2011/3/17 Eric Smith e...@fruitcom.com
Hi
I want to have some signal when a call is answered.
I can watch the asterisk debug or logs and see when a call is answered
of course but I want a sound notification.
I would try using Dial M or U options with which a macro or routine is run
when
2011/3/18 Frank Tarczynski ft...@mindspring.com
I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS
modules. I'm trying to set-up things to route analog fax calls from a FXO
port to an analog fax machine on a FXS port on the same card.
Outgoing faxes work just fine.
Probably this will help you...
http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288901
Check the section 'Controlling when to join and leave a queue'.
--AM
On Thu, Mar 17, 2011 at 9:15 PM, Dan Journo
d...@keshercommunications.comwrote:
Hi,
I'm trying to work out an issue with
Probably this will help you...
http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288901
Check the section 'Controlling when to join and leave a queue'.
Thanks. Thats perfect!
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
Hello,
Im tryng to setup DISA on my server.
Outlines comes via VoIP to my asterisk server.
When i dial from outside to my disa number it answers.
I dial the extension that i want to dial but Dial tone keeps up playing about
3-4 seconds more even i start to enter numbers..Then when timeout
On Tue, 15 Mar 2011 14:54:53 +0100, Gilles codecompl...@free.fr
wrote:
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
For those
On Tue, 15 Mar 2011 11:44:20 -0500, Danny Nicholas
da...@debsinc.com wrote:
Don't depend on the tutorials you read to be 100% accurate or up-to-date.
The default action on a failure in Asterisk is usually going to be an s
jump, either to s,1 or s+100. Personally, I would replace failed,1 with
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, March 18, 2011 10:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [1.4] Failed callfile doesn't jump
tofailedextension
On Fri, 18 Mar 2011 10:14:37 -0500, Danny Nicholas
da...@debsinc.com wrote:
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
exten = start,n,Goto(${EXTEN}-${REASON})
;not run
;exten = failed,1,NoOp(Call ended with ${REASON})
;not run
;exten = s,1,NoOp(Call ended
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, March 18, 2011 10:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [1.4] Failed callfile doesn't
jumptofailedextension
On Behalf Of Gilles
Unfortunately, it can only jump to h, and ${REASON} is empty.
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
Hi list!
We currently have a PRI gateway composed by a box with two Digium quad-span
PRI cards (a TE420 and a ).
One of the cards is filled with TELCO1, while the other has first two slots
filled with TELCO2, and 3rd slot with TELCO3.
I am currently having (timer ?) issues on TELCO3 (span 7)
Just a follow up with a bit more information
asterisk*CLI module show like timing
Module Description Use
Count
res_timing_pthread.so pthread Timing Interface 0
*res_timing_dahdi.soDAHDI Timing Interface
Is there a problem having 2 telcos on the same PRI card?
I think you go with one master timer as the Telco. Then the other spans
are secondary, tertiary, quaternary timers.
Adrian
--
_
-- Bandwidth and Colocation
On Fri, Mar 18, 2011 at 3:15 PM, Tiago Geada tiago.ge...@gmail.com wrote:
Just a follow up with a bit more information
asterisk*CLI module show like timing
Module Description Use
Count
res_timing_pthread.so pthread Timing
On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
adrian-li...@wombit.com wrote:
Is there a problem having 2 telcos on the same PRI card?
I think you go with one master timer as the Telco. Then the other spans are
secondary, tertiary, quaternary timers.
Adrian
Adrian
This only works when
Hi! I can try that tho. Where do I configure what timer to use??!
Thanks in advance.
On 18 March 2011 18:21, Andrew Latham lath...@gmail.com wrote:
On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
adrian-li...@wombit.com wrote:
Is there a problem having 2 telcos on the same PRI card?
OK I found it.
In /etc/dahdi/system.conf
I have for this span:
# Span 7: TE4/1/3 T4XXP (PCI) Card 1 Span 3 HDB3/CCS/CRC4
span=7,7,0,ccs,hdb3,crc4
# termtype: te
bchan=187-201,203-217
dchan=202
echocanceller=mg2,187-201,203-217
should I use span=7,*5*,0,ccs,hdb3,crc4 instead? (5 is the first
On 03/18/2011 01:23 PM, Tiago Geada wrote:
Hi! I can try that tho. Where do I configure what timer to use??!
If your telcos are not synchronizing their network clocks to each other,
you will not be able to solve this problem on a multi-port Digium T1/E1
card. Digium T1/E1 cards select a
Sorry to keep bugging, but after making changes to /etc/dahdi/system.conf,
do I need unload res_timing_dahdi.so and chan_dahdi.so; and load them, or
can I just reload them??
Thanks in advance
On 18 March 2011 18:26, Tiago Geada tiago.ge...@gmail.com wrote:
OK I found it.
In
Probably overkill, but Every time I make a change to dahdi, I do this
Service asterisk stop
Service dadhi restart
Service asterisk start
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Sent: Friday, March
Hi Kevin,
Thanks for your elaborated answer. I will try and set them on the same clock
and see if no problem occurs. If so, Different telco's clocks would be in
SYNC (I do doubt it).
This machine has no more PCI slots available and hardware is damn expensive.
Will have to look into it with my
On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
On 03/18/2011 05:43 PM, Gilles wrote:
On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being
On Fri, 18 Mar 2011 17:56:12 -0500, Anthony Messina
amess...@messinet.com wrote:
You need to define the 'failed' extension in your context to have the
${REASON} variable set (I've found).
exten = failed,1,NoOp(Failure reason is: ${REASON})
Thanks but for some reason, after calling out through a
On Sat, 19 Mar 2011, Gilles wrote:
Thanks but for some reason, after calling out through a call file,
Asterisk doesn't jump to it although the callee hangs up while
Asterisk is still playing:
Somehow, I'm guessing that 'failed' means that something failed while
processing the call file or
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If I can the fax machine from the outside the call is routed to the
expected voice extension as it is not a fax call:
Starting simple switch on 'DAHDI/4-1'
-- Executing [s@from-pstn-4:1
machine from a cell phone, for instance ?
Can you ear the fax machine answering ?
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If I can the fax machine from
machine from a cell phone, for instance ?
Can you ear the fax machine answering ?
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If I can the fax machine from
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