Displaying 20 results from an estimated 7000 matches similar to: "Hook Flash INFO messages"
2010 Feb 20
2
Sending a hook flash to a DAHDI channel
I've got a piece of CPE equipment that has an FXS port that I have tied
to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a
standard telephone connected to it, I get a dialtone. If I dial a digit,
and send a hookflash, the device will provide a dialtone back for the
next available channel on the device.
I'm trying to recreate this same behavior with Asterisk,
2003 Oct 29
2
Call transfering, conferencing
hello,
my questns are about few * functionality.
1) how can I make call tranfer. Not call parking.
If I'm talking with some one a I want to tramnfer call to the another
extension, to the other person.
2) how can I make call confernece. Not Meetme
If I'm talking with some one and I want to join another person to our talk .
I haven't found this in any manual :(
hudecof
--
mail:
2003 Jul 11
3
mgcp problems
I strange error messages when using mgcp and ata186 .
This session is simply dial into 600 demo extension - echo test
...
Handling request 'NTFY' on aaln/1@10.0.1.19
Transmitting:
200 29 OK
to 10.0.1.19:2427
-- Endpoint 'aaln/1@10.0.1.19-1' observed '0'
-- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode:
sendrecv
Posting Request:
RQNT 306
2010 Jun 26
2
Detecting hook flash in asterisk
Hello,
Is it possible to detect a hook flash in asterisk. I want to be able to
perform some functions an hook flash.
I have the following entry in features.conf which executes a Macro on
detecting key press '**'.
[applicationmap]
test => **,caller,Macro,testflash
Is it possible to do this action on hook flash?
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2008 Jun 17
1
looking for help / input with Blind transfer from asterisk to zap
List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap] (SIP/sv0071iv) answers.
Zaptel -> Asterisk -> SIP extension
SIP extension then blind transfers [from-sip]
---
SIP extension -> Asterisk -> Zaptel
During this whole process, the original channel off the trunk
(lineside T1) is
2006 Nov 23
1
asterisk 1.4 chan_h323, help please...
Hi,
My configuration is SipPhone<-->*1<--->*2.
My asterisk version is 1.4beta3.
I installed pwlib,openh323,chan_h323.
When i call from
SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2,
there is no audio.
Using 'rtp debug', I can see that rtp packets are
being received.
Rtp packets are being exchanged.
I also tested chan_ooh323, but to fail.
Can anyone recommand best
2003 Jul 10
2
OH323 + G729 + Go2Call
hi ..
i've just installed and licensed an instance of the G729 codec.
I am trying to connect through asterisk to Go2Call server ..
According to their info it involves dialling extension 729 on
voip01.go2call.com, to get the IVR.
my extensions.conf shows :
exten => s,2,Dial(OH323/h323:729@216.52.153.206)
which I think is correct, I have G729 enabled in the OH323.conf
file and it seems to
2004 Dec 29
2
RE: Hook/Flash, Hold, Call Waiting, Three Way Calling
Do you have threewaycalling and transfer set in your zapata.conf? Here's
mine (four TDM400's, seems to be working so far). I didn't do anything in
my extensions.conf for any of these features (what confused me at first is
the t and T options of the Dial application in extensions.conf are for
transfers via the # key), when you flash you get another dialtone that works
just like the
2003 Apr 14
2
Weirdness on "hookflash call pickup"
I'm sure dumb when it comes to describing things that happen on my system.
I'm making an outbound call on my ATA186 when another call comes in. I
first get the nasty CID screech and then the periodic call-waiting tone.
So far, so good.
Then I hookflash, and just like it's supposed to, the waiting caller is
on the line.
But during the duration of that conversation, my console
2003 Nov 27
6
Help for oh323
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the extention.conf like this:
exten => _87.,1,Dial(OH323/16.52.153.206)
oh323 dials out (I can ring a
2007 Dec 27
3
CDR
Hi Steve,
> .. I'll try to sort all this out, and then I'll attack
this
> problem. Hopefully, I get it all into svn before the next release of
> 1.4...!
Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling.
I for one
2005 Jan 20
4
softswitch dilemma
Hello everybody,
Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc.
Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.
2004 May 17
0
Zap callwaiting hookflash idiosyncracy/flaw?
Don't know what else to call this. Googling and some time on the IRC
channel haven't gotten me anywhere.
Here's the sitch, which is a bit complicated but is something my
customers are in fact encountering on an everyday basis:
1. Bob is on a Zap channel talking through the PSTN to Carol. Both have
the misfortune, like so many of us, of having LECs who do not offer
disconnect
2004 Dec 27
1
transfer: hookflash vs #
I think I've managed to figure out that there are two ways to transfer a Zap
call, using hookflash (defined in zapata.conf) or the # key (the t and T
options of the Dial command in the dialplan), but not why there are two ways
to do this, nor what the difference is between them. Is there something
that explains this?
thanks
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2009 Jan 10
3
Asterisk/GXW410x IP Analog Gateway
Hello All,
I am trying to setup a small system where Nextone Softswitch will send
traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog
Gateway but for some odd reasons the call are flashed back from
Grandstream to Asterisk and creating a Black loop...
I did follow the instructions provided by Grandstream support but it
doesn't seems to be working...
2003 Aug 06
2
Semi-newbie question "Softswitch" and Asterisk - Is there a difference?
I've been working in the VoIP industry for just a bit over a year now...
Mostly taking care of the underlying systems. I've now reached the
point where I'm being drawn more and more into the call processing side
of things. My background is in computer and "classic" telephony systems
(DMS250/MTX, DSC 400, T1, channel banks. telabs analog echo supressor
modules and
2007 Jun 29
4
asterisk call unique id in dialplan
Hi
how can i retrieve the call unique id of asterisk in the dialplan?
I have enabled the cdr logging on a postgres database.
In the table cdr each record has a field that assumes an unique id
(for example: 1141628669.51)
Can i retrieve this from the dialplan?
For example:
exten => 203,1,Answer
exten => 203,2,Set(CALLERID(name)=UNIQUE_ID - ${var_name_unique_id})
exten =>
2004 Dec 28
1
Asterisk / 183 message
Hello,
My company is doing some * testing with our Class 5 softswitch and had
some questions regarding ringback being provided to our PSTN users (off
--> on net calling)
Currently with MGCP subscribers, we know the PSTN ringing is provided by
a digital PBX for example, However, it looks like with SIP, our
softswitch is relying on MGCP signaling on our PSTN gateways to provide
ringback
2003 Oct 07
1
Digium FXO
Is it possible to send an external hookflash command to the Digium FXO
card from the asterisk PBX?
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2008 Oct 18
1
strange h323 delay issue
Hello,
I have a strange h323 issue. After executing command
"Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18
22:32:23. Meanwile I have sniffing traffic on port 1720. The call was
established just at Oct 18 22:33:03 (New H.323 Connection created.) and also
packet sniffer grabs the h323 invites at this time also. So my question is
what