Hi all,
I'm trying to write an application to dialout something then forward the
call to a context/exten depending on some parameters.
These questions may be trivial, not for me ;-)
---
Action: Originate
Channel: Local/1@internal
Exten: 384087
Context: SIP-UA-00128
Priority: 1
CallerID:
Hello List,
I have an Elastix 2 machine with digium fax modules (with license).
When I try to create an extension that also works with FAX, Asterisk does
not detect any incoming fax. Even when I use 'fax set debug', it does not
display anything.
It's Asterisk 6.2.x . Any ideas what can I do to
Hi
read indications.conf and you will get the solution of your question.
http://www.voip-info.org/wiki/view/Asterisk+config+indications.conf
On Wed, Nov 2, 2011 at 11:11 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello,
What is the method for changing the country for indications (eg.
After origination successfully complete and channel will be created
you probably should link ActionID and channel name.
Origination action will be next:
Action: Originate
Channel: Local/1@internal
Exten: 384087
Context: SIP-UA-00128
Priority: 1
CallerID: 601
ActionID: FFA02C6A03
Variable:
Dear Abdul Basit,
http://nerdvittles.com/index.php?p=784 works, I tested it few months back
and it works. Cant say if its still working or not.
On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote:
Any has Skype For Asterisk (SFA) license.
On Thu, Oct 20, 2011 at 3:45 PM, Diego Alejandro Sanchez Quiroga
diegosanch...@gmail.com wrote:
Very Good day to everybody, I had a problem with an implementation that
recently made in my Company, dealing in a Grandstream gateway
which will enable echo cancellation but until you hear noise
Hi,
I'm running Asterisk 1.8.7.1 on Gentoo. I set `core set debug 9` but don't
see any debug messages on the console. I do get the verbose messages from
ast_verbose. Is there something I need to configure to see these messages?
Regards,
Yahya
--
On Thu, Nov 17, 2011 at 16:31, Danny Nicholas da...@debsinc.com wrote:
Fax is detected at ring/start level. Check chan_dahdi.conf.
The chan_dahdi.conf contain faxdetect=both
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com]
I am running into an issue installing asterisk 10.0.0-rc1
I have centos 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:35 EDT 2010 i686 i686 i386
GNU/Linux installed.
I am at the point of trying to install the dahdi and I am getting the error
message when I do a make all:
You do not appear to
Hi list,
something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both
having an extension [115], one as type peer (caller side 1.4) and one as
friend (callee side 1.8). Phones from both location connect to Asterisk
from LAN. Router are Linux boxes.
Connection between the 2 sites
On 17/11/2011 13.11, Yaroslav Panych wrote:
exten =
384087,1,UserEvent(LinkOriginate,CHANNEL:${CHANNEL(name),ACTIONID:${ActionID}}
UserEvent application will generate event into AMI in form
Event: LinkOriginate
CHANNEL: channle-name (channel id created by asterisk)
ACTIONID: FFA02C6A03
Yes, her extension is now on one line key, but this does not mean she
cannot make a new call if she is already on a line. If she is on a call
and needs to call someone else she would put the first call on hold,
then the New Call soft key will appear. You can press that to get
dialtone and
Excuse me if I am off the mark here, I don't have the chance to read too well into your post. But if it is what I think it is, I remember I had a similar situation a few years ago, and I ended up having to create an internal table in my code, so that I could keep track of the channel ids + action
Il 17/11/2011 15.46, Danny Nicholas ha scritto:
You have two items to join with in the example you provided; #1
local/1@internal-99fd,1 is call 99fd on local/1. If you started 10
calls from local/1, they would all have differing - (99fd). #2
UniqueID - part 1 is a constant.
On 17/11/2011 19.45, c.savinov...@itntelecom.com wrote:
if it is what I think it is, I remember I had a similar situation a few
years ago, and I ended up having to create an internal table in my code,
so that I could keep track of the channel ids + action ids .
Which is exactly what I'm doing
The easiest thing to do is to create userevents in your dialplan to passed to
AMI details you want to key off of. In the original originate you can set so
variable that you pass to various macros and what have you. These then generate
userevents that AMI can use to track the flow of the call.
On 11/17/11 3:30 AM, ik wrote:
I have an Elastix 2 machine with digium fax modules (with license).
When I try to create an extension that also works with FAX, Asterisk does not
detect any incoming fax. Even when I use 'fax set debug', it does not display
anything.
It's Asterisk 6.2.x . Any
So I did a little more digging and found a real simple answer:
${CHANNEL(audionativeformat)}
tells me 'ulaw' or 'siren14' and lets me pick the right file extension
for the record function.
On Tue, Sep 13, 2011 at 5:19 PM, Tom Browning ttbrown...@gmail.com wrote:
Sorry if this is an obvious
Thanks Danny for the replay.
My requirement is registeration with expire 0 need to send from
asterisk to unregister the trunks, but the CLI command sip unregister
peer or sip qualify peer will not send any REGISTER packet out,it
just remove the entry of incoming registration .
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