phone answered the
On Mon, May 3, 2010 at 4:25 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
so my assumption is that you would need 40 ports or a range of
1-10039.
Sounds reasonable, I was going to suggest 100 would easily do, but an
actual measured value is even
Assuming that the ESI system phones are SIP protocol, you should be able to
do native sip dialing like 1...@foo or 1...@bar. You would set up Regis in
asterisk with this line in the dialplan
Exten = 120,1,Dial(SIP/1...@esi,20,m)
In other words, you would treat the 45 ESI lines like softphones,
- you could also consider the M() option to Dial together with the CDR
userfield for logging whatever channel variable make sense to you
I'll see if I can sort it out with that.
- have you looked at the destination channel in the CDR?
The destination channel says:-
SIP/sipprovider-002c
What ports to you have available on the ESI ?
Analog Trunk Lines?
Analog Station Lines?
PRI?
You could bridge with maybe a small 4 or 8 port FXO/FXS device depending on
what you have available in on your ESI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi there.
I have the similar problem (Digium fax - sending fax call file vs
manager originate) sending faxes with Asterisk 1.6.2.6 and Digium
res_fax. Receiving is OK.
I use Local channel in Call file and context [fax-out] in dialplan.
My setup: Asterix-SIP (T.38)- Cisco(MERA MSIP v.1.0.2)-
On 05/03/2010 11:59 AM, Ilmars Knipshis wrote:
Problem in short is as following:
after reINVITE from Cisco to negotiate T.38:
--- SIP read from UDP:193.110.9.17:5060 ---
INVITE sip:37166101...@159.148.78.220 SIP/2.0
Via: SIP/2.0/UDP 193.110.9.17:5060
From:
I am trying to run a script before and after the Page application in
order to mute/un-mute my whole house audio when my phones are being
used as an intercom. Unfortunately, I am unable to get the system call
after the Page line to run (i.e. /bin/vol_restore). I have also tried
running it using the
Run BVR as a DeadAGI in the h extension.
In /var/lib/asterisk/agi-bin create this file
Vol_rest.agi
#!/bin/sh
Run /bin/vol_restore
From the dialplan
Exten = h,1,DeadAGI(vol_rest.agi)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Eddie Mikell wrote:
All:
My company has an existing ESI IVX E-class system with 45 phones. I can add
one more card, to expand it another 6 phones, but it's $8000, and then the
system will have to be replaced.
That is worse than highway robbery.
I feel sure with some careful
Miguel Amez wrote:
Hi Sean,
Do you know about t38modem and hylafax?
There are lots of wonderfull options with both of them.
If you need config files with both of them tell me.
See ya
2010/5/2 sean darcy seandar...@gmail.com mailto:seandar...@gmail.com
I can't get a test T.38
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of François
BERGANZ
I saw that Asterisk don't calcultate fine the ANSWEREDTIME. I want that
when ANSWEREDTIME =~ 5.6 become 6 and if =~10.3 become 10 because, now,
if ANSEREDTIME =~ 15.9, it become 15! it isn't
How do you configure Asterisk to dial, in order, each channel from a
group of channels until it either finds an available channel, or runs
out of channels?
We recently got VoIP, so when we make a call, Asterisk should first try
to make the call with VoIP, but in case either our VoIP or our
Hey all.
My boss asked me to implement the following
When DID 713xxx is dialed send an email to mmos...@xxx.com. with the
time date and CID included in the email. I know how to code some but am
looking for the best way to do this.
Sorry I might have asked this a couple months back. I
I was in a similar situation with a Toshiba CIX PBX. I had 150 phones
on the Toshiba and wanted to switch over to SIP phones slowly. The
Toshiba already had PRI cards connecting to the phone company. I
purchased Sangoma PRI cards for the Asterisk server. I connected the
Toshiba PRIs to the
Untested, just throwing something out off the top of my head... modify to
suit...
exten = _713.,x,System(/bin/date | /bin/mail -s ${EXTEN} mmos...@xxx.com)
--Tim
- mike mosier trixbo...@gmail.com wrote:
Hey all.
My boss asked me to implement the following
When DID 713xxx is
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of John Novack
Sent: Tuesday, May 04, 2010 12:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bridging old
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