On Sunday 25 Nov 2012, Dmitry wrote:
[snip]
3) Still know nothing about odbc support for queue_log
Happily using ODBC (PostgreSQL, but should be mostly DB-independent) for
queue_log here. The setup is a bit hairy, but can share if enough
people show interest.
Regards,
-- Raj
--
Raj Mathur
as semantic markup, so no Django
this side. Fine with Perl or Java, though.
Regards,
-- Raj
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On Friday 16 Nov 2012, martin f krafft wrote:
also sprach Raj Mathur (राज माथुर) r...@linux-delhi.org
[2012.11.16.1005 +0100]:
Warning: Not a fan of using whitespace as semantic markup, so no
Django this side. Fine with Perl or Java, though.
As long as we can agree on using a database
a
DAHDI group would be fine. Is there some way to achieve this?
Asterisk 1.8.8 on Debian Squeeze.
Regards,
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On Wednesday 17 Oct 2012, Tony Mountifield wrote:
In article 201210171813.45334.r...@linux-delhi.org,
Raj Mathur (à€°à€Ÿà€ à€®à€Ÿà€¥à¥ à€°) r...@linux-delhi.org wrote:
Our client has DAHDI groups with 4 PRIs in each group (one 4-port
interface per group), up to 6 groups per server. When
this installation files to be under my directory
myasterisksetup
can any one provide me step by step installation of asterisk in a
single directory
i m completely new to asterisk
so any help would be appreciated
sh ./configure --prefix /home/sudeep/myasterisksetup
Regards,
-- Raj
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able to hear you.
Regards,
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).
In short, my vote goes for case-sensitivity with a grace period for
switching over.
Regards,
-- Raj
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. Don't go by what the
provider tells you, figure out the cable connections yourself.
Regards,
-- Raj
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a different
cable signalling scheme.
Please don't top-post, it may get your posts ignored on the list. Use
bottom- or inline-posting, and trim your replies.
Regards,
-- Raj
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in getting
lines to work in Mumbai until we found the correct wire connections for
each of our providers (Reliance and BSNL AFAIR). Eventually Raj Kumar
(the wizard) did manage to figure it out and after that the lines worked
just fine.
Regards,
-- Raj
--
Raj Mathur || r
,
at least, the called party cannot hang up an incoming call on an analog
line -- the caller has to do it.
In any case, what happens if you execute HangUp() in your dialplan
instead of in AGI?
Regards,
-- Raj
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on a digital line. The problem is only with Analog
lines.
Regards,
-- Raj
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On Friday 14 Sep 2012, Patrick Lists wrote:
On 09/14/2012 05:26 AM, Raj Mathur (राज माथुर) wrote:
[snip]
Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL
Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card
(PCI-Express)
Your DAHDI and Asterisk versions are old so
number is 9811066XXX):
You appear to be suffering form
https://issues.asterisk.org/jira/browse/ASTERISK-19610
It is fixed in the just released v1.8.17.0-rc1.
Thanks, will test that on the weekend then.
Regards,
-- Raj
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(inclusive), the MySQL query would be:
select * from cdr
where calldate = '2012-09-01' and calldate '2012-09-08'
and duration 5;
Regards,
-- Raj
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to sort out the Asterisk packaged for Debian issue. Worst case,
will package it myself :(
Regards,
-- Raj
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support for these issues on
this card?
Regards,
-- Raj
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]: app_queue.c:1487 handle_statechange:
Device 'DAHDI/1' changed to state '0' (Unknown) but we don't care
because they're not a member of any queue.
Regards,
-- Raj
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handle_statechange: Device
'DAHDI/1' changed to state '0' (Unknown) but we don't care because they're not
a member of any queue.
Regards,
-- Raj
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It is the mind
the first one by loading the appropriate zones, loading the
Digium card driver with the correct opermode (which must be one of the
best-kept secrets on the Internet!) and enabling busydetect in Dahdi.
Stuck at the second one for now.
Regards,
-- Raj
--
Raj Mathur || r
is good.
We've faced stability issues with more than 500 simultaneous calls on a
single high-powered server, with no transcoding. However, that's
probably more a limitation of our own architecture and application than
a hard Asterisk limit.
Regards,
-- Raj
--
Raj Mathur
and we are happily running vanilla Asterisk Debian packages
with (even though I say so myself) some scripting to die for. Setup is
completely stable and is being used to generate some $15M of business
annually for the client.
Regards,
-- Raj
--
Raj Mathur || r
probably better to wrap this into a Perl or equivalent
script, but it should work on the shell too.
Regards,
-- Raj
--
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On Wednesday 22 Aug 2012, Roberto Piola wrote:
I would simply send the message with sendmail -v and then grep the
output for the error message
Er, that works too :) Much better solution (as long as you are root).
Regards,
-- Raj
--
Raj Mathur || r...@kandalaya.org
.
Of course, India has some retrograde VoIP rules, but blocking 5060/UDP
isn't an official requirement.
Regards,
-- Raj
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It is the mind that moves
,
-- Raj
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--
_
-- Bandwidth
(priorities) by 10. I presume a dialplan would work fine even
if the priorities aren't sequential, as long as they're increasing
monotonically.
Could someone confirm?
Having said that, I use n with abandon.
Regards,
-- Raj
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and find the place where my
skills and experience are the most valuable (to both myself and my
employer).
Thanks for all the pertinent and helpful advice and suggestions on the
list, Kevin, and all the best in your new assignment.
Regards,
-- Raj
--
Raj Mathur || r
there's a problem
As far as I know my client's blood pressure has remained stable over the
non-Redfone = Redfone transition :)
Regards,
-- Raj
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It is the mind
Answer() the SIP phone before dialling DAHDI?
Regards,
-- Raj
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character and
stop at first match.
If you have ODBC installed: your SQL back-end should be able to perform
this function for you.
See if the dialplan function FIELDNUM can be used.
There must be many more ways.
Regards,
-- Raj
--
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the IP stack on the Asterisk server to
choose the wrong source IP address for outbound packets.
I've usually seen this (wrong interface chosen for outbound) happen when
NAT is in the picture. However, the OP doesn't mention any NAT-related
configuration.
Regards,
-- Raj
--
Raj Mathur
been very positive,
leaving aside the advantage of having an E1 interface outside of the
computer (device and computer can be power cycled independently if
required, device can be easily interfaced to a different computer, etc.)
Regards,
-- Raj
--
Raj Mathur || r
of
communication servers, where any adds, moves, changes, failures or new
routes are automatically absorbed within the cloud with no additional
configuration.
Regards,
-- Raj
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to be a novelty than anything else.
Regards,
-- Raj
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On Tuesday 17 Apr 2012, cjwstudios wrote:
Looking for quotes on a very simple script that will require a pin
number before allowing a call to be placed. The pin number would be
recorded to their mysql CDR. Thank you.
Will the DISA application do what you need?
Regards,
-- Raj
--
Raj
anyway, what are the first two?
Regards,
-- Raj
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that can handle this problem generically, and
presumably you're available to alpha- and beta-test in any case :)
Regards,
-- Raj
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It is the mind that moves
know that the
other person is on a call..
i.e. When A calls B, while B is already on a call with C, Is there a
way to let A get a message that B is busy on another call.
If SIP, have a look at the busylevel and call-limit sip.conf parameters.
Regards,
-- Raj
--
Raj Mathur
to a separate SCSI
controller, which makes a lot of difference to performance.
Regards,
-- Raj
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within the Asterisk
process itself.
Regards,
-- Raj
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to
accomplish this?
Regards,
-- Raj
--
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Van Meggelen, and
Russell Bryant)
That's brilliant, and answers all my questions. Thanks!
Also thanks to Danny Nicholas for the initial push in the right
direction.
Regards,
-- Raj
On Wed, Feb 15, 2012 at 6:32 PM, Raj Mathur (राज माथुर)
r...@linux-delhi.org wrote:
A client is looking
IP issue, not just for connectionless
protocols. The same behaviour can be seen with TCP too. Unless you
mangle with iptables or something, all information about the received
interface has been stripped from the packet by the time it reaches the
IP layer.
/nitpick
Regards,
-- Raj
--
Raj
your default route is set to eth1, in the absence of custom routing
Linux will send the responses over eth1.
There are ways to overcome this for specific situations, but no general
method that I'm aware of (though I'm happy to be corrected).
Regards,
-- Raj
--
Raj Mathur
for the wrapped code.
Regards,
-- Raj
--
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in queue show... changes to BUSY, even though he's not taking
any calls.
We've tried both soft and hard phones as clients for this agent, but the
problem remains the same. Any help appreciated.
Regards,
-- Raj
--
Raj Mathur || r...@kandalaya.org || GPG:
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, which is
what we use for testing since it's light and fast.
Regards,
-- Raj
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,
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/dahdi/system.conf
Regards,
-- Raj
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from the government. Apart from that, as long as you continue using it
within your own organisation, any protocol is fine.
IANAL. TINLA.
Regards,
-- Raj
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in principle. This is why, e.g., Skype is
permitted: it doesn't connect to the Indian PSTN at any stage.
Once again, IANAL and TINLA. This is purely from my (mostly informed)
understanding of the current laws.
Regards,
-- Raj
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the called number in the CDR userfield in the 902000XXX
extension itself using something like:
exten = 902000111,n,Set(CDR(userfield)=${EXTEN})
There're probably many other (better) ways of achieving this too.
Regards,
-- Raj
--
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://etc.kandalaya.org/IVR/test.yml
First off, does this look like a useful tool? And if it does, what
features would you like to see in the finished product? (No promises
about speedy implementation, but I'll try!)
Regards,
-- Raj
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for discussion and testing, so please try to batch questions in one mail
itself!
Regards,
-- Raj
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cover the basic dual-licensing policy.
I am not a lawyer. This is not legal advice. In fact, as others have
also stated, it is highly recommended you consult with a lawyer who is
well-versed in licensing and specially FOSS licensing.
Regards,
-- Raj
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to GPLv2 as to be indistinguishable
from it. You cannot distribute a GPLv2 derived work under, e.g. a BSD
or Artistic licence.
Regards,
-- Raj
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It is the mind
command-line arguments. That is supposed
to turn off colours (though I've had mixed results with it). Also see
the -B and -W options (again, not always effective for some reason).
Regards,
-- Raj
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GPG: 78D4 FC67
at all (that's the only difference in
the production and test beds), but am still hopeful. An i686 Asterisk
installation on the production server should confirm or deny the theory
that the problem was purely for Asterisk x64. Will post once we know
more.
Regards,
-- Raj
--
Raj Mathur
On Sunday 30 Oct 2011, Raj Mathur (राज माथुर) wrote:
After looking further, the problem seems to be purely in playing
recorded messages over IAX2. Looking at the debug logs on the SIP
server (which is playing the recorded messages) shows that it stops
playing one of the messages at some point
://www.mail-archive.com/ilugd@lists.linux-delhi.org/msg28166.html
Regards,
-- Raj
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PsyTrance Chill: http://schizoid.in/ || It is the mind that moves
about the queue. Presumably
you can do the same through AGI too. In addition, I believe there are
some ready-made packages (both FOSS and proprietary) that will display
this information nicely formatted.
Regards,
-- Raj
--
Raj Mathurr...@kandalaya.org http://kandalaya.org
that my quick search didn't find.
Regards,
-- Raj
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On Saturday 29 Oct 2011, Raj Mathur (राज माथुर) wrote:
[snip]
Callers coming in from the PSTN (through the Dial server, over IAX2)
can also talk normally after an agent has picked up the call.
However, callers from the PSTN get the announcement and/or MOH
blanked out after a random period
a caller calls the Dial
server from the PSTN, which then passes the call to the SIP server over
IAX2.
Regards,
-- Raj
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PsyTrance Chill: http://schizoid.in
ONLY on incoming calls
from the PSTN. Any help appreciated.
Regards,
-- Raj
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call and you
will get a call.
On Mon, Mar 28, 2011 at 5:21 PM, Roger Burton West ro...@firedrake.org
wrote:
On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote:
Is there a better way of handling the post-hangup
processing?
Callfiles?
Thanks, call files worked beautifully. Takes
when the called party actually picks up the
phone? Or if she doesn't, the status of the call as it progresses?
Regards,
-- Raj
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PsyTrance Chill: http
Hi,
I'm trying to setup Asterisk so that:
1. I call a specific number that goes to a defined extension from my
phone (an external line).
2. Asterisk notes my phone number (the CLID) and hangs up without
picking up the call.
3. Asterisk initiates a call to my phone and prompts me for a passkey.
available) on multiport devices
where some of the ports are not populated. So if you have a 4-port PRI
device with just one PRI plugged in you would get that error regularly
on the remaining 3 PRI ports.
Regards,
-- Raj
--
Raj Mathurr...@kandalaya.org http://kandalaya.org
Hi,
Would you recommend some standalone SIP phones that work well with
Asterisk? Personal experience preferred.
Thanks,
-- Raj
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On Wednesday 09 Mar 2011, Raj Mathur wrote:
Would you recommend some standalone SIP phones that work well with
Asterisk? Personal experience preferred.
Thanks to all who replied.
Regards,
-- Raj
--
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GPG: 78D4 FC67
to a specific channel by typing in a ...# key sequence
during a spy session?
Regards,
-- Raj
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to the incoming lines.
4. Music-on-hold if all the operator extensions are busy.
Queues seem to handle this trivially, but I'm not sure if there are any
gotchas. Your advice would be appreciated.
Regards,
-- Raj
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