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
--
Raj Mathurr...@kandalaya.org http://kandalaya.org/
GPG: 78D4 FC67
Hi,
I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5
digits). ChanSpy is working fine for listening in to conversations
initiated by these channels, and I can use '*' to randomly switch
channels. However, is there any way in this scenario to be able to
switch ChanSpy
On Friday 25 Mar 2011, satish patel wrote:
*CLI*CLI dahdi restart
[Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi:
Ignoring any changes to 'userbase' (on reload) at line 23. [Mar 23
14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring
any changes to 'vmsecret'
Hi,
I'm using IAX2 between our SIP and PSTN servers, both running Asterisk
1.6.2. Users connect to the SIP server and dial; the SIP server
forwards the call to the PSTN server over IAX2, which then dials out
over the connected PRI. Since users need detailed call progress
feedback, the first
Hi,
Problem with Asterisk 1.6.2.9 on Debian Squeeze.
* Infrastructure
We have two servers, SIP and Dial.
The SIP server handles SIP clients; it also receives incoming PSTN calls
from the Dial server and makes outgoing PSTN calls on the Dial server.
The Dial server is connected to multiple
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
On Sunday 30 Oct 2011, Sammy Govind wrote:
Try turning on the Sip debug for the PSTN call as well as RTP debug.
Paste the output here.
There's no SIP involved until the call is picked up -- only PSTN and
IAX2. No RTP log available either.
The Dial server is connected to multiple 4-port
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
On Sunday 30 Oct 2011, Sammy Govind wrote:
hmmm so IAX channel is playing with you guys.
1- Cant you guys use SIP, does this happen with SIP trunk as well !?
2- Which version of asterisk are there on both servers.
3- See the output of the command core show file versions in your
both
On Sunday 30 Oct 2011, bilal ghayyad wrote:
In case I need to retreive the real time data (for example, how many
calls currently in the queue, and how many calls currently waiting
in the queue, how many agents currently are logged in ... etc).
How to get this?
Is it using the AGI? From
On Sunday 30 Oct 2011, bilal ghayyad wrote:
Actually I need to do a dash board for reporting, so I beleive the
only way is to use the AGI, correct? But where I can find documents
or link that can help me to do this?
About ur sentence:
some ready-made packages (both FOSS and proprietary)
On Monday 31 Oct 2011, Sebastian Arcus wrote:
Every time I start Asterisk (just by issuing /usr/sbin/asterisk),
the bash console text turns white. I'm using rxvt, so this makes
everything pretty much invisible. If I login into the Asterisk
console (asterisk -rvvv) - the text turns black
On Monday 31 Oct 2011, Alex Kauffmann wrote:
Sorry if i missed it, but is IAX2 trunked? IF so, perhaps you are
running out of bandwidth in your IAX2 trunk. The setting
'trunkmaxsize' defaults to 128000 bytes.
From the readme file:
...Once this limit is
; reached, calls may be dropped or
On Wednesday 09 Nov 2011, Kevin P. Fleming wrote:
[snip]
* The GPLv2 places no restrictions on what you can 'write', it only
places restrictions on your distribution of things that you write
that could be considered 'derivative works' of a GPLv2-covered work
(in this case, Asterisk). If you
On Wednesday 09 Nov 2011, Yaroslav Panych wrote:
I shall contact when(and if) decision will be made. But such decision
cannot be made basing only on this paragraph, because it does not
describes anything. There are no description of licensing procedure,
nor pricing, nor liability, rights or
Hi,
Having problems with a client trying to login to Asterisk 1.6.2 from
behind a DSL router. The account can be accessed perfectly from other
clients.
Would appreciate if you could look at the the attached log and see if
you spot any glaring issues. The user is very infrequently available
Hi,
Have started work on a Perl script to automatically generate Asterisk
IVR dialplans from a YAML configuration. It's pretty rudimentary right
now, but working for the couple of test cases I've thrown at it.
Current features:
- Built-in navigation (GoTop, GoToMenu). A GoUp is planned.
-
Please start a new thread for new conversations.
On Monday 12 Dec 2011, Albert wrote:
I have following problem. For statistical reasons I need to know what
was initiall number dialled by customer. I have 2 premium numbers,
for which customers are billed differently per minute. But in my CDR
On Tuesday 20 Dec 2011, Steve Edwards wrote:
On Mon, 19 Dec 2011, Nick Khamis wrote:
SIP in India is illegal.
What about IAX, Skype, VPN, etc?
The only thing that is not permitted is bridging Internet calls with the
Indian PSTN. In fact, that too is allowed if you have a VoIP licence
On Tuesday 20 Dec 2011, khalid touati wrote:
Thank you Raj,
so with VOIP license calls can go beyond our pbx to PSTN (india),
right, if so this what i needed to know to call Indian cellphone
from US (or other countries)
If your objective is to originate calls in the US (using whatever
On Thursday 22 Dec 2011, Olivier wrote:
Testing 1.8.8.0 (with Dahdi 2.5.0.2 and asterisk-gui 2.1.0-rc1), I'm
seeing this on my console:
WARNING[25363]: config.c:1208 process_text_line: Unknown directive
'#' at line 1 of /etc/asterisk/../dahdi/system.conf
This warning is repeated for every
Hi,
I have a queue with a number of (static) agents. Is there an easy way
for an agent to indicate that she is away from her seat, so that her
phone is not rung when a call comes in? And the converse, of course: be
able to notify Asterisk when she is back and ready to accept calls?
Regards,
On Saturday 07 Jan 2012, Tom Poe wrote:
Just installed asterisknow 1.6. I can access freepbx. I need to
test system on my LAN. Which softphone is best to use? I'm running
ubuntu on Dell optiplex G260 desktop at home. I'm hoping to setup
basic IP PBX for incoming/outgoing calls. No video.
Hi, we have a queue with some 20 agents. Agents are defined statically
in queues.conf (in an include, actually). One of the agents is showing
up as NOT AVAILABLE in queue show... when his client is disconnected
from Asterisk. However, the moment his system gets connected, his
status in
On Tuesday 07 Feb 2012, Josh wrote:
[snip]
Unfortunately, (IIRC) Asterisk does not reply to the same interface
packets are received from which limits the usefulness of multiple
interfaces.
What do you mean by that? If a request is received over eht1 are you
saying that Asterisk does not
On Monday 06 Feb 2012, John Cahill wrote:
logger -s checksetexternip.sh: External IP address
has changed, changing /etc/asterisk/sip_general_custom.conf grep -v
externip /etc/asterisk/sip_general_custom.conf
/etc/asterisk/sip_general_custom.conf.tmp echo externip=$EXTERNIP
On Tuesday 07 Feb 2012, Jakob Hirsch wrote:
Steve Edwards, 2012-02-06 01:43:
Unfortunately, (IIRC) Asterisk does not reply to the same interface
packets are received from which limits the usefulness of multiple
interfaces.
Right, that's what I also observed. We had to take special
Hi,
A client is looking for a way to have queue agents available over their
mobile or land-line phones. In other words, some queue members would be
local (over SIP channels) while others would only be reachable by
dialling their (mobile) phones over the PSTN. Is there some easy way to
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
On Tuesday 06 Mar 2012, Jason Parker wrote:
I don't know if I would call it a bug since the switch to install was
intentional, but I wouldn't say it's necessarily expected either. I
don't really have a strong opinion either way though. If anything, I
might be inclined to argue that 750 (or
On Tuesday 13 Mar 2012, Amit Patkar | Avhan Technologies Pvt Ltd wrote:
Thank for your views. Where as no one is ready to share real numbers.
I am looking at benchmarks so that I can plan for resources.
Since asterisk project is active for so many years, I was expecting
some published numbers.
On Wednesday 14 Mar 2012, NaJIm wrote:
When I make a call to an extension, which is on another call, the
called party (who is on call waiting) will get a BEEP sound. But
the caller gets the normal ringing tone. Is there any way to have a
different dialer tone for the Caller, which lets him
On Thursday 15 Mar 2012, Markus wrote:
With like 10 different ratesheets from 10 different providers, of
which many change their rates every few days, manually doing it in
Excel is too time consuming...
Is it possible to get samples? I'd be interested in looking into
developing a script that
On Thursday 22 Mar 2012, Danny Nicholas wrote:
Hi gang,
I've put 10.X on about 15 different VM's now, but I've
run into a buzzsaw on this one and my google-fu has failed me
Output of make
CC=cc CXX= LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent
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
On Friday 11 May 2012, Carlos Alvarez wrote:
On Fri, May 11, 2012 at 6:15 AM, eherr email.eherr9...@gmail.com
wrote:
What is the lowest end machine to run a production asterisk server.
Depends on a lot of variables. I've got some old 1.8GHz 1U servers
running hundreds of calls.
How
On Saturday 19 May 2012, Mikhail Lischuk wrote:
I've been playing around with clustering some
Asterisk servers for sake of fail-over and load balancing with DNS
round-robin, and came to one problem.
If I have, say, 2 servers, and
clients register either on 1 or 2, how can I route extensions
On Thursday 31 May 2012, David Klaverstyn wrote:
I would not recommend the redfone devices. I have had bad
experiences with them. Have a look at the Digium G100 or G200
devices. They look far superior.
Hmm, that's odd, since we've had an excellent experience with Redfones
(purchased
On Thursday 12 Jul 2012, Kevin P. Fleming wrote:
On 07/11/2012 11:36 PM, Jeff LaCoursiere wrote:
This does exhibit the problem though. Your OS stack assumes one of
those addresses - the first identified interface? - is the one that
all replies will appear to come from. So phones on the
On Tuesday 24 Jul 2012, Ishfaq Malik wrote:
It there a native asterisk dialplan function which will tell me the
position of a specific character in a given string?
eg if I wanted to find what position the '@' was at in ${SIPURI}
Worst case scenario: write a loop to iterate over each
On Friday 27 Jul 2012, Tim Nelson wrote:
Another mystery for the list, hopefully someone has ideas on a fix...
:)
I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS,
fractional 1-8). Outbound dialing works correctly, but while the
call is in progress, there is no 'ringing'
On Sunday 29 Jul 2012, Mike wrote:
what are folks using for PRI gateways these days? Obviously there's
lots of folks using TE410s and related cards, which work well, and I
know reasonably well.
However, anyone using anything standalone that stands out as being
particularly stellar?
One of
On Tuesday 31 Jul 2012, Kevin P. Fleming wrote:
[snip]
This is yet another incredibly exciting, career changing opportunity
in my life, and I can't wait to see what it will bring. I'll be
forever thankful for the opportunity that Digium and the Asterisk
community provided me to learn, grow
On Friday 03 Aug 2012, C. Savinovich wrote:
You don't use 'n's in your dialplan?, you number it yourself?
man, what if you have a 300 line dialplan and then you decide to
insert a new line in the middle?
If you ever used BASIC you'd remember the trick is to increment line
numbers
On Sunday 12 Aug 2012, Steve Edwards wrote:
On Sat, 11 Aug 2012, SamyGo wrote:
It takes a VPN or in near future WebRTC(in other words Knowledge)
to become one powerful guy. With these technologies you don't need
to care what your ISP or govt. is blocking.
Where there is will, there are
On Saturday 11 Aug 2012, Kannan wrote:
I am planning a multi-tenant VoIP services system with Asterisk,
using configuration tweaks. Having all the tenant configurations in
one configuration file is overwhelming. I would like to segment the
configuration files and include them in the main
On Tuesday 21 Aug 2012, Ruben Rögels wrote:
Hello,
no problem at all, I think this is the tricky part.
A smtp dialogue between your email client and a smtp server normally
looks like this:
user@box:~? netcat mx1.example.com
220 postfix ESMTP mx1.example.com
helo me.local
250
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
On Monday 27 Aug 2012, DHAVAL INDRODIYA wrote:
i would like to know if anyone has done or having idea regarding PRI
terminations in asterisk.
i have a requirement where i need to support 80 PRI in one machine i
have found a machine which have 10 PCI slots available
now i am thinking of
On Tuesday 28 Aug 2012, DHAVAL INDRODIYA wrote:
Thanks for everyone input on this, this was just mine thoughts to put
80 PRI line in that.but after reading inputs from everyone i think
there are some options to achieve it.
it means i need to put a gateway which convert my SIP calls to PRI
Hi,
Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai.
Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
When Asterisk executes HangUp() on an incoming call, the line remains
connected for the caller.
Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to
start
Hi,
Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai.
Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
When Asterisk executes HangUp() on an incoming call, the line remains
connected for the caller.
Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to
start
On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote:
Raj,
I am just confirming it happens here as well.
CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1.
Digium, Inc. Wildcard TDM410 4-port analog card (rev 11)
Loadzone = us
The problem started manifesting itself after I switched to 1.8.x
Hi,
Continuing with the saga of Digium vs MTNL Mumbai, looking for
suggestions on handling incoming Caller-ID issues. The card manages to
grab a couple of (random) digits of the incoming CID, but they're more
or less useless. Is there any way to fix this?
Asterisk 1.8.13, Dahdi 2.5.0.1 on
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
On Friday 14 Sep 2012, Richard Mudgett wrote:
Continuing with the saga of Digium vs MTNL Mumbai, looking for
suggestions on handling incoming Caller-ID issues. The card
manages to
grab a couple of (random) digits of the incoming CID, but they're
more
or less useless. Is there any way
On Friday 14 Sep 2012, RSCL Mumbai wrote:
I am trying to construct MySQL query(s) to get a list of calls which
lasted for less than 5 seconds between a given date range.
Any help is appreciated.
On the CDR database, to get all calls that lasted 5 seconds between
2012-09-01 and 2012-09-07
On Friday 14 Sep 2012, Richard Mudgett wrote:
Continuing with the saga of Digium vs MTNL Mumbai, looking for
suggestions on handling incoming Caller-ID issues. The card
manages to
grab a couple of (random) digits of the incoming CID, but they're
more
or less useless. Is there any way
On Monday 17 Sep 2012, Mehdi Rahimi wrote:
I need to use agi to handle some issue , after finishing agi i want
to hang up the channel , if i call from an extension there is no
problem but i want to be the same for PSTN (outside) caller , if
someone call asterisk show the hang up channel but
On Monday 17 Sep 2012, Mehdi Rahimi wrote:
Thank you for your reply
i did it in both ways (AGI and DIALPLAN) but not working.
so you mean it is because of telco ?
what about digital lines such as E1 ?
From my experience: the call gets disconnected if the called party
executes HangUp on a
On Sunday 23 Sep 2012, Ashish Agarwal wrote:
vi*CLI dahdi show status
Description Alarms IRQbpviol CRC
Fra Codi Options LBO
T4XXP (PCI) Card 0 Span 1OK 0 0 0
CCS HDB3 0 db (CSU)/0-133 feet (DSX-1)
T4XXP (PCI)
On Monday 24 Sep 2012, Ashish Agarwal wrote:
I have used 1 2 4 5 combination. Is that right?
I wouldn't know, since I'm not the wizard :) But basically we had to do
each provider's connections from scratch -- Airtel, BSNL, MTNL,
Reliance, Tata. And as far as I recall, each provider had a
On Monday 24 Sep 2012, Mitul Limbani wrote:
Signalling frm remote side is down.
Also just add crc4 in span 1,0,0 in dahdi/system.conf just like other
spans.
what is signalling= defined in your asterisk/chan_dahdi.conf ?
Not necessarily. I guess you remember the problems we had in
On Tuesday 02 Oct 2012, Mark Michelson wrote:
[snip]
Some of you might be eager to propose a configuration option to
decide which it should be. I'm sick of having hundreds of options
in Asterisk to slightly tweak the behavior one way or another. This
needs to go one way or the other, not be
On Thursday 04 Oct 2012, frangky robert wrote:
Here is my IP-PBX setupmy setup is : sips softphone - asterisk -
xorcom PSTN gateway - pstn line to telcoi'm using xlite for
windows when I make a phone call (sip - outgoing channel),I can hear
my own voice so clear. it's very annoying mewhen
On Monday 15 Oct 2012, sudeep melekar wrote:
hello,
i want to install asterisk 1.8 in a single directory
myasterisksetup i.e after asterisk installs it put some of it's
installation files in different directories
e.g /var/log/asterisk
/var/run/asterisk
and many more
i want all this
Hi,
Our client has DAHDI groups with 4 PRIs in each group (one 4-port
interface per group), up to 6 groups per server. When we dial, we can
specify the group to be used for dialling, and our dial plan
automatically distributes calls over multiple servers and multiple
groups within a server.
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
On Friday 16 Nov 2012, martin f krafft wrote:
also sprach Paul Belanger paul.belan...@polybeacon.com
[2012.11.08.2304 +0100]:
Either way, it sounds like you need to store your data some place
and start building it out.
To recap: given that Asterisk RealTime doesn't really provide
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
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
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