Hello List,
I have scenario as follows,
1. A call comes to queue.
2. Available agent will answer the call.
3. BridgeEvent wil be generated in AMI with channel1 and channel2.
4. Parse channel1 and channel two from the event and redirect them to a
meetme room,
Dialplan,
Exten =
In sip.conf you point to a context. For 101 en 102 you should point to a
context that allows using the trunk while for the other numbers you doesn't
grand this privilege .
Erik
Verstuurd vanaf mijn iPad
Op 24 mrt. 2011 om 16:58 heeft Peter den Hartog peterdenhar...@gmail.com het
volgende
Hello Users,
We have Thirdlane Multi tenant PBX system in production. Asterisk version
is 1.6.2.15.
Attendant transfer is working, but blind transfer is not working with
Grandstream (gxp-2000) phone.
We have read from google that it is a bug in Asterisk 1.6.2.15.
We saw the below links:
Maybe this helps:
https://issues.asterisk.org/view.php?id=18603
Arjan
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Jerry Geis
Verzonden: 20-03-2011 21:24
Aan: Asterisk Users Mailing List - Non-Commercial
Hi Group,
In Queue application, we have AGI,macro and gosub parameters that allow us
to perform some operations when Queue member gets connected with caller. But
it seems that right now there is no such mechanism (except CEL,AMI) for
situation where we want some operations to be performed when
Hi,
I'm trying to setup Asterisk so that:
1. I call a specific number that goes to a defined extension from my
phone (an external line).
2. Asterisk notes my phone number (the CLID) and hangs up without
picking up the call.
3. Asterisk initiates a call to my phone and prompts me for a passkey.
On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote:
Is there a better way of handling the post-hangup
processing?
Callfiles?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
On Sat, 26 Mar 2011 14:58:30 +0100, magnu...@inputinterior.se wrote:
Celluar Network - E1 - Avaya - OOH323 - Asterisk
Thanks for the tip.
So here's how it works:
1. The web app calls a script that uses AMI + Originate to send a call
to the Avaya PBX
2. Avaya is able to check that a number
Its not the Avaya that makes the call back, it is mobile.
-Ursprungligt meddelande-
From: Gilles
Sent: Monday, March 28, 2011 1:57 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Checking status of a cell phone
On Sat, 26 Mar 2011 14:58:30 +0100,
Is anyone using asterisk with fail2ban? I have it working except it takes
way more break-in attempts than what is set in maxretry in jail.conf
For example, I get an email saying:
The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts
against ASTERISK.
when maxretry = 5 in
On Mon, Mar 28, 2011 at 9:20 AM, vip killa vipki...@gmail.com wrote:
Is anyone using asterisk with fail2ban? I have it working except it takes
way more break-in attempts than what is set in maxretry in jail.conf
For example, I get an email saying:
The IP 199.204.45.19 has just been banned by
Hi!
Guess I am doing something totally wrong here: Some smart person could maybe
plz tell me what.
From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n
From dialplan i can “access” the variable “x” and see the value “5”
From dialplan i modify “x” to “8”.
But from AMI i
Hey all!
I'm trying to figure out how to have a queue accept an inbound caller's key
press to action on. At first I'm just trying to implement a Press 1 to
leave a voice mail announced and at any time in the queue, the user can
press 1 and go to the queue's voicemail. Later I'd like to have it
Hi,
Short version:
Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA
indication into a DAHDI/q.931 ALERTING signal when your ISDN provider
does not pass early media on receipt of an PROGRESS(8) indication?
Long version:
I have an Asterisk 1.6.2.18-rc1 based system with a
Thanks Tilghman for your response.
I have the following in my cdr_mysql.conf
I put it in sometime yesterday and did not have it till then.
However, it did not make any difference.
[columns]
static value = column
alias cdrvar = column
alias start = calldate
alias callerid = clid
alias src =
On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote:
Hi!
Guess I am doing something totally wrong here: Some smart person could
maybe plz tell me what.
From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value:
5\r\n\r\n
From dialplan i can “access” the variable “x” and
Hi,
is there a way to know if originate call channel ended the call *before*
connecting to context/extension/priority?
DIALSTATUS is empty, HANGUPCAUSE is always 16, nothing in SIP Headers
nor in AST_CONTROL_FRAME_[HANGUP|ANSWER]
Asterisk is 1.6.2.16
Thanks for any hint
--
Daniel
--
I did use Action: Getvar when i read it back in AMI.
On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote:
Hi!
Guess I am doing something totally wrong here: Some smart person could
maybe plz tell me what.
From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value:
5\r\n\r\n
From
Yes I followed directions on that page
Running Asterisk 1.6.1.22, anybody else experiencing this?
On Mon, Mar 28, 2011 at 8:32 AM, Andrew Latham lath...@gmail.com wrote:
On Mon, Mar 28, 2011 at 9:20 AM, vip killa vipki...@gmail.com wrote:
Is anyone using asterisk with fail2ban? I have it
Don't know then, that's all I've got far ya today mate, sorry
On 3/28/2011 8:18 AM, magnu...@inputinterior.se wrote:
I did use Action: Getvar when i read it back in AMI.
On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote:
Hi!
Guess I am doing something totally wrong here: Some smart
Hi Asterisks Team,
I am getting the error below after getting a connection to a telco using
ss7. Anyone know how to solve it?
The link keeps coming up and down every 30 seconds.
Resetting CIC 3
[Mar 28 15:16:28] WARNING[20434]: chan_dahdi.c:11913 ss7_linkset: RSC on
unconfigured CIC 3
Received
On 28 Mar 2011, at 14:19, vip killa wrote:
Yes I followed directions on that page
Running Asterisk 1.6.1.22, anybody else experiencing this?
How often does fail2ban check the logs? It can only block that often, so if
more attempts happen in that time period it can't do anything until it knows.
this may be related with:
https://issues.asterisk.org/view.php?id=14662
El 28/03/2011 10:20, Sherwood McGowan escribió:
Don't know then, that's all I've got far ya today mate, sorry
On 3/28/2011 8:18 AM, magnu...@inputinterior.se wrote:
I did use Action: Getvar when i read it back in AMI.
On Thu, Mar 24, 2011 at 4:58 PM, Thomas Winter thowin...@googlemail.com wrote:
Hi list,
I have an 44100 Hz file with human voice, stereo with 16Bit.
When convertig this to 8 kHz, mono I loose a lot of quality and have
some ground noise. I tried several sox options but without success.
Can
fail2ban checks the logs every second. Does asterisk buffer log output?
On Mon, Mar 28, 2011 at 9:27 AM, Steven Howes steve-li...@geekinter.netwrote:
On 28 Mar 2011, at 14:19, vip killa wrote:
Yes I followed directions on that page
Running Asterisk 1.6.1.22, anybody else experiencing this?
I would be surprised that you did not always hang up the second channel you are
redirecting. Once you transfer one leg there is nothing connected to the second
leg so it goes away, I would think.
What we do is remember the agent number, transfer the caller, and then setup a
call to the agent
On 3/28/2011 7:54 AM, Louis Carreiro wrote:
Hey all!
I’m trying to figure out how to have a queue accept an inbound
caller’s key press to action on. At first I’m just trying to implement
a “Press 1 to leave a voice mail” announced and at any time in the
queue, the user can press 1 and go to
On 3/28/2011 10:02 AM, Jim Dickenson wrote:
I would be surprised that you did not always hang up the second
channel you are redirecting. Once you transfer one leg there is
nothing connected to the second leg so it goes away, I would think.
What we do is remember the agent number, transfer
Could be, do u think its a bug or do u think I am doing totally wrong?
I can easily reproduce it if any needs more info.
-Ursprungligt meddelande-
From: Sebastian
Sent: Monday, March 28, 2011 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hey Guys!
I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming
calls..
Not working
[from-pstn]
exten = s,1,Answer()
same = n,Playback(hello-world)
same = n,Hangup()
Working...
[from-pstn]
exten = _,1,Answer()
same =
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Monday, March 28, 2011 11:04 AM
To: asterisk-users
Subject: [asterisk-users] s extension not working
Hey Guys!
I have asterisk 1.8.x and somehow my 's'
Hi
What is special control 16?
I am getting this error quite often --
special control 16, then for some reason it puts on hold and then logs is full
of Asked to transmit frame type ulaw, while native formats is 0x8 (alaw)
read/write = 0x8 (alaw)/0x8 (alaw)
Both peer and trunk have same
If i use 's' then i got following error. This scenario is back to back
asterisk connected on PRI line (T1). for testing purpose i calling from one
asterisk to other and i want to land call on 's' extension.
shirley*CLI
-- Extension '7527' in context 'from-pstn' from '7623' does not
Hi,
Googling, I came across this document
http://www.cytek.biz/roller/designbox/entry/asterisk_diversion_and_history_info
which says History-Info header is supported in asterisk.
Unfortunately, some details are missing (aka asterisk version).
Reading latest 1.8 changelog does say much about
Uhm
That's because you're being passed 7527 as the extension, so it won't
match s
On 3/28/2011 11:38 AM, satish patel wrote:
If i use 's' then i got following error. This scenario is back to
back asterisk connected on PRI line (T1). for testing purpose i
calling from one asterisk to
@Sherwood,
I was also thinking about that But then how 's' extension match any unknown
number ? Like when call coming from PSTN then how IVR picked up...?
-Satish
Date: Mon, 28 Mar 2011 12:58:28 -0500
From: sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re:
On 3/28/2011 1:33 PM, satish patel wrote:
@Sherwood,
I was also thinking about that But then how 's' extension match
any unknown number ? Like when call coming from PSTN then how IVR
picked up...?
-Satish
The 's' extension does not match anything other than 's'. If your sip
On Monday 28 March 2011 07:57:10 Eric W. Davenport wrote:
Thanks Tilghman for your response.
I have the following in my cdr_mysql.conf
I put it in sometime yesterday and did not have it till then.
However, it did not make any difference.
Did you reload after making the change to the
- Original Message -
Thanks for providing these - can you just clarify your policy on the
following:
- file locations - same layout as the regular Debian packages?
Yes, same layout.
- upgrade policy - is it intended that someone who has Debian 6 with
the existing Asterisk 1.6
On Mon, 28 Mar 2011 14:12:09 +0200, magnu...@inputinterior.se wrote:
Its not the Avaya that makes the call back, it is mobile.
I thought the way you handled things, is that Asterisk would call your
cellphone through the Avaya PBX just to check whether the cellphone is
in_use/busy. At what point
Wow... completely missed that. It was right there in the text. Sorry and thanks
Sherwood!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan
Sent: Monday, March 28, 2011 11:07 AM
To:
I was a little unclear, it is not the cell phone that does the call-back, it
is the cell-phone-network.
We can define 3 traffic-cases per cell-phone:
1) If cell-phone wont anser in x seconds call number a.
2) if cell-phone is busy call number b.
3) if cell-phone is unavailable call number c.
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