Displaying 20 results from an estimated 138 matches for "corydon76".
2010 Mar 10
2
PGSQL application
Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons.
Atenciosamente,
Vin?cius Fontes
Gerente de Seguran?a da Informa??o
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brazil
+55 54 2104-7000
2009 Sep 15
3
dCAP Exam
Hi folks,
Is there anywhere I can possibly get a model of the exam itself, maybe
possible scenarios for the prac, etc?
To people who have done the exam....any helpful hints ?
Thanks,
2010 Mar 12
3
Time counting down and # detect
Hi all,
Here is the script i want to make
- Caller call to a number to record a message
- Asterisk answer and start recording message as following
+ User press * to start recording
+ Record is finished if:
+ User press #
+ OR message duration reach 60 second
+ Hangup
How do you counting down 60s, and how to detect # (i make a test using
Read() but it cant read #)
Thanks in advance
2009 Sep 23
4
Error When Using Postgresql Schema With Realtime Sip
I am using asterisk 1.6.1.6 and have been setting up a system to use a
Postgresql database as the realtime DB via the ODBC route. I have got
extensions and voicemail working but am having trouble with SIP
The problem seems to be with using a schema. If I put the table "sip" in
the schema "foo" then I add this entry to extconfig.conf
sippeers => odbc,psqldb,foo.sip
Restart
2010 Aug 02
5
mapping of disconnect reasons
Hi All,
Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is
2009 Sep 18
4
console color
Hoping someone can help me understand what is happening here;
we start asterisk as a service at boot (actually, with heartbeat) on
CentOS using the asterisk init script installed with "make config"
upon reboot of the server (when the asterisk service is first started by
heartbeat) we get color in the console when we connect to it using
asterisk -r
after the execution of
2009 Dec 17
6
Feature Request: GotoIfTimeWithOffset
Hi,
When I was testing an IVR, I realized I miss a function I would call
GotoIfTimeWithOffset.
Today, this IVR is using function AEL GotoIfTime in several places.
The problem is if it's 11pm at the moment I'm testing this IVR, I can't
nicely test the 9am or 2pm branch.
GotoIfTimeWithOffset would get 2 incoming arguments :
- the first is a time range (just like GotoIfTime),
- the
2003 Apr 06
1
Call completion/error codes and extensions.conf call flow
There was a conversation last night on the IRC channel between
myself, Corydon76, citats, and kram on the ability of a call process
to access the error (or success?) codes underlying a call. I'm
uncertain if anything came out of it, but I'll re-hash here to
solicit other comments.
My idea: I'd like to be able to get to error codes when a call passes
through so...
2010 Mar 04
9
30 mins GSM file
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
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2010 Nov 03
6
Migration from 1.2 to 1.8 in production
Hello Everyone,
We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version?
I would like if you suggest me which version would be good for production since
2010 Aug 28
4
Asterisk does not translate from wav to alaw
Hello list,
I have a file to be played in wav-format.
I thought Asterisk would automatically take the wav-file and translate
it to the codec used, but I see this :
[Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File
/var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any
format
[Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to
open
2009 Jul 20
0
No subject
...io, it might not be). What it probably
indicates is that the DTMF sent to your system is _incredibly_ short, and if
a
DTMF detector is employed, it's possible that the DTMF audio is simply too
short to be reliably detected.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
aste...
2009 Sep 07
1
Is not yet available ODBC support for queue_log in asterisk 1.6?
Hi list,
I hope someone could help me. I've started using Asterisk 1.6.0.14 to get
queue logs in real time with odbc (our databases are all PostgreSQL) but
it's not working. However, cdr odbc is working well. When asterisk starts
next message appears:
WARNING[4217] config.c: Realtime mapping for 'queue_log' found to engine
'odbc', but the engine is not available
My
2009 Sep 08
1
Function to query ASTDB families
Hi,
Asterisk database is made of <family><key> records such as:
fam key1 val1
fam key2 val2
...
fam key100 val100
I'm looking for the smartest way to iterate among different keys associated
to a given family.
One way to do this is to parse "database show fam" response.
Is there something smarter ?
Something like ${DBKEYS(fam)} which would evaluate to "key1
2009 Sep 16
1
res-crypto dependencies
I'm trying to enable res_crypto on a 1.4 installation, but menuconfig says
ssl is needed. I've installed openssl, openssl-devel, openssl-perl
but it's still not happy.
Anyone know what else is needed?
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2009 Sep 19
1
DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digiuim card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over SIP trunk from which
calls get routed to third server (C) (1.6.0.9) via IAX trunk.
SIP clients are connected to third server. A is the PSTN termination
server, B runs the menu and AGI and C is where SIP
2009 Sep 20
1
A in ACL of sip show peers.
Hello.
>> ubuntu*CLI> sip show peers
>> Name/username Host Dyn Nat ACL Port Status
>> voipprovider xxx.xxx.xxx.xxx A 5060 Unmonitored
I've ben trying to connect an asterisk server to a voip provider, and
I'm currently wondering what the 'A' in the ACL field of the 'sip show
peers' command might
2009 Sep 23
1
SFA - No channel cause 66
Hi,
after having tested SFA in august, I didn't use it for some times and
now I receive the subject error when calling through Skype channel.
Has anyone an idea on what can be the problem?
Thanks
--
Daniel
2009 Oct 05
1
Peculiar error message when using Q-SIG
I'm using QSIG between Asterisk and an NEC SV8300. Whenever I make a call
from the SV8300, I see:
[Oct 4 21:02:49] ERROR[5729]: chan_dahdi.c:12226 dahdi_pri_error: !! < Unknown IE 50 (cs5, len = 3)
I see an IE 50 in the Q.932 specification, so I don't understand why
this error is occuring.
2009 Oct 29
1
Astreicon presentations
Hi Folks,
Are all the astricon presentations up?
I'm especially after the one that tilghman did. I caught the tail end of
the prez when I decided to skip the session I was attending and go for
that one.
:)