Did you install the MySQL libraries?
Debian's command is - apt-get install libmysqlclient15-dev
Andy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthias
Urlichs
Sent: 27 November 2008 16:05
To: asterisk-users@lists.digium.com
Subject: [SPAM] - Re:
Have you set port 2 as 'NT' in the mISDN config file (not the Asterisk one)?
Also, you will probably need to set it to ptmp.
You need to configure them in misdn.conf (the Asterisk one this time).
Here's the tail of my misdn.conf (4 x BRI):
[trunks]
ports = 1,2 ; physical port numbers
It looks like you are trying to dial out on your 'NT' instead of your
'TE'.
Try changing Dial(DAHDI/g1/${EXTEN:1}); to Dial(DAHDI/G1/${EXTEN:1});
Oh, and I'd use mISDN for BRI as DAHDI always gave me problems.
HTH
___
-- Bandwidth
Apart from you were dialling out on your inbound context and
vice-versa.
The best advice I can give now is to change to mISDN - as this
is proven to work with v1.4 and v1.6.
Actually - have you tried putting the 100ohm termination on for
your NT
go for v1.6 and it's built in fax detection :).
I hope that makes sense.
Cheers
Andy
Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF
Just seconding Freddi's idea - as it makes perfect sense. Otherwise we
could quite easily start testing a call that hasn't actually finished
yet.
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asterisk-users mailing list
...or something along the lines of a setting a variable (like we do for
MONITOR_EXEC)...
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,b410p
and looping from 1 port to another - Email found in subject -Email found
in subject - Email found in subject
2008/12/1 Andrew Thomas [EMAIL PROTECTED]
Apart from you were dialling out on your inbound context and
vice-versa.
The best advice I can give now
For a ptmp setup where you have multiple phones.
Or even a single phone if the port is set to ptmp.
Proof of this point is the way I am using our B410P card. Ports 1 and 2
are TE (ptp) and ports 3 4 are NT (ptmp).
I have a single ISDN modem connected to port 3 and the B410P would not
even
asterisk-users@lists.digium.com has now been added to the filters white
list!
Anyway, 100ohm termination isn't required for ptp - but is required for
ptmp.
I know the DAHDI package(s) no longer include make b410p - hence the
reason it is included in the docs.
Give this a go:
exten = s,n,MYSQL(Query resultid ${connid} SELECT `name` FROM `cnam`
WHERE `ani` = '${CALLERID(number)}')
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- but that puts out a lot more information than
is needed for simple logging (and requires something to prune and store
the events in a database of some sort).
Any thoughts?
Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph
as it is and create a new CEL (Call Event
Logging) module (optional in modules.conf) that puts out (and does not
accept) call event information (ie. a one-way fire-and-forget output
from Asterisk).
Hope that makes my positiion a little clearer.
Cheers
Andrew Thomas
You are looking in the wrong place.
Have a look at the following:
Core show function QUEUE_WAITING_COUNT
-= Info about function 'QUEUE_WAITING_COUNT' =-
[Syntax]
QUEUE_WAITING_COUNT(queuename)
[Synopsis]
Count number of calls currently waiting in a queue
[Description]
Returns the number
Design
On Fri, Dec 5, 2008 at 8:26 AM, Andrew Thomas [EMAIL PROTECTED]
wrote:
In summary: Leave CDR exactly as it is and create a new CEL (Call
Event
Logging) module (optional in modules.conf) that puts out (and does not
accept) call event information (ie. a one-way fire-and-forget output
from
Quote : I couldn't disagree more. The CDRs is the MOST reliable
source for billing purposes
...at the moment. Have you read about Greyman's transfer problem?
If you are billing customers purely on the CDR output from Asterisk -
then good luck to you :).
I'd disagree. In some cases a event based system would be the best
solution, but in systems with high call volumes, scanning through events
looking for the proper billing information and parsing them would be a
hard job compared to CDRs.
That's just it - you wouldn't be 'scanning' any CDR's -
Pardon me,
Granted ;).
I have created realtime stats package that's based on CDR, you see new
info immediately after call leg/event is over
I see what you are saying but can you show hold-times etc? For example,
call comes in to A, A puts call on hold, A dials B, B answers A, A
transfers call
at).
Who wrote that?
[snip the rest of the reply]
Andrew Thomas wrote:
[snip]
Like I said earlier - the CDR's aren't reliable enough for a billing
platform (as you've rightly pointed out) but are OK for very basic
call
logging (something the customer can look at).
Why didn't you place your reply
]
---
Andrew Thomas wrote:
I'd disagree. In some cases a event based system would be the best
solution, but in systems with high call volumes, scanning through
events
looking for the proper billing information and parsing them would be a
hard job compared to CDRs.
That's just it - you
Message -
From: Andrew Thomas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 5 December, 2008 13:49:59 GMT +00:00 GMT Britain, Ireland,
Portugal
Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design
Have a look at ATA devices. Any good VoIP equipment reseller should have them
available.
http://www.voip-info.org/wiki-ATA is worth a look.
Cheers
Andy
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: 05 December 2008 14:17
Q: What is the most annoying thing in e-mail?
Spam and useless replies when I've already asked for this topic to be
closed *sigh*.
-- -Original Message-
-- From: [EMAIL PROTECTED]
[mailto:asterisk-users-
-- [EMAIL PROTECTED] On Behalf Of Gergo Csibra
-- Sent: 05 December 2008 14:41
Well, it seems this opened one large can of worms.
Anyway, just to repeat my previous plea - and to echo David's request - can we
please stop all this 'top post' rubbish and move on with our lives?
Thanks and Merry Christmas
Andy
-Original Message-
From: [EMAIL PROTECTED]
I've just spotted another interesting CDR 'feature'. Data calls don't
get flagged as such. In other words - if I make an ISDN modem call to
another ISDN modem via. the PSTN, the source and destination channels
are set correctly (as is everything else in the current CDR) but there
is no record if
-11 at 11:37 +, Andrew Thomas wrote:
-- I've just spotted another interesting CDR 'feature'. Data calls
-- don't
-- get flagged as such. In other words - if I make an ISDN modem
call
-- to
-- another ISDN modem via. the PSTN, the source and destination
-- channels
-- are set
Use setvar=variablename=value
Eg: under [client1]
setvar=dialplan=NZ
Then just reference ${dialplan} in your extensions.conf
Cheers
Andy
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On
Since when can you segment PRI channels off at the telco end? I know
you could do with DASS - but I'm not aware you can do it with PRI.
Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF
I can only assume it's a T1 thing - as E1's tend not to have that
facility. Oh well, you live and learn :)
Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF
If you are connecting to BRI lines then you should be TE - not NT.
You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried
the new release).
HTH
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] libpri and NT-Point to multi-point
Hello Andrew,
2008/12/17 Andrew Thomas a...@datavox.co.uk
If you are connecting to BRI lines then you should be TE - not NT.
Yes of course, you're right.
I was mostly referring
the 123456.
--
-- Julian
-- Andrew Thomas wrote:
-- Isn't that the ${exten} number? In other words, the number
called.
--
--
--
-- -- -Original Message-
-- -- From: asterisk-users-boun...@lists.digium.com
-- [mailto:asterisk-users-
-- -- boun...@lists.digium.com] On Behalf
-- Where are you actually doing the diverting? In Asterisk at the
telco
-- exchange?
...or at...
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Andrew Thomas a...@datavox.co.uk
I have piggy backed a few PBX's off the back of a B410P (4 x BRI) card with no
problems. The ones I used for testing were the Avaya IP Office, Siemens
Hi-Path/Hi-Com and various old Panasonics.
All I had to do was to turn on the 100ohm termination on my S0 ports
Isn't that the ${exten} number? In other words, the number called.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Tony Mountifield
-- Sent: 17 December 2008 10:17
-- To:
You don't really need to use any local MTA if you use the sendEmail
script.
I got it from - http://www.caspian.dotconf.net/menu/Software/SendEmail/
This actually works by 'talking' directly to any SMTP server - even
remote ones (I use our Exchange server for our e-mails).
HTH
Andy
JFYI - I run (successfully) agx-addons with 1.4.22 and Etch.
Make sure you have the right version of SpanDSP installed (as well as the tiff
libraries).
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
--
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 22 December 2008 09:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4withLenny
Hi Andrew,
2008/12/22 Andrew Thomas
December 2008 10:58
-- To: Asterisk Users Mailing List - Non-Commercial Discussion
-- Subject: Re: [asterisk-users] Install app_rxfax and app_txfax in
-- 1.4withLenny
--
-- On Mon, 22 Dec 2008 23:46:28 Andrew Thomas wrote:
-- ...described in the README file ;).
--
-- SpanDSP - 0.0.4 pre
Try http://forums.vtiger.com/viewtopic.php?t=14314
Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF
-- -Original Message-
-- From: asterisk-users-boun
Have you got termination set correctly?
I have a B410P working with 2 x NT and 2 x TE ports successfully.
I had to turn the 100ohm termination on on the NT ports (even though I
have them set as PTP in mISDN.conf).
HTH
-- -Original Message-
-- From:
-- In many cases, this just isn't possible. While it would be nice
to
-- have all
-- posts in the King's English, a great many users are in locales
which
-- don't
King's English???
Anyway - to quote Ralph Wigham Me fail English? That's unpossible!.
Put faxdetect = none in the misdn.conf and you'll be fine.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Vieri
-- Sent: 06 February 2009 12:40
--
Use Set(CALLERID(num)=99) instead of using CALLERID(all).
Remember to set this BEFORE you Dial.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Vieri
-- Sent: 06 February
(), of course.
--
-- I've read somewhere that the misdn debug message:
--
-- -- P[ 1] -- TON: Unknown
--
-- may mean that the carrier did not recognize the caller id I set.
Is
-- this true?
--
--
-- --- On Fri, 2/6/09, Andrew Thomas a...@datavox.co.uk wrote:
--
-- Use Set(CALLERID
Hello,
I'm just wondering if anyone has fixed the 'InUseRinging' problem.
* v1.4.23.1
Ta
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...@lists.digium.com [mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Philipp Kempgen
-- Sent: 09 February 2009 11:50
-- To: Asterisk Users
-- Subject: Re: [asterisk-users] InUseRinging
--
-- Andrew Thomas schrieb:
-- I'm just wondering if anyone has fixed the 'InUseRinging
svn co https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons agx-ast-addons
./build_sh from the trunk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 10 February 2009 18:35
To:
Hi helpers,
I seem to have a problem of intermittent DTMF tones being played during
a conversation.
Eg: Extn 100 takes an inbound call and all is fine. Except, at an
undetermined time the person on extn 100 will here a DTMF tone for no
apparent reason (it's not the caller pressing buttons).
Hi Francois,
I am using the latest *, dahdi/zaptel and libpri (1.4-current).
This happens with both Zaptel and Dahdi and various versions of *
(1.4.22.1 and 1.4.23).
So, even the latest 'stable' would seem to have a problem.
Cheers
Andy
-- -Original
Thanks for this Jared (look - back on topic!). I've just ordered the
print and downloaded the pdf. It does look very good (the bits I've
managed to read so far).
I'll give everyone my humble and worthless opinion of it when I get to
read it some more.
Andy
-- -Original
Have a look for agx-ast-addons and spandsp.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert McGilvray
Sent: 06 March 2009 01:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fax
Hi all,
I am having trouble setting the signalling method for the B410P using
DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or
'bri_net' - but it doesn't mind having 'pri_cpe' etc.
ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown signalling method
'bri_net'
Dahdi -
of execute
--
-- strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI
-- Telephony'
--
-- It's LibPri install before of Dahdi package?
--
-- JL.
--
-- El Lun, 9 de Marzo de 2009, 6:36 pm, Andrew Thomas escribió:
-- Hi all,
--
--
-- I am having trouble setting the signalling
You could always run a shoutcast server and stream from that.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: 09 March 2009 19:02
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Don't forget to mention that the BRI signalling method doesn't work in 1.4 (and
probably 1.2) ;).
Andy
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 10 March 2009 12:51
To: Asterisk Users
Post up your chan_dahdi.conf and we'll fix it :)
Hint - you are missing : 'signalling = fxo_ks' and 'signalling = fxs_ks' from
it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aqua Man
Sent: 10 March
You now need to compile and install SpanDSP-0.0.6pre3 at least (AGX has been
changed).
After you've done that - try AGX again.
HTH
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 11 March
-- Subject: Re: [asterisk-users] DAHDI and B410P (BRI)
--
--
-- I wish it was available too - I have just had to back dahdi out of a
-- system and revert to misdn after a whole day of testing.
--
-- PaulH
--
--
-- Andrew Thomas wrote:
-- I have LibPri installed and working (.../wPRI
Please explain (in English) what you mean by ANI.
Thanks
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith
-- Sent: 12 March 2009 10:21
-- To: Asterisk Users
I think I understand what you mean now. The biggest difference between
CLI and ANI is that ANI can't be blocked/withheld (like you can with CLI
by using 141). It also uses different signalling. This is mainly used
by law enforcement agencies to trace calls etc.
So, you want the number -
There are various ways of doing this.
You could use an analogue port/ATA and connect any good old fashioned
intercom to it (Pantel are a good make).
You can now get SIP intercom systems as well. I haven't tried on of
these - but they look good (and can contain a camera as well if needed).
HTH
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PBX to gate interface
How does a Push-to-talk intercom interface with Asterisk?
Andrew Thomas wrote:
There are various ways of doing this.
You could use an analogue port/ATA and connect any good old fashioned
intercom
Just to add
P[ 1] Transmitting 128 samples 2 misdn
P[ 1] writing 128 bytes 2 asterisk
P[ 1] Sending :160 bytes 2 MISDN
P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0
P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0
P[ 1] Transmitting 128 samples 2 misdn
P[ 1]
This sounds like you have pri_net instead of pri_cpe in Zapata.conf.
When inserting the cable going into TE122 into an ISDN phone, the
phone
works perfectly.
Any suggestions would be greatly appreciated :-)
___
-- Bandwidth and Colocation Provided
as well. Oh the joys of Asterisk and hotdesking!
HTH
Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF
-Original Message-
From: asterisk-users-boun...@lists.digium.com
, a bit of a hack, but it works for me ;) I know that this
won't work for 1.6, but we are coming up with an alternative plan using
Minivm
Julian
Andrew Thomas wrote:
The quick answer is 'no'.
It is not currently possible to monitor 'hints' for Agents - as an
Agent
never actually dials out
Mailing List - Non-Commercial Discussion
Cc: aster...@dotr.com
Subject: Re: [asterisk-users] Grandstream 2010 and blinky lights
On 8/7/09 8:52 PM, Andrew Thomas wrote:
That's exactly the way I do it as well :D
-Original Message-
From: asterisk-users-boun
Why are you putting semi-colons at the end of every line? The dialplan
isn't written in PHP ;).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L.
Kline
Sent: 15 July 2009 23:46
To:
I have just the thing in PHP.
Drop me a personal e-mail and I'll whiz it over.
Andrew Thomas
Technical Services Manager
a...@datavox.co.uk
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF
-Original Message-
From: asterisk
It appears I opened some flood gates when I offered my 'alternative'
version.
So, rather than send hundreds of e-mails out - here's the link :
http://www.dv-ip.com//downloads/files/misc/weather.txt
Any questions - just 'yell'.
Andrew Thomas
Technical Services Manager
a...@datavox.co.uk
DataVox
'host=dynamic' is your problem - as this allows any IP address to register as
that friend - assuming they know the password/username combination.
Why not simply have group 1 as 'secret=pass123' and group2 as 'secret=pass456'?
Just don't tell group 1 uses the password for group 2 - and
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Remember, setvar in sip.conf makes that variable a global variable.
Andrew Thomas
Technical Services Manager
Juan C. Crespo R
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen
Sent: 24 July 2009 14:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Music on hold based on user
Andrew Thomas schrieb:
I do this using the setvar facility in sip.conf.
eg. setvar=MOH
[peer]
defaultip=xxx.xxx.xxx.xxx
host=xxx.xxx.xxx.xxx
deny=0.0.0.0/0.0.0.0
allow=xxx.xxx.xxx.0/255.255.255.0 read what you've put!!! The
'allow' should be 'permit' as Jared already told you (and he should know
what he's talking about).
insecure=port,invite
Underscore won't help as that's for pattern matching.
Under the sip conf, have you tried adding 'fromuser=8001187e0' to the
[8001187e0] bit?
I have this in my Sipgate setup and it works. Worth a try.
Cheers
Andy
-Original Message-
From: asterisk-users-boun...@lists.digium.com
V1.6.1.0
[9290740]
type = peer
username = 9290740
fromuser = 9290740
secret = you-wish!
host = sipgate.co.uk
fromdomain = sipgate.co.uk
insecure = port,invite
context = inbound
caninvite = no
canreinvite = no
nat = yes
disallow = all
allow = ulaw
allow = alaw
dtmfmode = info
qualify = 5000
That
This sounds more like the alarm system putting pulses/tones on the line
(maybe the alarm has a dialler/anti-cut-line-detection?
So, as the alarm is adding stuff AFTER the asterisk box - I doubt you
will see anything on the PC itself.
-Original Message-
From:
The only way around the 'auto-logout' problem I found was to call a script when
agents login. This script checks to see if they are already logged in or not -
then, if they are, it does whatever I want (I manually log them off the other
phone first - you could play a message instead).
HTH
...and did you switch the termination dip switches over (on the NT ports of the
B410P card)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy
Sent: 17 August 2009 07:56
To:
exten = did,1,Answer
exten = did,n,Playtones(ring)
exten = did,n,Wait(10)
exten = did,n,StopPlaytones()
exten = did,n,BackGround(sound file)
did = the DID number as presented and note the '1' before Answer.
This works for me.
exten = 820055,1,Answer()
exten = 820055,n,PlayTones(ring)
exten =
Hello all,
Just wondering if anyone ever solved the Aastra 50-BLF limit when used
with Asterisk (any flavour)?
I know it's not strictly and Asterisk question - but I'm sure there's
plenty of you out there using Aastra's on the end.
Cheers,
Andrew Thomas
dCAP #1473
Apologies if this has been asked before.
Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1?
Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the
source files for 1.4.34 over the top of the existing 1.4.24.1 files.
Obviously, I will need to keep my config files
Hi Danny,
I understand (and welcome) the separate src directories. This would
allow me to 'revert' should I feel the need (assuming I can just
re-compile over each one). I just need to know if I can re-compile over
the existing first.
Thanks for your reply.
-Original
into their respective folders on your system. Then just start
asterisk. If you need to revert, stop asterisk, run make install in the old src
directory, then start asterisk.
Ryan
On Mon, Jul 26, 2010 at 9:45 AM, Andrew Thomas a...@datavox.co.uk wrote:
Hi Danny,
I understand (and welcome) the separate src
This is a problem with extconfig.conf - not your res_ or cdr_ ones.
In your case - extconfig.conf probably contained something like
'sippeers = mysql,MyDBase,sippeers'. The 'problem' is that the middle
parameter is no longer for the database name - it is for the context in
res_mysql.conf. So,
As a side note to this - do NOT try and use Aastra's - as they tend to
crash after 50 BLF's!
Also, could you please send me (perhaps off-list to a...@datavox.co.uk)
your Yealink T28 findings - as I am a beta tester for them?
Cheers
Andy
-Original Message-
From:
Hi list,
Does anyone know how to send * a semi-colon from a realtime database. I
know that * uses the semi-colon as a 'seperator' - but I need to be able
to use one in a command. I know I can use \; in the non-realtime
configs, but this doesn't work in realtime.
Cheers,
Andrew Thomas
Technical
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime semi-colon
On 16 Sep 2010, at 12:56, Andrew Thomas wrote:
Does anyone know how to send * a semi-colon from a realtime database.
I know that * uses the semi-colon as a 'seperator' - but I need to be
able to use one
What happens if you put in a 'room' number?
Eg: exten = 8080,3,MeetMe(500|MDci)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
touati
Sent: 17 September 2010 14:24
To: Asterisk Users Mailing List -
a
conference number I guess, but i don't it's going to solve my issue.
actually I'm atill wondering is there a way to debug just Meetme app
output or the only way is turn the whole debug thing on?
On Mon, Sep 20, 2010 at 4:11 AM, Andrew Thomas a...@datavox.co.uk
wrote:
What happens if you put in a 'room
Downgrade your LibPri instead (1.4.10.2 is fine).
See https://issues.asterisk.org/view.php?id=17270 for more info.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
Sent: 29 September 2010 13:39
To:
The cause is bad programming. You can't go from an 's' to an '_X.' the
way you tried.
exten =s,1,Answer()
exten =s,n,Wait(1)
exten =s,n,Dial(DAHDI/3)
exten =s,n,Hangup
Is correct (that's why it works).
What is it you are trying to achieve?
-Original Message-
From:
${EXTEN:1:3}
http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/
asterisk-CHP-5-SECT-3.html#asterisk-CHP-5-SECT-3.6.3
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent:
What happens if you change to:
signalling=bri_cpe_ptp
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Sent: 01 October 2010 11:37
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
The D-channel isn't actually 'dropped' - it is put in to a 'power-save'
state.
See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to
'Activation / Deactivation' for more information.
Anyway - this is a known 'problem' -
https://issues.asterisk.org/view.php?id=17270
As there is no
saving mode ?
2010/10/7 Andrew Thomas a...@datavox.co.uk
The D-channel isn't actually 'dropped' - it is put in to a 'power-save'
state.
See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to
'Activation / Deactivation' for more information.
Anyway - this is a known 'problem' -
https
2 ways:
Read http://www.voip-info.org/wiki/view/Asterisk+AGI
or in PHP - system (asterisk -rx 'core restart now' /dev/null);
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe D'alessio
Sent: 29
2010, Andrew Thomas wrote:
Read http://www.voip-info.org/wiki/view/Asterisk+AGI
An AGI is executed in the context of a channel. Are you suggesting the
OP
write an AGI so he can call into his system to ask it to hang up all
channels?
--
Thanks in advance
For the Yealink - you can use a 'remote' XML file. The XML is stored on
a web server and is retrieved by the phone every time you press the
phones 'key'. This has the advantage of not needing the directory to be
pushed to the handset - and the handset always gets the latest version.
Of course,
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