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
Hi all!
On outgoing call sometimes Asterisk use/give back the caller id sent back by
called number instead of number called by me. This is annoying and
misleading statistics if other side use some exotic number. For example I
have called number 12345678 and CDR include the number 333 as callerid
Hi,
On Wed, Feb 25, 2009 at 10:02 AM, Vieri rentor...@yahoo.com wrote:
Is Asterisk fully QSIG-compliant?
I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4.
Zaptel versions are 1.2.26 and 1.4.11.
That is a good question. I had the same dilemma here. Finally I am
using my OXE
Hi all!
I would like to make a service with recording sounds and playing back
to caller. I had wrote the script but it failed at Read statement with
file not found error. I have put some file test into script and this
is what happen on verbose level 9.
-- Executing [8...@default:8]
Thanks! Perfect! Such an easy solution shame on me.
Bye,
Zsolt
On Wed, Jan 28, 2009 at 11:10 AM, Steven J. Douglas stev...@moij.biz wrote:
In your Read command, leave out the .wav extension in the file name.
Regards,
Steve
Artifex Maximus wrote:
Hi all!
I would like to make a service
On Wed, Dec 17, 2008 at 2:16 PM, Olivier oza-4...@myamail.com wrote:
2008/12/17 Artifex Maximus artife...@gmail.com
On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote:
2008/12/17 Artifex Maximus artife...@gmail.com
If you don't expect to get more than 15 (or 12) calls
On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote:
2008/12/17 Artifex Maximus artife...@gmail.com
Is anyone using the $subject setup?
What I would like to do the following setup:
1. OXE is setup for receiving calls, handling Agents
2. Asterisk as external IVR on extension
Hi all!
Is anyone using the $subject setup?
What I would like to do the following setup:
1. OXE is setup for receiving calls, handling Agents
2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI
The incoming calling route:
1. OXE handles incoming calls, answer
2.
Hello!
On Tue, Nov 25, 2008 at 2:25 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 24 November 2008 06:19:44 pm Tilghman Lesher wrote:
On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote:
On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED]
wrote:
I've
Hi all!
I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
and tools but my calls aren't logged. I'd enabled mysql log and
noticed that asterisk send a 'DESC cdr' so connection is working
between asterisk and mysql and I am able to call other phones so
Asterisk is working as
On Tue, May 20, 2008 at 12:49 AM, Kevin Smith [EMAIL PROTECTED] wrote:
I almost hate to admit this...but I'm still running Asterisk 1.2 on
Fedora 4 :D
Same here (FC4 + Asterisk 1.2.10 + TDM + TE + iaxmodem + Hylafax).
Working flawless for ages. Old * because I had problems with upgrading
zaptel
On Wed, Mar 26, 2008 at 4:02 PM, Artifex Maximus [EMAIL PROTECTED] wrote:
Is it possible play background sounds while talking?
I would like to make an outgoing campaign with the possibility playing
sounds in background by command. But the extra is I would like to
choose which sound
Hello!
Is it possible play background sounds while talking?
I would like to make an outgoing campaign with the possibility playing
sounds in background by command. But the extra is I would like to
choose which sound to be played. In short operator calls a number,
talking to callee and sometimes
Hello!
Looks like list does not get this message. If it is duplicate sorry.
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign with this setup
on 4 PRI. The scripts for this:
Hello!
Thanks for your answer!
On Jan 19, 2008 12:49 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Jan 18, 2008 at 01:14:51PM +0100, Artifex Maximus wrote:
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign
Hello!
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign with this setup
on 4 PRI. The scripts for this:
caller php script write this to outgoung folder:
On Dec 7, 2007 11:07 AM, Philipp Kempgen [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
Is Asterisk capable forwarding D-Channel and making Asterisk box
totally transparent?
No.
Thanks Philipp. Bad news.
We need recording calls with using nice functions like time synchro
and tarification
Hello!
I am using Asterisk as transparent voice recorder for calls (isdn -
asterisk - pbx). Voice recording (therefore voice forwarding) is
working great but seems that Asterisk does not route/bridge/forward
D-Channel messages which means PBX cannot get time synchronization
answer from provider
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck. Extension 'failed'
does not know anything about ISDNCAUSE and 'h' extension is not called
at all. Any idea?
I
On Nov 12, 2007 3:22 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck
On Nov 12, 2007 2:04 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have
Hello,
On Nov 12, 2007 5:52 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
On Nov 12, 2007 2:04 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote
On 10/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 06:20:54PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote
Hello!
I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using
asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one
TE405P. When I upgrade to 1.2.20.1 the order of cards mess up and
therefore zaptel.conf is unusable and gives error. Why is it happen
and what do I need to
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote:
Hello!
I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using
asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one
TE405P. When I upgrade
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote:
Hello!
I have been trying upgrade zaptel from
Hello,
I have this setup:
Telco --PRI(g1,ext-incoming)-- Asterisk TE405P
--PRI(g2,int-incoming)-- Alcatel OXO
extensions.conf:
[ext-incoming]
exten = _X.,1,Noop
exten =
_X.,n,SetVar(CALLFILENAME=${TIMESTAMP:0:4}/${TIMESTAMP:4:2}/${TIMESTAMP:6:2}/${TIMESTAMP}-${UNIQUEID}-${EXTEN})
exten =
Try using set without ' or . I mean:
exten = s,n,Set(sep=;)
And next step try using CUT with and without ${..}.
exten = s,n,Noop(${CUT(v,sep,1)})
or
exten = s,n,Noop(${CUT(v,${sep},1)})
First parameter is using variable without surrounding ${..}.
bye,
a
On 11/30/06, Koen Van Impe [EMAIL
Hello,
Where should I find any updated AGI informations?
I am using wiki now but there are many outdated info (old pages) and
might some detail changed since it written.
For example I need to playback a sound file and there is a STREAM FILE
command. The wiki page notice a bug but I don't know
% 99.975586%
99.975586% 99.975586% 99.975586%
--- Results after 18 passes ---
Best: 99.975586 -- Worst: 99.975586 -- Average: 99.975586
bye,
Zsolt
On 9/24/06, Lee Howard [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
zttest is often on 99.975586% with final result:
--- Results after 67 passes
Hello,
I dunno what exactly happen but try gmake. Sometimes that helps on FBSD systems.
bye,
Zsolt
On 10/1/06, boon dash [EMAIL PROTECTED] wrote:
I am having issues with a new installation, configure completes
successfully, then once a make is run, the following errors show;
configure:
Hello,
For example I have this dialplan:
[context1]
exten = s,1,Noop
exten = s,n,Dial(...)
exten = s,n,Playback(${CONTEXT})
exten = s,n,Hangup
[context2]
include = context1
[context3]
include = context1
Then I make dial-out call files with context2, context3, etc. What is
the value of
IS: context4 HANGUP) in new stack
So looks like ${CONTEXT} is equal with originating context.
bye,
Zsolt
On 9/26/06, C F [EMAIL PROTECTED] wrote:
How about trying Noop(CONTEXT IS: ${CONTEXT}) before you post it? but
now that you have posted just try it and report back.
On 9/26/06, Artifex Maximus
/23/06, Ma Zhiyong [EMAIL PROTECTED] wrote:
zttest don't disturb your other active calls. when you have 10-20
calls maybe got a better score.
also cat /proc/interrupt and 'lspci -vb' to find any IRQ interrupt on
your system.
2006/9/22, Artifex Maximus [EMAIL PROTECTED]:
Hello,
On 9/21/06, Lee
Hello,
On 9/21/06, Lee Howard [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
Everything is fine when caller use ECM but when ECM isn't in use I
often got unusable incoming faxes (much often that it should be). But
when I switch back to fax machine that receive faxes perfectly (at
least
Hello,
My setup is PRI card, Asterisk, iaxmodem, hylafax or PRI card,
Asterisk, channel bank, fax machine. I'm using Fedora Core 4, iaxmodem
0.1.14, hylafax 4.3.0, asterisk 1.2.10.
Everything is fine when caller use ECM but when ECM isn't in use I
often got unusable incoming faxes (much often
Hello,
On 9/21/06, Doug Lytle [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
Hello,
Everything is fine when caller use ECM but when ECM isn't in use I
often got unusable incoming faxes (much often that it should be). But
when I switch back to fax machine that receive faxes perfectly
Hello,
On 9/21/06, Ma Zhiyong [EMAIL PROTECTED] wrote:
Are you use digium card?
digium pri card offen cause many problems, check zttest
Yes, it's a T405P.
Is zttest disturb the current calls or might works in parallel with
calls? Because it's very busy in worktime. And what should I look for
related to
resizing) in fax2ps/lp combo.
bye,
Zsolt
On 9/18/06, Artifex Maximus [EMAIL PROTECTED] wrote:
I'm using snapshot 20060915 for days and it's much better than before.
Still have some missing lines might related to bad quality line.
Thanks again!
bye,
Zsolt
On 9/13/06, Steve Underwood
I'm using snapshot 20060915 for days and it's much better than before.
Still have some missing lines might related to bad quality line.
Thanks again!
bye,
Zsolt
On 9/13/06, Steve Underwood [EMAIL PROTECTED] wrote:
Hi Bruce,
Looks like your typing is as bad as mine :-)
Try
On 9/15/06, Jerry Geis [EMAIL PROTECTED] wrote:
I have gone to http://soft-switch.org/downloads/spandsp/
looking to app_rxfax and I dont see it? Where is it?
http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/asterisk-1.2.x/app_rxfax.c
Hello,
I had received a lot of unreadable pages with rxfax. I've been doing
some search on net and found this:
http://threebit.net/mail-archive/asterisk-users/msg15708.html
It looks like rxfax/spandsp doesn't support ecm error correction. Bad
news for me. Is it still the case? app_rxfax.c dated
- Original Message -
From: Artifex Maximus [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 13, 2006 6:31 PM
Subject: [asterisk-users] rxfax, spandsp and lack of ecm
Hello,
I had received a lot of unreadable pages with rxfax. I've been doing
some search on net
Hello,
You don't need Agent. Use this instead:
member = SIP/148,1
member = SIP/143,2
Agent is for members defined in agents.conf.
bye,
Zsolt
On 9/13/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
I have queue with member defined as:
member = Agent/SIP/148,1
member = Agent/SIP/143,2
And when
Hello Steve,
On 9/13/06, Steve Underwood [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
Hello,
I had received a lot of unreadable pages with rxfax. I've been doing
some search on net and found this:
http://threebit.net/mail-archive/asterisk-users/msg15708.html
It looks like rxfax
, http://www.soft-switch.org/downloads/snapshots/snapdsp
http://www.soft-switch.org/download/snapshots/snapdsp,
On 9/13/06, *Artifex Maximus* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hello Steve,
On 9/13/06, Steve Underwood [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote
Hello,
announce = support-department
plays support-department.wav so playing support-department-recording.wav needs
announce = support-department-recording
bye,
Zsolt
On 9/12/06, gc [EMAIL PROTECTED] wrote:
I have this line in my queues.conf:
announce= support-department
and I have an
Hello,
Cups unfortunately don't support multi-page tiff printing.
http://www.cups.org/str.php?L1117
I have tried tiff2ps and tiff2pdf but both just embed original tiff
file and give the first page only.
Is there any solution for printing multi-page tiff easily? More likely
an alternative lp
use, works fine.
Yum tells me it comes from libtiff
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Artifex
Maximus
Sent: 08 September 2006 11:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Trouble with rxfax multi-page printing
Hi,
I have the following problem.
I need queue because of dynamic agents but I only want service as many
callers as available members are and want zero length waiting queue.
For example. I have two queues (q1,q2) and I use AddQueueMembers and
RemoveQueueMembers for maintain queue members.
Hello,
I found commands AddQueueMember and RemoveQueueMember so no need for
agent id and password. You just dial the extension and your extension
are in the game. Nice.
;Agent Login
exten = 450,1,Noop
exten = 450,n,AddQueueMember(q1)
exten = 450,n,AddQueueMember(q2)
exten = 450,n,Wait(1)
exten
Hello!
On 8/29/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Trying to add faxing to asterisk but get a compile error. Any ideas? Is
it broken for Asterisk 1.2.11 or was it me again :-)
app_rxfax.c:105: error: structure has no member named `column_resolution'
app_rxfax.c:105: error:
Hello!
Thanks for your answers!
Everything works fine now there was some problem at my provider. I
compiled and use rxfax successfully.
bye,
Zsolt
On 8/15/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
I didn't try that way, only tx fax in call file. But my experience is when u
r working with
Hello,
I'm looking for an agent managing dialplan/software/agi/whatever that
independent from asterisk queue management. I already tried this
http://www.voip-info.org/wiki/view/Agents+without+agent+channel
with no success but a lot of warning. I'm using asterisk 1.2.10 and
the dialplan above
Hello,
I have an incoming PRI (T405P card) and a TDM 2400P with 6 modules
connected to a channel bank with 50 pins cable. There are places for
16 phones, 4 faxes and 4 GSM gateway. Outgoing call are perfect but I
will need some debounce tuning because sometimes line ring back. But
it's another
May hotline function will help. I never been use with Asterisk just
with Welltech FXS device so it's just a hint.
artifex
On 4/21/06, Jerry Geis [EMAIL PROTECTED] wrote:
Anyone know how to set the wellgate unit so incoming calls
pass on directly to asterisk?
Right now incoming calls ring
I had conversation with Welltech support and I got this description (I
can't send attachment through the list):
The 380x has a routing table function. There are two default route
exist in the routing table, one is for IP incoming call another is for
FXO incoming call, the IP call will be routed
My asterisk system seems to have problems detecting hangups. I am
getting a LOT of voicemails with dialtone or silence.
I am using an external gateway (wellgate 3701a) and don't have zaptel
at all.
I think your 3701a don't understand hangup tone (as our 3802 did and
keep line busy after
Hello!
On 3/7/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Mar 7, 2006, at 7:02 AM, artifex maximus wrote:
I have the following setup:
Phone lines - traditional PBX - Welltech 3802
- VPN -
Asterisk - Linksys PAP2/Welltech ATA-151 - phone
There is 2 pieces of Welltech 3802 (2 port
Hi all!
I have the following setup:
Phone lines - traditional PBX - Welltech 3802
- VPN -
Asterisk - Linksys PAP2/Welltech ATA-151 - phone
There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2)
PBX extensions. Asterisk is a proxy here. Each device successfully
register itself. I
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