Hi List,
Is there any way by which we can get the length of any recorded files into
seconds ?
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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Hy guys,
I have setup a chan_mobile trunk on AsteriskNow and Elastix also ,
using bluetooth libs and asterisk addon module ( asterisk 1.6) using
repository , is working fine but only one way , the client is hear me
but I can hear.I use Cambrige Silicon Canyon and a Samsung S3310 ,a
motherbord
Thanks for that, the table got created. Should be plain sailing from
here on :)
Ish
On Tue, 2011-06-07 at 10:43 +0530, Satish Barot wrote:
I use following for MySQL...
CREATE TABLE queue_log(
id int(11) NOT NULL auto_increment,
time datetime not null,
queuename VARCHAR(50),
agent
Hi All,
I tried to play a little bit with IPv6 to test our VoIP quality software
with IPv6 RTP streams.
I add bindaddr=:: to the general section of the sip.conf and netstat
shows that Asterisk is listing also on IPv6.
My Asterisk server is behind a IPv4 NAT and was working absolutely
Hello
I just read this article about a kid in England who built a box with a
3G SIM card:
www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html
When someone rings your intercom, the box will call your cellphone so
Hello
I just read this article about a kid in England who built a box with a
3G SIM card:
www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html
When someone rings your intercom, the box will call your
Hi,
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was configured dial plan below mention.
exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext
On 07/06/11 09:47, Gilles wrote:
Hello
I just read this article about a kid in England who built a box with a
3G SIM card:
www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html
When someone rings your intercom,
Realtime table queue_log@asterisk: Column time cannot be a datetime
Works fine once the time column is turned to char...
On Tue, 2011-06-07 at 10:43 +0530, Satish Barot wrote:
I use following for MySQL...
CREATE TABLE queue_log(
id int(11) NOT NULL auto_increment,
time datetime not null,
Hi all,
I try to figure out why I have empty :
sip show subscriptions
list in may asterisk 1.6.
When device is registering to asterisk I can see in log:
NOTICE[25603]: chan_sip.c:21518 handle_request_subscribe: Received SIP
subscribe for peer without mailbox: 1010
but
sip show subscriptions
How do you want to map callerid with your extensions? Do you have any DB
table for such a mapping?
[SATISH]
On Tue, Jun 7, 2011 at 2:29 PM, mahesh katta maheshka...@flexydial.comwrote:
Hi,
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was
Sir,
I have MYsql database in myserver.
On Tue, Jun 7, 2011 at 4:57 PM, Satish Barot satish4aster...@gmail.comwrote:
How do you want to map callerid with your extensions? Do you have any DB
table for such a mapping?
[SATISH]
On Tue, Jun 7, 2011 at 2:29 PM, mahesh katta
Hi,
Since raising this ticket about broken CDR data:
https://issues.asterisk.org/jira/browse/ASTERISK-17826
I have been researching how CDR records work in various circumstances.
CEL will do most things that people want, but that does not change
that CDR records are likely to persist into
On Tue, 7 Jun 2011, Gilles wrote:
Hello
I just read this article about a kid in England who built a box with a
3G SIM card:
www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html
When someone rings your intercom,
I mean, what do you want to see in callerid when Extensions
100-110,200-210,300-310 dial/receive the calls?
Like, 044578900 for 300, 044578901 for 101 and something like that.
Using,exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
and exten =
Hi,
what can I do for call tranfer settings and conference settings in asterisk
server with SIP extensions, internal and outbound. and also conference if
suppose I press the # and then press extenstion no. it will transfer.I tried
this in features.conf file .
--
Best Regards,
Mahesh Katta
On Tue, Jun 7, 2011 at 5:53 PM, Satish Barot satish4aster...@gmail.comwrote:
I mean, what do you want to see in callerid when Extensions
100-110,200-210,300-310 dial/receive the calls?
Like, 044578900 for 300, 044578901 for 101 and something like that.
Using,exten =
unsubscribe
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On Tue, 7 Jun 2011 13:06:23 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Why bother when you can buy off the shelf stuff to do it for you.
The trick is that this connector must work with existing interphones,
such as this one at home:
Hi Virendra,
Set DTMF option in the Makefile to 1 and then recompile/install the
app_konference module.
Thanks
Krishna
On Tue, Jun 7, 2011 at 1:31 AM, virendra bhati virbh...@gmail.com wrote:
Hi List,
I am trying to get DTMF into conference room. for conference I am using
Konference
Hello,
We have 1 server installed with centos and asterisk and the sip in
configured we can do the external and internal call without issue
Now I have installed the same centos and asterisk 1.4 in second computer in
the same network.
My question if there is any possibility to create a
Greetings list,
Has anyone compiled (or could point me at) a list of the minimum required
modules and conf files for a very basic 1.8 deployment?
We have lots of 1.4 boxes in production, and I'm currently setting up a pair
of 1.8 boxes to bounce calls coming in via IAX over IPv6 over to the
On Tue, 7 Jun 2011, salaheddine elharit wrote:
We have 1 server installed with centos and asterisk and the sip in
configured we can do the external and internal call without issue
Now I have installed the same centos and asterisk 1.4 in second computer
in the same network.
My question if
On 06/07/2011 08:04 PM, Chris Bagnall wrote:
Greetings list,
Has anyone compiled (or could point me at) a list of the minimum required
modules and conf files for a very basic 1.8 deployment?
Basic deployment is hard to specify, but in any case you can use the
following modules as a base to
Hello,
I tried using these options in my chan_dahdi.conf, but when i call my
asterisk box using the fxo line, i get this error:
[Jan 9 01:28:03] WARNING[12295]: sig_analog.c:2365 __analog_ss_thread:
DTMFCID timed out waiting for ring. Exiting simple switch
and no callerid appears in my
Call from 'sip' to extension '+1xxxyyy' rejected because extension
not found in context 'out'.
But
[out]
exten = s,1,NoOp( this is the extension: ${EXTEN})
exten = s,n,Answer()
exten = s,n(weasels),PlayBack(weasels-eaten-phonesys)
If I set s to _. it works.
Shouldn't s work here?
sean darcy wrote:
Shouldn't s work here?
S stands for start. Inbound calls via a provider, to your inbound
context would match. For you example, it'd have to be:
[out]
exten = +1xxxyyy,1,NoOp( this is the extension: ${EXTEN})
exten = +1xxxyyy,n,Answer()
exten =
On Tue, 07 Jun 2011 14:17:41 -0400
sean darcy seandar...@gmail.com wrote:
Call from 'sip' to extension '+1xxxyyy' rejected because
extension not found in context 'out'.
But
[out]
exten = s,1,NoOp( this is the extension: ${EXTEN})
exten = s,n,Answer()
exten =
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
sean darcy
Sent: Tuesday, June 07, 2011 2:18 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] why doesn't s accept incoming call
Call
On 6/7/11 1:59 AM, mahesh katta wrote:
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was configured dial plan below mention.
you have 9 digits on the starting number 8 digits
on the ending number. i'll assume it's a typo and
the ending number
On 11-06-07 02:31 AM, virendra bhati wrote:
Hi List,
Is there any way by which we can get the length of any recorded files into
seconds ?
$ sox foo.wav -e stat
[1] -
http://www.thegeekstuff.com/2009/05/sound-exchange-sox-15-examples-to-manipulate-audio-files/
--
Paul Belanger
Digium,
Hi ALL,
Is there any way i can reload chan_dahdi.conf without disconnecting active PRI
calls ?
I want to change pridialplan= option
-S
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Hi ALL,
Is there any way i can reload chan_dahdi.conf without disconnecting active PRI
calls ?
I want to change pridialplan= option
-S
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
satish patel
Sent: Tuesday, June 07, 2011 5:13 PM
To: asterisk-users
Subject: [asterisk-users] reload chan_dahdi.conf without
disconnect active calls
On 6/7/11 7:40 PM, Steve Edwards wrote:
Hi!
Agree on the IAX. Had never used it, I was a SIP-Only one man band. Not
got a VPS in a different location and configured IAX between them. Cool
thing is it has encryption built-in, in case you don't have both of the
machine locally.
As a really
Hi everyone,
What is wrong in below asterisk application? The output should be content of
field booth_status from table booths:
[extension-status]
exten = _X.,1,MYSQL(Connect connid 127.0.0.1 root password my-extensions)
exten = _X.,n,MYSQL(Query allow_call ${connid} SELECT extension_status
We have 2 PRI from ATT
And all is well but only few numbers having following issue. We are getting
hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and
this issue raised
[Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got
hangup request,
On Tue, 7 Jun 2011, Bruce B wrote:
What is wrong in below asterisk application? The output should be
content of field booth_status from table booths:
I don't see 'booth_status' or 'booths' anywhere below.
[extension-status]
exten = _X.,1,MYSQL(Connect connid 127.0.0.1 root password
hi:
there is no way to do that. why do you do that?
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 7 Jun 2011 21:12:37 +
hi:
make sure your pri is up and active.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: ca...@usawide.net
To: asterisk-users@lists.digium.com
Date: Mon, 6 Jun 2011 20:24:06 -0500
Subject: Re:
hi:
yes, make sure you also have a fxs to connect your fax if you want to receive
fax by fax Mac.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: asterisk_l...@earthshod.co.uk
To:
The problem is the OP never performs a Fetch of the data returned by the
Query...
From the VoIP-info page for Cmd MYSQL
MYSQL(Query resultid ${connid} query-string)
Executes standard MySQL query contained in query-string using established
connection identified by ${connid}. Result of query
Hello Guys,
After the Wiki was updated to the 3.5.X version, my username is no loger
available:
user: khratos
mail: j...@slackware-es.com
I had some documents on my personal space. Is there a way to recover the
account?
Regards,
--
Jose P. Espinal
http://www.eslackware.com
IRC:
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