Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
> 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.
>
something like this?
exten => _713X.,1,System(/web/ht
> -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 o
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" wrote:
>
Hey all.
My boss asked me to implement the following
When DID 713xxx is dialed send an
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 Asterisk
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 forgot.
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 interne
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 corr
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 mailto:seandar...@gmail.com>>
>
> I can't get a test T.38 fax betwe
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 se
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
[mailto:asterisk-users-boun...@lists.d
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
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: ;tag=74ff1200077fff10ff0
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
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
[mailto:asteri
> - 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-00
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,
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
Hi!
> exten =>
> 020300,3,Dial(SIP/44208...@sipprovider&SIP/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 answer
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 yo
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 care
>> I am diverting an incoming call to a mobile phone and a landline using the
>> following:-
>>
>> exten =>
>> 020300,3,Dial(SIP/44208...@sipprovider&SIP/4470...@sipprovider,120,r)
>>
>> For billing purposes, i need to be able to work out whether the diverted
>> call was answered by the
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
wrote:
> Hi,
>
>
>
> I am diverting an incoming call to a mobile phone and a landline using the
> following:-
>
>
>
> exten =>
> 02
On Mon, May 3, 2010 at 3:04 PM, Danny Nicholas 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 measured valu
Hi,
I am diverting an incoming call to a mobile phone and a landline using the
following:-
exten =>
020300,3,Dial(SIP/44208...@sipprovider&SIP/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 whe
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: asterisk-users-boun...@list
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 (hunting
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
_
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 yo
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: M
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 instalat
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 P
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, be
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 => _XXX.,n,Playback(callbox-th
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
> Hello to all.
>
> This h
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 get
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
toda
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: 'Aste
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