Displaying 20 results from an estimated 1000 matches similar to: "Generate two calls from Asterisk and bridge them"
2006 May 23
3
Transfer extensions processing control to Manager
I'm developing an application that monitors the state of the incoming
calls using Manager events. So, as a part of it, I need to "override"
the control of the extensions by the dialplan itself. The problem is
that, if I don't declare the incoming extension, Asterisk hangs up the
call by default. So I want to know if there's some kind of
"ManagerControl() application
2006 May 11
2
Problem setting locale for voicemail
I've set voicemail almost successfully, only a minor detail remains :-)
I can't get the dates in my local language (spanish). In sip.conf,
zapata.conf and voicemail.conf, I've set:
language=es
and my locale is "es" also. However, the days and months names still
appear in english in the emails!!!
Thursday 11 de May de 2006, 18:49:34.
instead of
Martes 11 de mayo de
2006 Mar 15
2
Speeding up the dial of DTMF's in SIP channel
I'm dialing DTMF's in a SIP channel using the options:
[sip.conf]
dmtfmode=info
[extensions.conf]
exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN}))
(this is a custom SIP gateway, which receives the DTMF's sent from
softphones through Asterisk, and based on them, build the destination
PSTN number).
My problem is that Dial send the DTMF's to the SIP/gateway user at a
rate
2006 Mar 09
2
Extracting info from the $EXTEN variable
Is there a way to access only certain positions in the $EXTEN variable?
I'd like to filter my international calls based on the destination country:
My dialplan looks like this (1XX0. is the international calling
convention for Chile)
exten => _1XX0.,1,Dial(SIP/${EXTEN:4}@external_provider)
But, I'd like to, depending on the destination country (digits 5 and
eventually 6 of EXTEN),
2006 Feb 23
3
Codec order sent wrong from Asterisk
I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000.
The codec order on each one is the next:
SJPhone: GSM - iLBC - PCMA - PCMU
GXP2000: G729 - GSM - PCMA - PCMU
(I have a G729 license, so there's no problem with transcoding G729)
In my sip.conf, I've defined the following codec order:
disallow=all
allow=g729
allow=gsm
allow=g726
allow=alaw
allow=ulaw
And my
2006 May 10
2
Is there a way to not propagate a context included inside other context?
I've defined my dialplan as showed below. My internal lines are numbered
as 12345XX, and internal users can call another by the entire 7-digits
extension, or by just last 2 digits.
[invalid]
exten => _X.,1,Playback(pbx-invalid)
exten => _X.,2,Hangup()
[internal]
include => invalid
exten => _XX,1,Dial(SIP/12345${EXTEN}) ; Short alias for internal lines
exten =>
2006 Apr 04
2
Can't get Pickup app working
I'm trying to set the Pickup feature. I'm setting my extensions.conf as:
exten => _*.,1,Pickup(SIP/${EXTEN:1})
but if, for example, extension 03 is ringing by a call made from
extension 01, and I try to pick it up from extension 02 (by dialing *03
from extension 02), I can see in the Asterisk console (Verbosity set to 10):
-- Executing Dial("SIP/01-512c",
2006 May 12
3
VoiceMail application: "j" option not working as I supposed
I've the following dialplan.
exten => _XX,hint,SIP/${EXTEN}
exten => _XX,1,Dial(SIP/${EXTEN},10,j)
exten => _XX,2,VoiceMail(${EXTEN}@default,u|j)
exten => _XX,3,Hangup()
exten => _XX,102,Goto(110)
exten => _XX,103,Playback(pbx-invalid)
exten => _XX,104,Hangup()
exten => _XX,110,VoiceMail(${EXTEN}@default,b|j)
exten => _XX,111,Hangup()
exten =>
2006 Feb 23
1
How can I force Asterisk t not override my codec order?
I've noticed the following situation:
In two softphones, I've configured the next codec order for each one
softphone 1: 1 - PCMA
2 - GSM
softphone 2: 1 - GSM
2 - PCMA
and in Asterisk, the order is:
disallow=all
allow=gsm
allow=alaw
If I call from softphone 1 to softphone 2, I presume that Asterisk
should do transcoding (canreinvite is set to no):
2006 Mar 08
1
Upgrading Asterisk witk G729 license installed
I've an Asterisk 1.2.4 installation, where I've also installed the G729
codec license. I'd like to upgrade that installation to 1.2.5, but I'm
not sure if I'll lost the license in the process (and if I'll be able to
recover it later!!!).
Is there any special consideration I've to keep in mind in this case, or
should I just run the typical "make + make
2006 May 23
1
Problem with options to "Dial" application
I'm trying to set a dialing rule in my dialplan. As a part of it, from
my point of view, this works wrong
priorityjumping=no
[test_context]
exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag
exten => 1234,2,Playback(digits/2)
exten => 1234,3,Playback(digits/3)
exten => 1234,102,Playback(digits/4)
In this case, if I dial the extension, and
2012 Dec 10
2
CISTI'2013 Doctoral Symposium - CFP, Lisbon, June 19 - 23, 2013
***************************************************************************************************
CISTI'2013 DOCTORAL SYMPOSIUM
8th Iberian Conference on Information Systems and Technologies
Lisbn, Portugal, June 19 - 23, 2013
http://www.aisti.eu/cisti2013/index.php?option=com_content&view=article&id=64&Itemid=68&lang=en
2012 Dec 10
2
CISTI'2013 Doctoral Symposium - CFP, Lisbon, June 19 - 23, 2013
***************************************************************************************************
CISTI'2013 DOCTORAL SYMPOSIUM
8th Iberian Conference on Information Systems and Technologies
Lisbn, Portugal, June 19 - 23, 2013
http://www.aisti.eu/cisti2013/index.php?option=com_content&view=article&id=64&Itemid=68&lang=en
2001 Dec 28
1
En: SETEUID
Please,
I can`t see my messages.
Can anyone confirm if it is reaching to the list?
Thnaks!
?lvaro
----- Original Message -----
From: Alvaro Lassance <lassance@sidercom.com.br>
To: <samba@lists.samba.org>
Sent: Thursday, December 27, 2001 1:39 PM
Subject: SETEUID
>
> > Hello!
> >
> > Anyone knows how I install the "seteuid method" in a RH 7.0?
>
2006 May 09
0
How to distinguish between UNEXISTENT channels v/s UNAVAILABLE channels
Is there a way to distinguish, in the answer of the Dial command,
when a channel is not available (for example, an unregistered but
valid SIP user) v/s when the dialed channel is inexistent, even
when it matches an extension?
For example, I've the following simple dial plan:
exten => _XX,1,Dial(SIP/${EXTEN},10,)
exten => _XX,2,GotoIf(DIALSTATUS = CHANUNAVAIL?4:3)
exten =>
2000 Jul 20
4
RFC: System and time support functions in R
I've been looking over system utility functions that we might want to
add to R. A few come out of specific needs, others from looking at
other systems and what people are using system() for. I've taken
account of Paul Gilbert's comments posted here a while ago (and I
think covered all except the use of mailers).
We currently have
date
*.socket
file.create
file.exists
file.remove
2006 Feb 22
0
Is SIP "canreinvite" working ok?
I've the following situation:
Phone A: Codec GSM supported
Phone B: Codec iLBC supported
in sip.conf:
[general]
...
disallow=all
allow=gsm
allow=ilbc
allow=alaw
allow=ulaw
canreinvite=yes
...
(There's a lot of other SIP users, that's why I made the default codec
list bigger than just GSM and/or ALAW)
If phone A calls to phone B the conversation is established at SIP
level, but
2006 Mar 06
0
Information to program a new driver for Asterisk
I'm interested in developing a new channel driver for a thrid party
telephony card for Asterisk. Is there any "official" document that
explains how to do this? We've been looking the doc/channel.txt and
doc/modules.txt in the source, but that's not a very complete source of
info :)
Thanks a lot for your attention.
--
Atly.
Alvaro Palma
2006 Mar 07
1
Changing REINVITE status of the channel dynamically
I've an Asterisk server running in my office, which forwards all
long distance calls to a third party SIP service using an extension rule:
exten => _1XX0.,1,Dial(SIP/{EXTEN:4}@external_sip_server.com)
(1XX0 is the international calls rule for Chile)
Also, in my sip.conf, I've defined canreinvite=yes to decrease the
network load to the server caused by the RTP.
However, the external
2006 Mar 23
0
Problem with INVITE's being sent
I've being testing a couple of GrandStream ATA 286 which with no reason
start responding 486 Busy to all new incoming INVITES. They are
connected to an Asterisk installation as SIP client. Running ethereal
between them, I could notice that, for some reason unknown for me at
this time, Asterisk sends some stranges INVITE's AFTER the communication
has been established and acknowledged