Discussion
Subject: [SPAM] - Re: [asterisk-users] queue moh - Email found in subject
Hello Andy,
Have you tried using SetMusicOnHold command before Queue command?
BR,
Ioan
On Wed, Jul 10, 2013 at 7:55 PM, Andrew Thomas a...@datavox.co.uk wrote:
Hi All,
Sorry if this has been covered already
Hi All,
Sorry if this has been covered already, but I don't tend to follow this
list as close as I should these days.
Problem is that if a call comes in to a queue without option 'r'
specified - moh plays as expected. Now, when that call is answered, all
is fine. Trouble comes when that person
The Debian command I use is:
apt-get install linux-headers-`uname -r`
That will get the bits you need and place them in /usr/src/.
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and
This is a brilliant idea. How do I contribute my attackers to this
list?
Cheers
Andy
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Huddleston
Sent: 22 September 2011 16:11
To: 'Asterisk
...@lists.digium.com] On Behalf Of Hans
Witvliet
Sent: 22 May 2011 22:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Fwd: FW: realtime mysql - p4]
On Fri, 2011-05-20 at 10:05 +0100, Andrew Thomas wrote:
Post your cdr_mysql.conf and res_mysql.conf and we'll take
Post your cdr_mysql.conf and res_mysql.conf and we'll take it from
there.
Don't forget to remove any 'private' info first (like passwords).
Cheers
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans
I would think that that is down to either your indications.conf (could
be wrong) or the handset itself.
I know most Yealink and GrandStream handsets let you change tones in
their individual config. Not too sure about others.
-Original Message-
From:
And why would you post a reply 5 days after my last post - and 4 days
after the threads last one?
Do you want to keep this thread going?
I suggest letting it die on it's own.
_
-Original Message-
From: asterisk-users-boun...@lists.digium.com
This sounds like you have it set for T1 somehow? Have you upgraded
anything lately? Other than that, a Trend tester will show the
problem(s) to you.
BTW - E1's are 32 channel (not 31). It's 30B+2D.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
https://issues.asterisk.org/view.php?id=15818
That's where I get it from.
If it contains errors, then why not report it there?
Cheers
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 13 May
Ah! Forgot about that.
Looks like your on your own Olivier.
Sorry
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif
Madsen
Sent: 16 May 2011 13:12
To: asterisk-users@lists.digium.com
Subject: Re:
Probably using XML - which is phone dependant.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 12 May 2011 21:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Cor-wrong (sort of).
There is a backport of DevState/Device_State for 1.4
https://issues.asterisk.org/view.php?id=15818
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
Wieling
Sent: 12 May 2011 20:01
, please. If you feel like you want to hurt yourself or
others, have yourself committed right away. I am serious. If you are
voluntary, you can leave when you want.
Thanks,
Steve Totaro
On Wed, May 11, 2011 at 12:13 PM, Andrew Thomas a...@datavox.co.uk
wrote:
Seems I have upset
Wow! How self-promoting was that post?
As for a simple 'that worked' post - as others have already pointed out
before you, it's not for self-gratification - it's to help anyone else
who has the same/similar problem. I used the list archives quite a lot
in my early days - and having the last post
Let's not get in to to pissing contest. I am not new to this list (jfyi
- I am also a dCAp). I do know who you are (and couldn't care less
anymore). I, also, have paying customers (but don't feel the need to
gloat about it in here). I am not pretending to know you - as I don't
know you on a
Snore...
Now move along please...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Totaro
Sent: 11 May 2011 14:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Seems I have upset the God that is Steve Totaro!
You want an example? OK - your last post. Has nothing to do with the
thread (or our 'discussion') but yet you chose to post it as yet another
self pat-on-the-back! I could produce a lot more - but you now bore me.
You know it must be so hard
Try getting rid of '/5001' (line 2 and 4) and try again!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh
katta
Sent: 10 May 2011 06:15
To: Asterisk Users Mailing List - Non-Commercial
Why do I get the feeling that this guy wants someone to write it for him
for free?
Especially seeing has how he has never posted what anyone who has tried
to help, have requested.
Maybe Mr. Katta needs to google for 'dcap'?
From:
...@lists.digium.com] On Behalf Of Russell
Brown
Sent: 13 April 2011 19:05
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [OT] Yealink Phones
Quoth Andrew Thomas:-
Have you seen the 'Action URL' bit yet? Makes everything almost
key-system like ;)
I saw it in the DSS key
Maybe I should have asked 'why do you want to put the status in to a
mySQL database'?
BTW - extensions.conf has mySQL functions built in - so no external
script is actually needed.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
...@lists.digium.com] On Behalf Of Ishfaq
Malik
Sent: 13 April 2011 10:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime SIP peer status
On Wed, 2011-04-13 at 10:28 +0100, Andrew Thomas wrote:
BTW - extensions.conf has mySQL functions built
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 13 April 2011 10:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime SIP peer status
On 04/13/2011 11:28 AM, Andrew Thomas wrote:
Maybe I should
Hi Russell,
Have you seen the 'Action URL' bit yet? Makes everything almost
key-system like ;)
BTW - one downfall of the Yealink is that it can't send different DND
commands to different accounts (it sends the one command to all
accounts). Not very useful if providers use different commands for
not externnotify
I am aware of extennotify, problem is, it runs script when someone
checks their voicemail, i need a script to run only when a voicemail is
left
On Mon, Apr 11, 2011 at 6:32 AM, Andrew Thomas a...@datavox.co.uk
wrote:
Not quite true. I use a PHP script to do my processing
Not quite true. I use a PHP script to do my processing (called from
voicemail.conf [externnotify = /usr/local/bin/vmnotify.php]).
The main three lines are:
$vm_context = $argv[1];
$extension = $argv[2];
$number_of_messages = $argv[3];
Self explanatory really.
-Original Message-
NT = Network Termination/Topology (or something like that) - used when
you want to be the network end.
TE = Terminating Equipemt - used when you want to be the consumer end (a
PBX or ISDN handset usually).
You probably want to be the TE - as you are running Asterisk PBX ;)
-Original
Just to respond to the IP range approach. My ISP recently changed my
external IP and now it appears that I am in New York (when I am actually
static in Manchester, England). I've also been in Birmingham,
Motherwell and Nottingham [UK] aswell! So, although banning certain
ranges may be a good
[18884732963@from-fax-machine:... - your call is hitting the
from-fax-machine context - yet your 'fax' exten is in the from-pstn-4
context. See the [2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c:
Fax detected, but no fax extension line.
When Asterisk detects an incoming fax tone - it tries to
The last Originate() option is ignored if using 'app'. It is only there
for 'exten'.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Originate tells all :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
[default]
exten = 777,1,Answer()
exten = 777,n,Record(/var/lib/asterisk/sounds/page:gsm)
exten = 777,n,Originate(Local/pb@dv-ip,exten,page-it,s,1)
exten = 777,n,Hangup()
exten = pb,1,Answer()
exten = pb,n,Playback(page)
[page-it]
exten = s,1,Set(page1=SIP/801SIP/802SIP/803) ; etc etc
exten =
If I was worried I'd record the 'page' first - and then play it back to
50 handsets at a time (using a loop).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish
patel
Sent: 14 March 2011 16:25
To:
...http://ofps.oreilly.com/titles/9780596517342/ch11.html if you're not
sure on Multicast (near the bottom).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven
Howes
Sent: 14 March 2011 16:30
To: Asterisk
Oops - from the very bottom of that page (no pun intended) : So far as
we can tell, Polycom sets do not support multicast. We certainly were
not able to find a way to use it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
...or for DAHDI channnels - the same thing in chan_dahdi.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bakko
Sent: 07 March 2011 19:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
in goto statement on h.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: Friday, March 04, 2011 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: 03 March 2011 17:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
On Thu, 3 Mar 2011, Andrew Thomas wrote:
Gentlemen, can we please not turn
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 04 March 2011 08:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
On Thursday 03 March 2011 08:42:42 Andrew Thomas wrote:
Does anybody know
Problem as follows:
[default]
exten = 777,1,Gosub(sub,1,1)
exten = 777,n,Hangup()
exten = h,1,NoOp(hung up in 'default' context)
[sub]
exten = 1,1,NoOp(in sub)
exten = 1,n,Playback(tt-monkeys)
exten = 1,n,Return()
exten = h,1,NoOp(hung up in 'sub' context)
This works fine if the caller listens
Nevermind - I've re-written my dialplan so that all subs are in one
context. Now I only need 1 more line of code.
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: 04 March 2011 11
Does anybody know of a way to test whether a mySQL connection invoked
from the dialplan is current or not?
For example:
extensions.conf
===
[context]
exten = _X.,1,MYSQL(Connect connid localhost user pass db)
exten = _X.,n,MYSQL(Query resultid ${connid} SELECT `something` FROM
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: 03 March 2011 14:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
Andrew Thomas wrote:
exten = _X.,n
- Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
Andrew Thomas wrote:
The wait is there as a test. This gives the 'tester' the option of
hanging up before the disconnect or not.
And the purpose for that would be to share available connections?
I've always
...@lists.digium.com] On Behalf Of Doug Lytle
Sent: 03 March 2011 16:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
Andrew Thomas wrote:
MYSQL_STATUS???
Is this documented anywhere (as I can't seem to find anything about
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
On Thu, 3 Mar 2011, Andrew Thomas wrote:
Does anybody know of a way to test whether a mySQL connection invoked
from the dialplan is current or not?
I've never been a fan of using database
Gentlemen, can we please not turn this in to an Asterisk and DB commands
bashing thread?
All I want is a simple answer to a simple question - not a debate on
using AGI/AMI or any other methods.
Thanks for your co-operation.
-Original Message-
From:
It seems like it is a v1.8 only function at present (unless a backport is
released).
From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
-
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
${HASH(SIP_CAUSE,channel-name)}
Asterisk 1.8 also comes with
It's all I use now.
I was luckily enough to be involved with quite a bit of the beta testing
in the UK - and, although there are a couple of 'nice-to-haves' missing,
they are excellent handsets. Polycom sound quality at Grandstream
prices ;)
I particularly like the 'use your own screen logo'
Changing
exten = start,n,While($[${MYVAR} != Some string])
to
exten = start,n,While($[${MYVAR} != Some string])
does the trick for me.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Try rrplacing MySQL(Query resultid ${conn_id} SELECT/ ramal/ FROM/
colaboradores/ WHERE/ ramal=${EXTEN});
With MySQL(Query resultid ${conn_id} SELECT `ramal` FROM
`colaboradores` WHERE `ramal`='${EXTEN}');
-Original Message-
From: asterisk-users-boun...@lists.digium.com
(DAHDI/15/${EXTEN})
but everytime i am getting the same DIALSTATUS
snip
-- Channel 0/1, span 1 got hangup request, cause 31
...
-- Auto fallthrough, channel 'SIP/2000-0002' status is 'CHANUNAVAIL'
/snip
Regards,
Robert
On 21.02.2011 12:13, Andrew Thomas wrote:
I'm curious as to what versions
I'm curious as to what versions of everything you are using. Reason
being this line -- DAHDI/i1/00256312261627-1 is proceeding passing
it to SIP/5000-.
It states DAHDI/i1/00256312261627-1... and I don't recall seeing that
before (my 2.4.0 says -- DAHDI/1-1 is proceeding passing
This sounds like a job for DISA.
http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA
helps.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius
Smith
Sent: 03 February 2011 19:46
To:
On Thu, Feb 10, 2011 at 6:08 AM, Andrew Thomas a...@datavox.co.uk
wrote:
This sounds like a job for DISA.
http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA
helps.
If OP is using Asterisk18, perhaps we should direct him to look here?
https://wiki.asterisk.org/wiki/display
Try changing 'hostname=127.0.0.1' to 'hostname=localhost' in the
cdr_mysql.conf. I seem to remember a problem I had when '127.0.0.1' and
'localhost' didn't marry up never did find out why.
If that doesn't work - try GRANT SELECT , INSERT ,UPDATE ON
`Asterisk`.`cdr` TO 'asteriskcdr'@'localhost';
@lists.digium.com
Subject: Re: [asterisk-users] MOH and parking
On 11-01-21 08:52 AM, Andrew Thomas wrote:
I know that the 'fix' has just been applied
(https://issues.asterisk.org/view.php?id=18262) - but why does it stop
the moh only to start it again? This, also, seems to cause a CDR
problem
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: 20 January 2011 18:44
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mailing list question
On 01/20/2011 11:16 AM, Andrew Thomas wrote:
Sorry Dannny - it didn't work :(
I can only
I know that the 'fix' has just been applied
(https://issues.asterisk.org/view.php?id=18262) - but why does it stop
the moh only to start it again? This, also, seems to cause a CDR
problem (see below).
-- Executing [7000@chambers:1] Park(SIP/2000-0008, ) in new
stack
== Parked
Hi,
I know you can access various sip variables via
'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of
the sip user - but what about variables?
I have a user that has setvar=123456 in their users.conf (sip.conf if
you prefer). I can read it with a 'sip show peer
-Commercial Discussion'
Subject: Re: [asterisk-users] Accessing a 'user' variable via. dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Thursday, January 20, 2011 11:26 AM
I always thought the last bit (after the /) is where the context in
sip.conf landed.
What about:
(sip.conf)
register = 119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
[52525252]
...
context = TRUNKin52
...
[59595959]
...
context = TRUNKin59
...
And
Hi,
Is the any kind of 'tag' that I can include at the end of my message to
make the list processing software ignore and dispose of my disclaimer?
In other words - something like disclaimer at the end of my message
would inform the list software to remove any lines after it.
My massive
list question
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: Thursday, January 20, 2011 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Mailing
On 01/20/2011 12:01 PM, Andrew Thomas wrote:
why not just subscribe with an account that doesn't do that like gmail
or yahoo ?
Hi,
Is the any kind of 'tag' that I can include at the end of my message
to make the list processing software ignore and dispose of my
disclaimer?
In other words
Sorry about this - testing this disclaimer problem :)
--
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mailing list question
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: Thursday, January 20, 2011 11:02 AM
January 2011 17:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mailing list question 2
On 20 Jan 2011, at 17:13, Andrew Thomas wrote:
Sorry about this - testing this disclaimer problem :)
I can give you a POP3 account on my server if it stops you
Top posting? Who cares? Get a life!
Now - can we get back to Asterisk et al?
Thanks!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark
Murawski
Sent: 18 January 2011 02:57
To:
Something that often gets forgotten is the on-site LAN infrastructure as well.
It could be a bad/faulty switch, rubbish cabling, induced interference etc.
etc. all at the customers premises.
Maybe a handset plugged directly in to the back of the router, before it hits
the LAN would tell you
Posting
On 11-01-18 04:22 AM, Andrew Thomas wrote:
Top posting? Who cares? Get a life!
Clearly not you, so why both even replying? At worst case it is just
redundant information for people, best case somebody reads the email
thread at starts bottom posting. I suggest taking a moment and
re
I also agree this is a pointless discussion because, clearly, nobody is
willing to budge, and it has nothing to do with Asterisk.
Amen :)
[oh no, a bottom post]
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The
at 03:18:49PM -, Andrew Thomas wrote:
Why do I top post? Simple. I read every message in the thread - and
if there are 10 messages (for example) in that thread - then why
should I have to read them all over again on the last one?
You mean: why should I have to read 10 messages worth
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,
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
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
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]
${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:
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:
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
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 -
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
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
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:
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
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
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
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 =
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:
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
[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
[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
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