When I call to VoicemailMain it just sits.
; Retrieve Voice Mail
exten = 2500,1,Wait(2)
exten = 2500,2,VoicemailMain(s100)
exten = 2500,3,Macro(endcall)
1.4.3 latest SVN.
voicemail(100) works and the mwi systems works. I am not using ODBC or SQL.
Voice mail to email works ok.
I just cannot
I looked at the voicemail.c code and you must have the res.adsi module
loaded. I was not loading it.
Now it works.
Something to remember.
Thanks
--john
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l was busy then itplays a message that it was busy would I like to
continue or quit. Thanks--John
Hill--This mail was scanned by AntiVir
Milter.This product is licensed for non-commercial use.See www.antivir.de for details.
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Tzafrir,
Thanks for the reply.
This is a 2.6.13 kernel. Runs very well.
It really is not hurting anything memory usage is ok and it is responsive.
Just my old school resource attitude.
Shana Tova
--john
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I have a small test system -- 6 phones. It is a dual processor server. I
noticed that asterisk spawns 12 child processes. Can this be controlled? I
would think 2-4 would be plenty for this test site.
Thanks
--john
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Has anyone noticed that a ? Entered at the root CLI does not work any
longer?
Petty I know but I did use it.
--john
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Look at the dial app. I think it has several options.
Most custom 'TONES' are wav, acc, mp3 etc. files.
If you can set a different MOH class or perhaps a playback file in the dial
app that plays a file that is a 'RING TONE' that may work.
-John
-Original Message-
From: [EMAIL
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
John Hill
Sent: Thursday, September 22, 2005 9:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: Re: [Asterisk-Users] custom ring tone
Look at the dial app. I think it has several options.
Most
://printel.hr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
John Hill
Sent: Thursday, September 22, 2005 9:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: Re: [Asterisk-Users] custom ring tone
Look at the dial app
I was thinking of PSTN over FXO cards. When I see PSTN I think pots.
You mentioned BRI whould PRI do as well?
--john
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Fernando Herrera
Sent: Thursday, September 22, 2005 3:07 PM
To: [EMAIL
Richard Kashdan wrote:
On Mon Sep 12 10:32:47 CDT 2005 John Hill wrote:
I am having the identical problem. I use the CVSHEAD
Asterisk and do an
update every couple of weeks or so. I did one last week and
the caller
id quit working on my two lines that have x100p cards. I
John Hill wrote:
I deleted all modules and did a make install of the beta1
source using the
cvshead of zaptel and libpri.
Caller id then works fine?
Something has changed in the asterisk code that is not
seeing callerid from
of my x101p.
I was thinking about doing a fresh
I have 1.2.0 beta 1 running and it works fine. My x100p returns caller id
with no problems. When I test the CVSHEAD callerid fails with checksum and
len 0 errors.
I can run with the cvshead of zaptel and libpri with beta1 but only the
beta1 source works for caller id. Any source after beta1
CVSHEAD to work.
Thanks
John Hill
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I store my speed dial numbers in the astdb key speeddial with the number and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Cut(number=temp,,1)
exten = _*0XX,3,Goto(house-phones,${number},1)
The log
In article 200509091317.j89DGtY3019393 at commserver.noach.com,
John Hill jhill at noach.com wrote:
I store my speed dial numbers in the astdb key speeddial with the number
and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial
The 1.2 beta1 works fine. When I install the current cvshead it gives me
different errors:
I have seen checksum errors, Got event ring 18, etc. all give empty
callerid.
I have an x100p.
Thanks
--john
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I get this error message in my syslog.
I have searched the list but I can't seem to find a answer that solves the
problem.
chan_sip.c: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed
for '192.168.100.2'
The asterisk server is running dev head. The server is the 100.1 ip. I have
3
I use the CDR CVS file for logging my home phone system. Can I force write
data to a CDR Field from an extensions macro? Say if the callerid was empty
and I dumped the call to put data in the CDR to let me know that is what
happened.
Thanks
--John
___
Is there a way to combine these lines into one?
exten = s,2,GotoIf($[${CALLERIDNUM:0:3} = 800]?s|108)
exten = s,3,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|108)
exten = s,4,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|108)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 888]?s|108)
Thanks
--John
question
Try this:
exten = s,2,GotoIf($[${CALLERIDNUM:0:3} == 800] ||
$[${CALLERIDNUM:0:3} = 866] || $[${CALLERIDNUM:0:3} = 877] ||
$[${CALLERIDNUM:0:3} = 888]?s|108)
Diego Aguirre
FWD# 459696
- Original Message -
From: John Hill [EMAIL PROTECTED]
To: 'Asterisk Users
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, December 22, 2004 8:12 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Why use 'Answer'?
Why is it that newcomers always feel like inserting
-22 at 08:41, John Hill wrote:
Question:
Do you need to answer to detect a fax?
Yes. You need to answer the line so the calling fax will start sending
the fax tones and * can detect them.
-Seth
--
That's what I thought. My dial plan works but it looks a bit messy.
--John
to get it to work again.
Have I missed a configuration change somewhere?
Thanks
John Hill
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Hill
Sent: Tuesday, December 21, 2004 8:40 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] upgraded source now ata's ring but stop silence
oninbound calls
I
, stable ain't that stable right now ;)
John Hill wrote:
I was doing a daily make update for asterisk. On the 19th the new version
compiled fine. I installed it. All of my ata 186's can call out to pstn
etc.
All inbound calls, the phones ring but when you pickup, just silence both
; Zap Fax
;
exten = 8021,1,Dial(SIP/8021,20)
exten = 8021,2,Hangup
[incoming]
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,DigitTimeout(10)
exten = s,4,ResponseTimeout(20)
exten = s,5,Background(vm-extension)
exten = fax,1,Goto(8021,30)
I thinkg this should be goto(8021,1)
I'm losing call files in /var/spool/asterisk/outgoing because * isn't
able to detect the busy signal. The call file looks like this:
Channel: Zap/g2/3036701917
MaxRetries: 1000
RetryTime: 60
WaitTime: 45
Application: TxFAX
Data: filename.tiff|caller
Using the |caller parameter, TxFax injects
Trying to send a fax using a call file and txfax.
Phone dials the remote fax answers but * gives me:
Call failed to go through, reason 3
And hangs up.
Any help.
Thanks
--John
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I may be wrong but after looking around all I could find was an email about
w and p. It said w is to wait for a tone and p was for a pause. I can't find
anything to verify this.
--john
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Henry
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Walt Reed
Sent: Wednesday, November 03, 2004 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Reject a call if no callerID
On Wed, Nov 03,
Use an ATA the plug in any cordless phone.
Works fine.
--john
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Christopher TenHarmsel
Sent: Tuesday, November 02, 2004 10:50 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Wireless VOIP
I have 3 ata186's from Cisco. I have never used any other ATA but my guess
is they are all ok.
The 186's have 2 analog ports. I use one box for two different cordless
phones. The only thing I could not get to work was the *8. It must trap this
string. I had to change *8 to an extension number
I have a test for all tool free numbers.
This works but would using regex in one statement be more efficient?
exten = s,4,GotoIf($[${CALLERIDNUM:0:3} = 888]?s|108)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|108)
exten = s,6,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|108)
exten =
Use this url:
If you have a valid userid.
http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Vallee
Sent: Tuesday, September 21, 2004 4:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Her is the 7905-12 page
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Vallee
Sent: Tuesday, September 21, 2004 4:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
information. I also sent one out to the (x100p) pots, it worked the
first time. How can I monitor this process?
Thanks
--John Hill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, September 16, 2004 2:49 PM
To: [EMAIL
Make sure you have the bios updated to recognise the card. If the hardware
does not see it the OS wont either.
I just built Asterisk on an old single 450 PII Compaq Prolient 800 with an
x100p and 3 ata186's. It works just fine.
Hope this helps.
--john
- Original Message -
From: Rodolfo
I can part a call (dial #700 it is parked on 701)
but ifI dial 701 I am told it is not a valid extension?
I have include = parkedcalls in my local
extension context. I have Ttr on all extensions and the incoming pots
line.
It parks, plays MOH but I can't retrieve
it.
--john
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