On 6/28/18 5:31 AM, bilal ghayyad wrote:
Hello;
Is it possible to configure one button on the IP Phone (like Polycom
or general SIP Phone) to indicate (at the phone display) that the line
(the line that is connected for FXO port) is busy or not? If it is not
busy, the user can press on the
Hello;
Is it possible to configure one button on the IP Phone (like Polycom or general
SIP Phone) to indicate (at the phone display) that the line (the line that is
connected for FXO port) is busy or not? If it is not busy, the user can press
on the button to place outside call.
Also, is it
Hi Joshua
Thanks for you explanation.
I found another problem. When I reject call by group limit, the CCSS do not
work, only if call limit on sip peers is set to 1. There is a way to using
ccss when call limit is different to one?
Thanks in advance.
[image: Sua Foto]
Hi
I need to set the number of incoming calls to one, but the outgoing calls
should be unlimited. I think the busylevel parameter is for it(incoming
calls), but not works. My config is:
cat sip.conf
[general]
[template](!)
qualify=yes
cc_agent_policy=generic
cc_monitor_policy=generic
On Wed, Aug 12, 2015, at 09:34 AM, Rafael dos Santos Saraiva wrote:
Hi
Kia ora,
I need to set the number of incoming calls to one, but the outgoing
calls
should be unlimited. I think the busylevel parameter is for it(incoming
calls), but not works. My config is:
snip
The busylevel
- Non-Commercial Discussion
Subject: Re: [asterisk-users] busy() not setting PRI_CAUSE
The description of busy() in the asterisk documentation wiki states:
This application will indicate the busy condition to the calling channel.
Wouldn't 'indicate the busy condition' on a PRI channel imply
Okay, I think I need a sanity check here - If I call a person that's on the
phone, I should get a busy signal.
Now more specifically, a call comes into the pbx via PRI. The destination
dialplan runs busy(20). Now, the PRI causecode should get set to 17 (user
busy) so that the originating end
-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 09, 2014 8:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] busy() not setting PRI_CAUSE
Okay, I think I need a sanity check here - If I call a person that's on the
phone, I
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 09, 2014 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] busy() not setting PRI_CAUSE
Generally if you want to send a cause 17
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, April 13, 2012 7:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] BUSY vs. CONGESTION
I have two lines, fax voice.
I usually call out on fax line (to have voice line available)
I need to set
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, April 18, 2012 12:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] BUSY vs. CONGESTION
If you are dialing out
I have two lines, fax voice.
I usually call out on fax line (to have voice line available)
I need to set the dial line based on dial-status. When I try to call out on
fax line and it is receiving a fax will I get a BUSY or CONGESTION signal?
What is the difference in dial plan condition:
On 06/01/2011 06:28 PM, Steve Davies wrote:
On 1 June 2011 15:10, randall rand...@songshu.org wrote:
On 06/01/2011 03:55 PM, randall wrote:
On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
Hi all,
After running fine for a few months now
On 06/01/2011 01:12 PM, Karsten Wemheuer wrote:
Hi randall,
Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall:
i get the following errors:
pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
of span 2
Your telco provider
On 06/01/2011 05:42 PM, Tzafrir Cohen wrote:
On Wed, Jun 01, 2011 at 04:10:34PM +0200, randall wrote:
On 06/01/2011 03:55 PM, randall wrote:
On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
Hi all,
After running fine for a few months now
Hi all,
After running fine for a few months now asterisk seems to hang
frequently , still functioning but the DAHDI channels seem busy (users
report a busy signal when calling or being called)
A reboot will allow it to run for another day or maybe 2 or 3 till the
problem occurs again.
On Wed, Jun 1, 2011 at 11:36 AM, randall rand...@songshu.org wrote:
Hi all,
After running fine for a few months now asterisk seems to hang
frequently , still functioning but the DAHDI channels seem busy (users
report a busy signal when calling or being called)
A reboot will allow it to
On 06/01/2011 09:04 AM, mahesh katta wrote:
On Wed, Jun 1, 2011 at 11:36 AM, randall rand...@songshu.org
mailto:rand...@songshu.org wrote:
Hi all,
After running fine for a few months now asterisk seems to hang
frequently , still functioning but the DAHDI channels seem busy
On Wed, Jun 1, 2011 at 1:07 PM, randall rand...@songshu.org wrote:
On 06/01/2011 09:04 AM, mahesh katta wrote:
On Wed, Jun 1, 2011 at 11:36 AM, randall rand...@songshu.org
mailto:rand...@songshu.org wrote:
Hi all,
After running fine for a few months now asterisk seems to
i get the following errors:
pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
of span 2
Your telco provider has crc on or off , that is not matching with
your server cross check with them.
and this problem solve 4
On Wed, Jun 1, 2011 at 1:30 PM, randall rand...@songshu.org wrote:
i get the following errors:
pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel
of span 2
Your telco provider has crc on or off , that is not matching
On 06/01/2011 10:07 AM, mahesh katta wrote:
On Wed, Jun 1, 2011 at 1:30 PM, randall rand...@songshu.org
mailto:rand...@songshu.org wrote:
i get the following errors:
pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel
of
Hi randall,
Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall:
i get the following errors:
pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
of span 2
Your telco provider has crc on or off , that is not matching
On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
Hi all,
After running fine for a few months now asterisk seems to hang
frequently , still functioning but the DAHDI channels seem busy (users
report a busy signal when calling or being called)
A reboot will allow it to run for
On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
Hi all,
After running fine for a few months now asterisk seems to hang
frequently , still functioning but the DAHDI channels seem busy (users
report a busy signal when calling or being
On 06/01/2011 03:55 PM, randall wrote:
On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
Hi all,
After running fine for a few months now asterisk seems to hang
frequently , still functioning but the DAHDI channels seem busy (users
report
On Wed, Jun 01, 2011 at 04:10:34PM +0200, randall wrote:
On 06/01/2011 03:55 PM, randall wrote:
On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
Hi all,
After running fine for a few months now asterisk seems to hang
frequently ,
On 1 June 2011 15:10, randall rand...@songshu.org wrote:
On 06/01/2011 03:55 PM, randall wrote:
On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
Hi all,
After running fine for a few months now asterisk seems to hang
frequently , still
Hi,
I'm having an issue with busy detection, the busy is not being detected.
Asterisk: 1.6.2.13
DAHDI: 2.4.0
Chandahdi: busydetect=yes, busycount=2
Indications zone = us, with the modifications for my country for busy:
425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off)
We have an employee who works from home. We sent her a SIP phone to work as an
extension off our Asterisk 1.6 system, but her DSL service is so bad she was
dropping calls all the time. It's not just a tuning or QoS issue. Her service
is simply unreliable.
She had a POTS line installed and I
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins
Sent: Thursday, October 21, 2010 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Busy detection
Of Danny Nicholas
Sent: Thursday, October 21, 2010 9:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Discussion'
Subject: Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6
Try changing KkTt to rKkTt. This should generate a phony ring until the call
is picked up or stops.
--
_
-- Bandwidth and Colocation Provided
Rob Many thanks for the pointer - I was missing limitonpeers=yes in the
general section - Sorry I didn't say version (1.4.33.1) etc forgot with
frustration ;-)
Paddy
--
_
-- Bandwidth and Colocation Provided by
On Fri, 2010-07-16 at 17:34 +0100, Paddy Grice wrote:
Seems BLF only work on called extensions - is there a way to show busy
for the calling extension?
You don't say what version of asterisk you are running this on, or have
any config snippets, so difficult to say what might be wrong.
Check
Hi all
A quick question about busy lamps
I have lamps working 'sort-of' on my GXP2000 lamps flash with ringing and
go solid red when call gets answered but stay green when a call is made from
the extension.
Setup is Ext 200, 201, 202, each monitor the other two
when 200 calls 202 - the BLF
Hi,
I'm running Asterisk 1.4.26.3 and I've noticed an interesting problem
when trying to play a Busy tone over a IAX trunk from the PSTN.
It seems as though Busy(20) returns non-zero immediately (it does not
wait 20s), so the caller never hears the busy tone, but
the call just appears to hang up.
Hi,
I have two asterisk boxes AB connected together via IAX.
Phones register to Asterisk box A, and Asterisk box B is the PSTN connection.
When dialing a number from a phone registered to A that DAHDI returns as BUSY,
the Busy(20) application returns immediately instead of playing the busy tone.
Using 1.4 svn, I want to implent the busy application.
With the following dialplan:
[inboundqueue]
exten = _X.,1,Answer()
exten = _X.,n,Goto(dropcall,1)
...
exten = dropcall,1,Busy(10)
exten = dropcall,n,hangup()
If I call any number in the inboundqueue, I get the following:
[Oct 1
Julian Lyndon-Smith wrote:
Using 1.4 svn, I want to implent the busy application.
With the following dialplan:
[inboundqueue]
exten = _X.,1,Answer()
exten = _X.,n,Goto(dropcall,1)
...
exten = dropcall,1,Busy(10)
exten = dropcall,n,hangup()
If I call any number in the
Hi,
Previously i was using asterisk 1.4 with freepbx installation.
To try the 1.6 version i installd anc configured everything..
Just one thing didnt work so far..
I am using grandstream 2000 and it has a line busy indicator for chef
secretary phones.
But now, this feature does not work.
I can see
@lists.digium.com
Subject: [asterisk-users] busy lamp filed
Hi,
Previously i was using asterisk 1.4 with freepbx installation.
To try the 1.6 version i installd anc configured everything..
Just one thing didnt work so far..
I am using grandstream 2000 and it has a line busy indicator for chef
secretary
Oguzhan Kayhan wrote:
Hi,
Previously i was using asterisk 1.4 with freepbx installation.
To try the 1.6 version i installd anc configured everything..
Just one thing didnt work so far..
I am using grandstream 2000 and it has a line busy indicator for chef
secretary phones.
But now, this
Oguzhan Kayhan wrote:
actually further to my last email another related question :
when programming the side buttons what are people using - ring groups or
individual extensions, other ?
I have setup sip devices with one series of extension numbers (ie every
line button on the phone,
I have this weirdness as well - depending on the phone (all gxp2000's)
I either get steady green regardless of sip registered or not OR no
green ever, and red always works as expected. Never yet seen green
follow the sip registration of that device like I would expect.
not sure if my message
for that or not?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan
Kayhan
Sent: Thursday, March 19, 2009 10:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] busy lamp filed
Hi,
Previously i was using
11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] busy lamp filed
Probably same thing I did.
In the GXP2000 BLF setup, set the account field to the Line that relates
to the extension you are trying to monitor. You can't monitor just any
old
Hi Ira,
for Aastra phones I have done this application to resolve busy/xfer
transfer:
extensions.conf
===
exten = _1X,1,GotoIf($[${SIPPEER(${EXTEN}|curcalls)}1]?free:busy)
exten = _1X,n(free),Dial(SIP/${EXTEN},,tTr)
exten = _1X,n,Hangup()
exten
Hi all,
maybe I find the problem and the solution.
I move the following parameters on section [general]:
[general]
port=5060
bindaddr=0.0.0.0
context=default
language=it
limitonpeers=yes
notifyringing=yes
and then on SIP account I put this:
[intphones](!)
type=friend
qualify=yes
host=dynamic
Marco Sambo schrieb:
Anyone know how to use busy-level parameter or some other useful parameters?
call-limit=2
busy-level=1
?
busy-level is not in Asterisk 1.4 of course.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk:
Ok, I read it.
Thank u. For busy on SIP I use also the Asterisk peer function SIPPEER with
field CURCALLS.
2009/3/17 Philipp Kempgen philipp.kemp...@amooma.de
Marco Sambo schrieb:
Anyone know how to use busy-level parameter or some other useful
parameters?
call-limit=2
busy-level=1
At 01:29 AM 3/17/2009, you wrote:
But there is another little problem. On Aastra phone (on other
phones I don't try yet), the xfer button doesn't work, until I set
call-limit=2, but making this I find the phone not busy.
As far as I can tell on my Aastra phones it takes 2 links to complete
a
Hi,
I have a question. How can I configure my sip.conf to make a SIP phone busy
on incoming and outcoming calls? I explain my problem.
When SIP phone receive a call and then I try to call that phone, I find it
busy.
When SIP phone make a call and I try to call that phone, I find it avaible
and it
On Mon, 16 Mar 2009, Marco Sambo wrote:
Hi,
I have a question. How can I configure my sip.conf to make a SIP phone busy
on incoming and outcoming calls? I explain my problem.
When SIP phone receive a call and then I try to call that phone, I find it
busy.
When SIP phone make a call and I
2009/3/16 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Mon, 16 Mar 2009, Marco Sambo wrote:
Hi,
I have a question. How can I configure my sip.conf to make a SIP phone
busy
on incoming and outcoming calls? I explain my problem.
When SIP phone receive a
On Mon, 16 Mar 2009, Olivier wrote:
2009/3/16 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Mon, 16 Mar 2009, Marco Sambo wrote:
Hi,
I have a question. How can I configure my sip.conf to make a SIP phone
busy
on incoming and outcoming calls? I explain my
I use Xlite and Asterisk.
Now, everything was working fine till yesterday.
But when my agent tried to login to asterisk through xlite, I see below line
sin CLI :
== Manager 'sendcron' logged on from 127.0.0.1
-- Got SIP response 486 Busy Here back from 192.168.0.17
Channel
Hi,
I have a snom 360 connected to two asterisk servers(both 1.6.0.5), via
two identities. Each asterisk server runs a queue and snom is a
member of queue in both servers. Currently when snom is receiving call
from one asterisk server, it can still receive a call from the other
asterisk, because
One simple thing that comes to my mind is to have the SNOM connected to only
one server, and send calls to from the queue on the second server to the
first server, so that you can enforce a acall limit.
l.
2009/2/19 Rajkumar S rajkum...@gmail.com
Hi,
I have a snom 360 connected to two
Rajkumar S schrieb:
I have a snom 360 connected to two asterisk servers(both 1.6.0.5), via
two identities. Each asterisk server runs a queue and snom is a
member of queue in both servers. Currently when snom is receiving call
from one asterisk server, it can still receive a call from the
On Thu, Feb 19, 2009 at 8:28 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
Easy solution: Disable call waiting on the phone.
But asterisk will attempt a call since it's status is idle, and will
generate events which will confuse ADM I am using to display a url
for call.
Advanced
Rajkumar S schrieb:
On Thu, Feb 19, 2009 at 8:28 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
Easy solution: Disable call waiting on the phone.
But asterisk will attempt a call since it's status is idle,
Unfortunately yes.
and will
generate events which will confuse ADM I am
asterisk-users@lists.digium.com
Sent: Thursday, February 19, 2009 6:46:20 AM
Subject: Re: [asterisk-users] Busy status of a snom connected to two asterisk
servers?
One simple thing that comes to my mind is to have the SNOM connected to only
one server, and send calls to from the queue on the second
Hi!
Advanced solution: Use local channels as queue members and Custom
hints. You could build a mechanism (outside of Asterisk) to sync
the states of your Custom hints between both servers.
I am already using local channels and will explore hints. I have not
used it till now, any hints
On Thu, 19 Feb 2009, Philipp Kempgen wrote:
Rajkumar S schrieb:
and will generate events which will confuse ADM I am using to display a
url for call.
ADM?
Asterisk Desktop Manager. http://adm.hamnett.org/
core show function DEVICE_STATE (on 1.6) is a good start.
Thanks.
raj
sorry if i ask it again, but where can i find the patch for enable
busy-level/limit in 1.4 ?
thanks
On Tue, Nov 18, 2008 at 12:09 PM, nik600 nik...@gmail.com wrote:
Thanks, is it possibile to retrieve a patch from Asterisk trunk? how?
On Tue, Nov 18, 2008 at 11:54 AM, Steve Howes
nik600 schrieb:
sorry if i ask it again, but where can i find the patch for enable
busy-level/limit in 1.4 ?
http://www.voip-info.org/wiki/view/Asterisk+sip+busy-level :
Contrary to popular belief this is only available in Asterisk
trunk and Asterisk 1.6
Philipp Kempgen
--
Hi to all
the busy-level / busy-limit setting in sip.conf is available for
Asterisk 1.4.22 ?
This is a piece of my sip.conf:
[202]
type=friend
secret=202
host=dynamic; This device registers with us
username=202; Username to use when calling this device
On 18 Nov 2008, at 10:30, nik600 wrote:
the busy-level / busy-limit setting in sip.conf is available for
Asterisk 1.4.22 ?
http://www.voip-info.org/wiki/view/Asterisk+sip+busy-level
___
-- Bandwidth and Colocation Provided by
Thanks, is it possibile to retrieve a patch from Asterisk trunk? how?
On Tue, Nov 18, 2008 at 11:54 AM, Steve Howes [EMAIL PROTECTED] wrote:
On 18 Nov 2008, at 10:30, nik600 wrote:
the busy-level / busy-limit setting in sip.conf is available for
Asterisk 1.4.22 ?
Carlos Chavez escribió:
Thank you. Unfortunately the phone Company in Mexico is not very
helpful when it comes to those services.
On Tue, 2008-05-20 at 16:48 -0500, Tilghman Lesher wrote:
On Tuesday 20 May 2008 16:08:19 Carlos Chavez wrote:
The problem is that I do not
On Tue, May 20, 2008 at 05:42:15PM -0300, Vin?cius Fontes wrote:
You could simply short-circuit the two wires of the line. The telco
will interpret that as a busy line.
Please
600 ohm, 1 watt wirewound resistor. If it's on a short loop, you might
have to go as high as 5 watts. POTS lines
On Tue, May 20, 2008 at 6:05 PM, Carlos Chavez [EMAIL PROTECTED] wrote:
Thank you. Unfortunately the phone Company in Mexico is not very
helpful when it comes to those services.
On Tue, 2008-05-20 at 16:48 -0500, Tilghman Lesher wrote:
On Tuesday 20 May 2008 16:08:19 Carlos Chavez
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would like
to busy out that line. Since that line is the head of the hunt group I
cannot simply disable that channel, I need to busy the line so calls
will
You could simply short-circuit the two wires of the line. The telco will
interpret that as a busy line.
Other than that, you could do this on extensions.conf:
[context]
exten = s,1,Answer()
exten = s,n,Busy()
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
-
The problem is that I do not have physical access to the server. The
other problem with that solution is that the first person to dial the
main number will always get a busy tone and only then can someone else
get to another line.
I need to leave the line offhook until further
On Tuesday 20 May 2008 16:08:19 Carlos Chavez wrote:
The problem is that I do not have physical access to the server. The
other problem with that solution is that the first person to dial the
main number will always get a busy tone and only then can someone else
get to another line.
Thank you. Unfortunately the phone Company in Mexico is not very
helpful when it comes to those services.
On Tue, 2008-05-20 at 16:48 -0500, Tilghman Lesher wrote:
On Tuesday 20 May 2008 16:08:19 Carlos Chavez wrote:
The problem is that I do not have physical access to the
Just create an extension like this:
[busyoutline]
exten = 111,1,Answer()
exten = 111,n,Busy()
then drop a .call file like this:
Channel: Zap/1/111
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: busyoutline
Extension: 111
Priority: 1
The above should work, however keep in mind if the
On Tue, 20 May 2008, Carlos Chavez wrote:
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would like
to busy out that line. Since that line is the head of the hunt group I
cannot simply disable that
Hello, all!
I've noticed a peculiar situation and I am hoping someone can shed
some light on it for me. We have an Asterisk (1.4.18 ) box talking to
the world via Zaptel on a PRI from a telco (USA). I have an extension
that returns busy signal (fast-busy or regular busy) (using US tones).
On Wed, 16 Apr 2008 08:40:42 -0500, Mark Gimelfarb [EMAIL PROTECTED]
wrote:
why do cell phones and Gizmo both detect busy tones and terminate the
call? Is that a standard behavior?
It *is* standard procedure for a cellphone to terminate a call immediately
it discovers that the called number
What country are you in?? Yes, it is common for cell phones to
disconnect the call if they receive CONGESTION, but not BUSY.
Horwich IT Services (Godwin Stewart) wrote:
It *is* standard procedure for a cellphone to terminate a call immediately
it discovers that the called number is busy. It
I'm in the US, so I was originally using the US tones.
Looks like I'm getting a disconnect with both CONGESTION and BUSY. In
fact, I wasn't actually using Congestion() and Busy(), I just did
Playtones() for both of those. There is no reason to send PRI messages
to cell phones, is there? The
Hi, I use:
Trixbox-2.2.4
FreePBX-2.3.1.0
Asterisk-1.2.17
BRIstuffed-0.3.0-PRE-1y-e
Zaptel-1.2.19
..with two ISDN cards, often but occasionally the dial out failed but is
possible to receive external call.
My zapata.conf conf is:
[trunkgroups]
[channels]
language=it
context=from-pstn
Hello all,
I was wondering what will be the proper way to manage BUSY state
notification in presence once call-limit, incominglimit and all those
settings are gone.
I'm using GROUP_COUNT for call limiting in Asterisk 1.4.13 but I have no
idea how to set up the settings needed for BUSY
Peter Galiovsky wrote on Wednesday, January 09, 2008 9:39 AM
I want the user to be presented as busy if he has at
least one call active, be it incoming or outgoing. How
should I set things up to achieve this?
I have a very similar need. We are using call queues and would like to
have only 1
At 01:42 PM 1/9/2008, you wrote:
I have a very similar need. We are using call queues and would like to
have only 1 call presented to our trouble call reps at a time, but would
like to give them the ability to initiate an outgoing call on a
different line (even while on an incoming call). We are
On 1/9/08, Don Pobanz [EMAIL PROTECTED] wrote:
Peter Galiovsky wrote on Wednesday, January 09, 2008 9:39 AM
I want the user to be presented as busy if he has at
least one call active, be it incoming or outgoing. How
should I set things up to achieve this?
I have a very similar need. We
Hi,
I've a huge problem with the following:
Setup:
Asterisk 1.4.11
I've got two Thomson ST2030s in an queue. After a while Asterisk logs the
following if somebody calls the queues number:
- Got SIP response 486 Busy Here back from 172.10.3.31
-- SIP/office1-0823d190 is busy
--
Erik Wartusch wrote:
- Got SIP response 486 Busy Here back from 172.10.3.31
I see that response when someone presses the DND button on our Polycom
phones.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
Hi,
I have a problem with busy tone detection.
the problem is busy tone with different length tone and silence! Means:
Busy tone = 400/400,0/345,400/230,0/520
400 on
345 off
230 on
520 off
Repeat
I try in Zapata.conf to enable busy tone detection by this way
busydetect=yes
callprogress=no
Hi,
I have a problem with busy tone detection.
the problem is busy tone with the different length tone and silence! Means:
Busy tone = 400/400,0/345,400/230,0/520
400 on
345 off
230 on
520 off
Repeat
I try in Zapata.conf to enable busy tone detection by this way
busydetect=yes
Hi,
Please discribe me how we define busy/hang/answer detection with PRI E1
channels.
Since busydetect, callprogress, busycount giving falts hangup and call drops
what is the solution on PRI channels?
--
Thanks Regards,
Vidura B. Senadeera.
___
You can use the hangupcause variable which us the pri cause code
supplied when a call is ended over a PRI line. For example this is the
maco we use to dial a number over PRI.
[macro-pridial]
exten = s,1,GotoIf($[${ARG1:0:2} != 00]?noint)
exten =
Vidura Senadeera wrote:
Hi,
Please discribe me how we define busy/hang/answer detection with PRI
E1 channels.
Since busydetect, callprogress, busycount giving falts hangup and call
drops what is the solution on PRI channels?
PRI channels have call supervision and Asterisk will see the
I'm having a problem with my IAXy not always connecting to my Asterisk box.
When I pick-up the phone plugged in to the IAXy I get a busy signal. I
have to hang-up the phone and wait a few seconds after the orange LED goes
out and then try again.
When this happens I don't see any connection
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 27 Nov 2006, at 19:57, Frank Tarczynski wrote:
I'm having a problem with my IAXy not always connecting to my
Asterisk box.
When I pick-up the phone plugged in to the IAXy I get a busy
signal. I
have to hang-up the phone and wait a few
Is yours S100i or S101i? I've seen the same issues with the S100i and then
noticed that SIP isnt as bad as Digium makes it seem (probably to market
IAX) and want to say the issue is caused by heat. I do believe the S101i is
the same exact hardware, uses the same firmware but is just designed not
When I pick up the handset to make a call, SIP 483 BUSY is returned
to the server and the below statement works. However, when I'm in a call,
the BUSY status is not returned on my Cisco 7960 phones.
Is there a way to make this work when I'm actually on a call? Maybe
DIALSTATUS is not the way
1 - 100 of 269 matches
Mail list logo
