Displaying 20 results from an estimated 600 matches similar to: "passing variables to h extension"
2009 Mar 26
3
Know who's logged in
Hi all,
For those of you people that use Agents (with Agentlogin, not
AgentCallbackLogin) on a call center, I have this need: when the agent
logs in, a channel keeps running all the time that the agent is logged
in to receive the incoming calls. How do I know which agent logged in
(code)? Right now, if I query the login channel, there is no information
about which agent is logged on:
#
2004 May 16
1
** Asterisk Sunday Morning News: Contribute to the community
Another Asterisk week has gone by. A lot of changes has been submitted into
the CVS head, only a few to CVS stable.
CVS stable only changes for bug fixes now.
* Using MGCP? Please step forward!!
-----------------------------------
There are a number of MGCP bugs and fixes in the bug tracker that needs more
activity. If you are using the MGCP protocol, please step forward and help us
fix this.
2009 Jan 28
2
How to retrieve a phone number from call forwarding?
Hi,
I'm very new to Asterisk and I have the following scenario.
1. Let's say I have a number of 1-222-222-2222 from my SIP service provider
(VoicePulse).
2. I point my phone, Verizon wireless cellphone (1-111-111-1111), voicemail
to the number provided by SIP service provider (1-222-222-2222).
3. I use another phone (1-333-333-333) to call 1-111-111-1111 and leave a
voicemail message.
2009 Feb 25
1
Stuck Parked Calls?
I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and
2017 Jul 29
1
rugarch package: VaRTest()
Dear all,
I want to backtest my Value at Risk output using the VaRTest() function in the rugarch package. I do not understand if the numeric vector of VaR which needs to be calculated is in negative or positive terms. Usually VaR is expressed in positive terms.
Do I have to use positive values for VaR in the VaRTest() formula?
Thanks for your help.
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2007 Feb 19
2
sip to sip ?
hi all
i've just setup an * box and want to test voip calling, initially from
sip user to sip user...
local sip users can call each other, no issues.
problem arises when i try and call a remote sip account, my * box
always returns "SIP/2.0 404 Not Found"
any ideas ?
2010 Apr 06
2
polarity reverse
Hi,
I have a problem with polarity reverse
this my dahdi config
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
2008 Mar 27
2
Calling users to the external domain using Asterisk
Hi All,
I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17
and using it to make SIP calls.
I have a configuration of Asterisk which serves the users in a
particular domain, say internal.com
I would like to make a SIP call from bob at internal.com to
charles at external.com
I have added the following lines in extensions.conf
exten =>
2014 Feb 24
1
Add SIPCALLID of egress leg to CDR
Hey all,
I've been fighting with this all morning, and I feel like this should be a
relatively simple task, but I just can't get it to work. I currently have
a very basic asterisk v11.6 setup with a single extension (a Bria
softphone) and a single sip trunk to my carrier.
What I'm trying to accomplish is simply adding the asterisk generated
SIPCALLID of the leg between asterisk and
2017 Dec 27
3
Answered time on channel
It seems that what ever I set in my answer handler does not show up in the
hangup handler. In order to do billing I can't rely on the g option where
the caller hangs up the call. Looks like I can either use h or a hangup
handler along with the shared function.
On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote:
> Don't use an 'h' extension, use
2007 Oct 08
3
get egress SIP call Id
Hi, Does anybody know how to get the SIP call ID of a "Dial" command?
Thanks in advance. Ray
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2007 Feb 25
2
Dialling ZAP channel from analogue
Hi,
Asterisk Version : 1.2.15
Card : TDM11B (1 x FXO , 1 x FXS)
I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP.
The problem comes when I try and make a outbound call.
Here is my extensions.conf :-
Code:
[incoming]
exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1)
exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)
2006 May 02
8
Zapata Telephony interface and torisa module error
Looking at my log file I found the following error:
May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196
May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000
May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded
May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error
May 2 12:00:45 debian
2018 Jul 13
2
Withholding Answer Supervision
Hi,
Is there any way of telling Asteirsk to withhold answer subversion on a
call till I call Answer.
My DP looks like this:
[incoming]
Exten => 18005551212,1,Noop()
same => n,Answer
same => n,Mset(__uid=${SIPCALLID})
same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV)
same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center
/n&Local/3 at
2003 Nov 18
2
SIP Context from domain?
Hi,
Is it possible to pick the context of a call from chan_sip based on the
domain of the To: header of the INVUTE? I've had a quick look throught he code
and can't see anything, I want to use the voicemail virtual hosting with
chan_sip. Can the sip domain be picked out with a global in extensions.conf?
This woud also solve my problem.
If not is there any specifc reason/restriction
2011 Feb 15
1
outbound call leg CALLID
Hello everyone
Is there a possibility to catch an outbound callleg ID for the follovong
scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ?
I can get inbound callid for asterisk1 with a ${SIPCALLID} in
extensions.conf or to look it up in cdrs field (are the same). But how about
outbound? I have all calls just forwarded through asterisk1, not answered
and for every call I
2006 Feb 08
2
Need to retrieve Call-ID from dialed SIP channel (w/o CDRs)
Is there a way to retrieve the Call-ID from a call made using the 'Dial'
command on a SIP channel without CDRs (i.e. variable) ?
Thanks,
- Darren
2010 Jul 05
1
SIP response 482 "Loop Detected"
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same.
Any ideas on how to overcome this problem as we dial
2011 Oct 22
3
Wine 1.3.31 fails to compile in git
I tested also with a clean git source. Something wrong with ole32.
Code:
-fPIC -Wall -pipe -fno-strict-aliasing -Wdeclaration-after-statement -Wempty-body -Wstrict-prototypes -Wtype-limits -Wwrite-strings -fno-omit-frame-pointer -Wpointer-arith -Wlogical-op -I/usr/include/freetype2 -g -O2 -U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=0 -o tmarshal_i.o tmarshal_i.c
2012 Jul 26
2
Call ID of the second call leg
Hello friends,
I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can
access the caller Call ID (fbasename field in voipmonitor cdr) looking at
the SIPCALLID variable in asterisk, but how can I access from within
asterisk the Call ID of the second leg of the call (the one originating
from asterisk to the destination peer)? is there a variable holding this
value?
Thank you