hi,all
as you know, we can use MYSQL command to visit mysql database
but if i use other database like Oracke,sybase,etc, Could i use MYSQL command ?
if not, is there any other alternative could do the same
function(visit database in dailplan)?
Thanks!
--
Best regards,
Sucan
--
hi , all
what's wrong with this command?
exten = 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
as i got the error:
-- Executing [...@95040:1] GotoIfTime(SIP/1001-0099,
11:00-14:00|mon|wed|*|*?1:3|1) in new stack
[Jan 20 11:21:11] WARNING[16804]: pbx.c:4118 get_range: Invalid day
Hello,
I am developing the free SIP softphone (audio+video) for Windows. And I have
some issues with asterisk 1.6 compatibility. I am new in asterisk, so I
guess, I have no enough skills to config asterisk properly. I have enable
tcp transport mode and register client, but can not make a call.
On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun bit...@gmail.com wrote:
exten = 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
but what should i do. if i want to set seperate weekdays,like mon,wed.
not continuous weekday like mon-fri.
I couldn't find any reference to multiple, non-contiguous
22 jan 2010 kl. 10.13 skrev Vitali Fomine:
Hello,
I am developing the free SIP softphone (audio+video) for Windows. And I have
some issues with asterisk 1.6 compatibility. I am new in asterisk, so I
guess, I have no enough skills to config asterisk properly. I have enable
tcp transport
Sorry to bump this one...
Anyone have any other ideas on it?
Regards
Steven Davison
Net Technial Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Davison
Sent: 21 January 2010 08:41
To:
2010/1/22 Randy R randulo2...@gmail.com:
On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun bit...@gmail.com wrote:
exten = 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
but what should i do. if i want to set seperate weekdays,like mon,wed.
not continuous weekday like mon-fri.
I couldn't find
report 491 Request Pending on invite message. Why server report the
error?
The server reports this when we already have an INVITE to handle.
Please check that you did not transmit two invites without waiting for
a response and sending an ACK from your softphone.
Yes, here is two INVITEs (I
22 jan 2010 kl. 11.51 skrev Vitali Fomine:
report 491 Request Pending on invite message. Why server report the
error?
The server reports this when we already have an INVITE to handle.
Please check that you did not transmit two invites without waiting for
a response and sending an ACK from
Hi,
I'm connecting a Linksys SPA3102 to 3 different PSTN analog lines.
With only one of those, CID is shown.
Beside that, everything is working OK.
Lines have different providers and/or locations.
All are located in France and CID Detection Method is ETSI FSK / Bell 202.
If I'm connecting a
I've been asked by my company to setup a conferencing system to support up
to 400 people on a conference calls, where all users will be dialling in
frpm the PSTN. I am exploring using Asterisk meetme to do this. I have two
questions in relation to this:-
For Meetme conferences is it better to
Hi,
Am Donnerstag, den 21.01.2010, 21:08 -0500 schrieb hugolivude:
Hi,
I'm having trouble getting feature codes to work in Asterisk 1.4.21.2.
Features.conf contians this:
blindxfer=##
atxfer=*2
automon=*1
disconnect=**
I'm really most interested in getting disconnect to work so that
Hi all!
I have installed a quite old Asterisk 1.6.2.0-rc2 with latest DAHDI on
Ubuntu 9.10 from repository. It is working now but mysql logging is very
strange. All calls have logged in mysql cdr table, which is fine, but
disposition is 'NO ANSWER' even if I had talked on phone. Duration is
http://www.snom.com/en/products/ip-phones/snom-870-touchscreen-voip-phone/
http://www.aastraintecom.com/cps/rde/xchg/SID-3D8CCB6A-935A2A1B/30/hs.xsl/38707.htm
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about
Anyone got any subjective (!) views on the merits of these two ranges ,
using asterisk 1.4 ?
I need to supply approx 30 handsets to a new client, with the senior
managers (6) having some slightly more managerial phones than the base
phones which will be used for one line only.
TIA
Julian
--
Hello,
I would like to see this as well, from an Asterisk CLI log perspective
with sip debug turned on.
The .log file for login and invite is attached, I have use asterisk -vr
command. Is it correct?
Yes, here is two INVITEs (I have missed first invite before), but the
server
respond 401
Thanks for the GREAT tip. Changing to a single feature digit of * for
blindxfer worked which led me to changing featuredigittimeout = 2000. Now I
can do blindxfer w/ ##. Why I didn't try changing featuredigittimeout long
ago is beyond me! *blush* Thanks again.
One thing that still doesn't
Hello,
I am using Queue application with multiple agents in each queue. I
want to set the CDR(userfield) for each cdr based on the agent
answering the call. Is it possible to do this?
Thanks
--
_
-- Bandwidth and Colocation
On Fri, Jan 22, 2010 at 1:26 PM, Julian Lyndon-Smith aster...@dotr.com wrote:
Anyone got any subjective (!) views on the merits of these two ranges ,
using asterisk 1.4 ?
The choice of phones is crucial. Setting aside my tastes, you really
need to get a couple of typical users to try them
Since userfield is a variable, that would be a qualified yes.
Here's how I use it in my dialplan
- exten = NXX,1,noop(Answer())
- exten = NXX,n,Verbose (Dial DAHDI g1 ${EXTEN})
- exten = NXX,n,Dial(DAHDI/g1/${EXTEN},60,miKkTtg)
- exten = NXX,n,Set(CDR(userfield)=${CDR(userfield)}
Let me know if you figure it out, I am interested in this as well.
Right now I have a cron job that executes this every 5 minutes..
UPDATE cdr SET userfield = MID( dstchannel, 1 , LOCATE( '-', dstchannel )-1)
WHERE disposition = 'ANSWERED' AND LOCATE( '-', dstchannel ) 0 and lastapp
= 'Queue'
I know this is not what you need, but you might postprocess recordings to
raise the volume level. I know this is not optimal but it's a start.
l.
2010/1/21 Scott Gifford sgiff...@suspectclass.com
Hello,
We are recording our calls to queues by putting the appropriate options in
our
hugolivude wrote:
I have this in features.conf:
[applicationmap]
testfeature1 = #9,caller,Playback,tt-monkeys
testfeature2 = #8,callee,Playback,tt-monkeys
and this in the context where the dial takes place:
include = featuremap
include = applicationmap
You need to re-read the sample
On Friday 22 January 2010 04:06:29 Zhang Shukun wrote:
2010/1/22 Randy R randulo2...@gmail.com:
On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun bit...@gmail.com wrote:
exten = 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
but what should i do. if i want to set seperate weekdays,like
On Friday 22 January 2010 02:06:08 Zhang Shukun wrote:
as you know, we can use MYSQL command to visit mysql database
but if i use other database like Oracke,sybase,etc, Could i use MYSQL
command ?
if not, is there any other alternative could do the same
function(visit database in dailplan)?
Maybe I'm saying something stupid, but I thought this was what
shared_lastcall would do with a leastrecent strategy.
; shared_lastcall will make the lastcall and calls received be the same in
; members logged in more than one queue.
; This is useful to make the queue respect the wrapuptime of
I want to do something like this
exten = 1234,n,Queue(6000,c)
exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
agent who answered the call
exten = 1234,n,Hangup
On Fri, Jan 22, 2010 at 8:01 PM, Danny Nicholas da...@debsinc.com wrote:
Since userfield is a variable, that would
Unless you're using MixMonitor to actively monitor calls, the optimum
solution would indeed (IMO) be postprocess adjusting with SOX or something
similar.
That being said,
From this link;
http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf
I got this notion
monitor-format =
On Friday 22 January 2010 01:49:13 Giedrius Augys wrote:
2010/1/21 Tilghman Lesher wrote:
On Thursday 21 January 2010 09:51:13 Giedrius Augys wrote:
Is it possible to free idle connections? When limit was 40, I had lost
part of data. My asterisk version is 1.6.0.20 .
We intentionally
Julian Lyndon-Smith wrote:
Anyone got any subjective (!) views on the merits of these two ranges
, using asterisk 1.4 ?
I need to supply approx 30 handsets to a new client, with the senior
managers (6) having some slightly more managerial phones than the
base phones which will be used for
2010/1/22 Tilghman Lesher tles...@digium.com
On Friday 22 January 2010 01:49:13 Giedrius Augys wrote:
2010/1/21 Tilghman Lesher wrote:
On Thursday 21 January 2010 09:51:13 Giedrius Augys wrote:
Is it possible to free idle connections? When limit was 40, I had
lost
part of data. My
Hi,
I know having Asterisk aware of Polycom Do No Disturb state wasn't working
before (1.4), but is this working in any recent version? Is there any
custom way of doing this?
Regards,
Mike
--
_
-- Bandwidth
Hello,
I managed to get it working. Seems like i was overwriting fields used in
computation of the digest response. Once i turn off authentication the
call flow works perfectly. I will need to make necessary modifications
to work with digest authentication.
As a next step i will be
On Fri, 2010-01-22 at 20:25 +0530, Deep D wrote:
I want to do something like this
exten = 1234,n,Queue(6000,c)
exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
agent who answered the call
exten = 1234,n,Hangup
Actually because the user will hangup within the Queue
On Friday 22 January 2010 09:53:33 Giedrius Augys wrote:
2010/1/22 Tilghman Lesher tles...@digium.com
On Friday 22 January 2010 01:49:13 Giedrius Augys wrote:
2010/1/21 Tilghman Lesher wrote:
On Thursday 21 January 2010 09:51:13 Giedrius Augys wrote:
Is it possible to free idle
On Fri, 22 Jan 2010, Zhang Shukun wrote:
as you know, we can use MYSQL command to visit mysql database
but if i use other database like Oracke,sybase,etc, Could i use MYSQL
command ?
ODBC will do what you want.
Personally, I'd vote for an AGI using whatever C API your DB provides
-- like
Le 13/01/2010 09:57, Benoit a écrit :
Le 12/01/2010 16:35, Tilghman Lesher a écrit :
On Tuesday 12 January 2010 04:44:36 Benoit wrote:
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while the later works prefectly i can't really make
The 'h' extension worked. Thanks.
The other option of 'memebermacro' did not work. On the asterisk
console I could see that the macro is executed and cdr userfield is
set when agent answers the call, but the userfield doesn't show up in
the generated cdr.
Also I had one more question. Doesn't
setinterfacevar=yes
Needs to be under each queue
What does your dialplan end up looking like?
I would like to add to mine, and stop running a cron job..
exten = 5000,1,Answer
exten = 5000,n,Queue(5000|rn)
exten = 5000,n,VoiceMail(5000,u)
exten = 5000,n,Hangup
-Original Message-
From:
Hey hey!
Anyone got any subjective (!) views on the merits of these two ranges
, using asterisk 1.4 ? I need to supply approx 30 handsets to a new
client, with the senior managers (6) having some slightly more
managerial phones
* Let the customer test and decide himself
* Polycom: great
http://twitpic.com/z8n36
On Fri, Jan 22, 2010 at 8:11 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hey hey!
Anyone got any subjective (!) views on the merits of these two ranges
, using asterisk 1.4 ? I need to supply approx 30 handsets to a new
client, with the
Randy R wrote:
http://twitpic.com/z8n36
*snicker*
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I have worked on many snom phones over the years I have never had
a snom phone go bad...
I have repaired stuck screens and overheated sticky bits but all in
all snom are great phones. I recently showed my personal phone to
some people including a VoIP engineer that fell in love with my snom
- Andrew Latham lath...@gmail.com wrote:
Having demo phones is priceless. Sometimes I show off the phones I
like with some phones I don't like to show the end users why it
matters... Bad screens, cost, features, color (snom snow rocks!),
etc...
Anybody else remember the white
On Fri, Jan 22, 2010 at 05:06:17PM -0300, Andrew Latham wrote:
I have worked on many snom phones over the years I have never had
a snom phone go bad...
I have had about 10 in the last 12-18 months, I had 1 with a fault hand
set plug - the reseller replaced it. Other wise they have been
We run with ringall strategy, and have had shared_lastcall on for six
months. afaik, this only shares the last call data for things like
wrapup time; it definitely does not share how the queues with ringall
hand off their queued calls.
As for the patch below, I have since tried it on 1.4.29,
Hi,
I was trying to use 2 of asterisk servers and interconnected, one of them as
a peer to other sever (configured in sip.conf), so all the calls to server 1
will just be passed to server 2 (has PRI Card, TE 412P, only one PRI
connected), i was sending calls to server 1 and that would send to
When you configure the Siemens gigaset handsets (I have S685IP), there
is a single option for all handsets to use either the POTS interface or
VOIP as the default outbound destination - you then need to add a dial
suffix if you want to use an alternate outbound route.
Does anyone have any
Hi all,
what is the signalling of IAX?
Currently I want to connect two switch through IP using asterik signaling,
and I want to transfer SS7 over IP (between the 2 asterisk), will IAX can
transfer SS7 signalling through IP (like TDMoIP does) If no which solution
can I use?
see below the
From my experience, unless you have another base station for sets you would
want to configure separately it is not possible.
I may be wrong, hopefully.
Sent from my Android phone
-Original Message-
From: Alan Lord (News) [alansli...@gmail.com]
Received: 1/22/10 7:01 PM
To:
On 01/22/2010 07:12 PM, mickael ropars wrote:
Hi all,
what is the signalling of IAX?
Currently I want to connect two switch through IP using asterik
signaling, and I want to transfer SS7 over IP (between the 2 asterisk),
will IAX can transfer SS7 signalling through IP (like TDMoIP does) If
Does anyone have any suggestions as to how to make just *one* of the
DECT handsets only use the POTS but others default to their Asterisk SIP
subscriptions?
Hi Al,
I've played with the Siemens Gigaset in the past and I don't recall being
able to do this.
Chris
--
Hi all,
First I'm very new. I want to build an Interactive Video-voice Response
system. There is number of choice I have found so far: FreePBX, TriBox,
Asterisk.
Which is the best in my case? and what do i need to build such IVVR
system?
Thanks.
Quyps
--
Hi
I was wondering if you can use the base station as a outbound pots
connection for asterisk.
I currently have a tdm410 to do fxs/fxo ports and would like to get rid
of it, I used to use a spa3102, but it only had 1 fxo (telephone
connector). I like the idea of the siemans but I would like to
I just added a line with 'h'extension.
My dialplan is like this
[mycontext]
exten = s,1,Queue(6000)
exten = h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
On Sat, Jan 23, 2010 at 12:14 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
setinterfacevar=yes
Needs to be
Quyps,
It looks like you mis-read the picture.
Asterisk is the core, it need to be there regardless you use FreePBX or
Tribox.
FreePBX is a GUI web interface to manage asterisk. Itself is not an IP-PBX.
Trixobx, still based on the Asterisk + freePBX, adds some more additional
applications based
On 01/22/2010 09:02 PM, Pham Quy wrote:
Hi all,
First I'm very new. I want to build an Interactive Video-voice Response
system. There is number of choice I have found so far: FreePBX, TriBox,
Asterisk.
Which is the best in my case? and what do i need to build such IVVR
system?
All use
Hi!
I was wondering if you can use the base station as a outbound pots
connection for asterisk.
I currently have a tdm410 to do fxs/fxo ports and would like to get rid of
it, I used to use a spa3102, but it only had 1 fxo (telephone connector).
I like the idea of the siemans but I would
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