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for this?
Thanks in advance!!!
Thanks,
Max Alex
Voip Developer
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The way that I understood this to work was that e164.org lists all toll-free
numbers to make it free to call those kinds of numbers (instead of using one
of your own trunks). Since ENUM can provide priorities, if I own a toll-free
and enter it into the system, the route that I specify will be
On Mon, Jul 20, 2009 at 05:58:51PM -0400, Brian McEntire wrote:
Thanks for the reply Alex. I'm not too scared of the soldering iron (I
own one, but my work with it isn't pretty ;-)
But can you confirm, are you just using the small power header on the
board to supply power to the pci card? I
the externip, localnet,
and nat settings. Thanks,
You just set up some SIP peers in sip.conf, and some simple dial plan
routes in extensions.conf. Calls in, calls out. B2BUA baked right into
the crust.
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BTW, if you need a generic, scalable, high-volume B2BUA, it is not a
best practice to use Asterisk for that purpose.
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Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
Andrew Pogrebennyk wrote:
Alex,
Alex Balashov wrote:
BTW, if you need a generic, scalable, high-volume B2BUA, it is not a
best practice to use Asterisk for that purpose.
Thanks for your advice. Well, I'm not considering any commercial
products at the moment. I'm also checking
Asterisk Development Team escribio':
The Asterisk Development Team is pleased to announce the release of Asterisk
1.4.26. Asterisk 1.4.26 is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
I don't see 1.4.26 anywhere in the directory, just
on this box though - I have some thing
similiar at another site, but a bigger box.
Alex
Soekris' description for the net5501-70 says, in part, it has support
for one or two low-power standard PCI board
I see on my Digium card that it requires a molex connector supplying
voltage
is to have a piece of middleware that
keeps track of when agents are connected to a caller via listening to
the Asterisk manager interface, or by reading the queue log. It can
then be queried via FastAGI.
Really, there should be a dialplan app for this. Feature request?
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:
Someone know how can I check for available members on a queue
Before
I queue the call, so I can do something else with it? Note that is
not
the case for joinempty
On Sat, 18 Jul 2009, Alex Balashov wrote:
It's going to take some sort of hack, since there appears to be no
dialplan
.
Where did you hear of this limitation?
Alex Balashov
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logic you choose. No
shoehorning--just write it yourself.
-- Alex
[1] Yes, FastAGI. Not local AGI. And most especially not in PHP;
contrary to a lot of the info out there, PHP could not possibly
be a less suitable language in which to write AGI scripts. I
don't know who comes
/asterisk-users
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Understood--thanks Trevor. I had wondered if the need to pay per
channel might somehow amortize the LD balance. Appreciate your
clarification.
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On Jul 17, 2009, at 5:14 AM, Trevor Hammonds tre...@concipient.net
wrote:
On Thursday, July 16, 2009, Alex Balashov
IMHO, anonymous calls should never, ever be accepted for a variety of
reasons. It is naive.
Just because it is convenient does not mean it should be done.
Trusted calls between indeterminate parties can be arranged through
peering federations, clearinghouses, etc. -- whatever VoIP peering
You're welcome.
What's TAPI?
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On Jul 17, 2009, at 5:38 AM, jonas kellens jonas.kell...@telenet.be
wrote:
Thanks Alex for your explanation.
Does this NAT-mapping means that TAPI would also be possible ??
Jonas.
On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov
a database for the
most appropriate agent and then call that agent's extension.
Julian
2009/7/17 Matt Florell astma...@gmail.com:
On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote:
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
We are trying to implement skill based
- with a little more effort we could probably remove the
need for the queue. But why would I do that if I can use the queue for
the bits I want ;)
Julian
2009/7/17 Alex Balashov abalas...@evaristesys.com:
The simplicity of this approach is elegant, but in that case, why use a
queue? Why not just
that is billed
long-distance relative to the destination (but still intra-LATA)? Or do
you pay normal LD rates on top of all that in the intra-LATA LD scenario?
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on this issue a little bit.
[1] http://www.openser.org/pipermail/users/2009-July/006873.html
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On Jul 15, 2009, at 8:20 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Alex, no need for personal attacks.
Zeeshan
On Wed, Jul 15, 2009 at 2:23 AM, Alex Balashov abalas...@evaristesys.com
wrote
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Balashov
Sent: Wednesday, July 15, 2009 1:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to ask questions the smart way
Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's
option is enabled in sip.conf fulfill this function.
Hope that helps,
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Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
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from
and change it. Go into
asterisk-x.y.zz.?/include/asterisk/cdr.h:
#define AST_MAX_ACCOUNT_CODE20
Then recompile the entire source tree.
If you didn't install from source, you're out of luck.
If you don't want to modify the source, you're out of luck.
-- Alex
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On Mon, Jul 13, 2009 at 11:50:00AM -0500, Brent Davidson wrote:
Alex Samad wrote:
Hi
I have setup forwarding - xfering - where you press # and then the
extension. I add t to the dial cmd.
My problem is that when you call something like internet banking they
want #, but when
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implementation, just the webphone software, which is usually
only a file or two.
On Sun, Jul 12, 2009 at 11:37 PM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
This is a tenuous concept to begin with. You can't expect good
implementations to be free
asterisk to listen to flash either
Alex
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Alex Samad wrote:
Hi
I have setup forwarding - xfering - where you press # and then the
extension. I add t to the dial cmd.
No, that's simply the order of evaluation. If the caller is inside an
Asterisk application that listens for #, it is going to be intercepted
and preempted instead
Nick Hill escribio':
Hello Sasa
The page you point to doesn't talk about USB connectivity for chan_mobile. It
does talk about bluetooth connectivity, which can be achieved by way of a USB
bluetooth dongle, but that is not the same thing.
I am talking about using standard interfaces
This is not currently possible. Work in progress.
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On Jul 8, 2009, at 1:31 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello All,
can anybody tell me how can i integrate asterisk and skype users
so that skype users can dial my asterisk number or dial
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Define available.
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On Jul 4, 2009, at 6:24 AM, Michael as...@nettrust.co.nz wrote:
Is there a way to test if a peer is available PRIOR to doing a dial
attempt?
Michael
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On Tue, Jun 30, 2009 at 04:57:29PM +0400, M C wrote:
Hello,
i have just installed asterisk 1.6.0.10 on debian 5.0 like:
./configure;make menuselect; make;make install
any reason to not use the deb files ?
There are no erorrs, but folder /usr/lib/asterisk/modules is empty.
What am i
It does this by default unless you have allowguest set to yes, and/or
any insecure parameter options on any individual peers.
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On Jun 29, 2009, at 10:33 AM, michel freiha mich...@gmail.com wrote:
Hi all,
i would like to ask please about how to force asterisk to
Without getting into a lot of detail, this will not work. Period.
You just can't do reliable modem passthrough with VoIP in most cases,
some clever proprietary hacks notwithstanding.
To the extent it is possible, nobody is going to send you the
procedure.. This list is for specific
/gemeinschaft/trunk/opt/gemeinschaft/sbin/gs-sip-ua-config-responder/gs-sip-ua-config-responder
Best regards,
Loïc Didelot.
On Thu, 2009-06-18 at 21:25 +1000, Alex Samad wrote:
Hi
I am trying to setup asterisk to do a mass deploy of some snom phones. I
can't find where i configure
Xavier Cardil wrote:
Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster
debugging ?
I'm not sure what you mean by for a faster debugging. As for sniffing
the traffic, tcpdump works well.
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Tel
voltage, any one in oz using this card ?
Thanks
Alex
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)
[1083340.340492] Port 2: Installed -- AUTO FXS/DPO
uses complex impedance (220+820Ohm resistors with a 120nF capacitor)
whereas the US uses a straight resistor.
Did yo buy from the us or local ?
Alex
Alex Samad wrote:
Hi
I was wondering if any one has used these cards, I am looking
On Tue, Jun 23, 2009 at 11:32:08AM -0500, Shaun Ruffell wrote:
Alex Samad wrote:
I am having some problem forcing my tdm410 to alaw over ulaw...
You will want to set the alawoverride module parameter to 1. i.e.
'modprobe wctdm24xxp alawoverride=1' or alternatively, edit your
/etc
standard kernel. I am running on a soekris
board - amd geode cpu.
Is recompiling the kernel to the 1000Hz going to be beneficial to me,
the box is primarily used for firewall router / voip (Asterisk)
Alex
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On Wed, Jun 24, 2009 at 01:02:08AM +0300, Tzafrir Cohen wrote:
On Wed, Jun 24, 2009 at 07:10:15AM +1000, Alex Samad wrote:
Hi
I was reading this article on installing asterisk 1.6 + debian
http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr
show channel 1
Default law: ulaw
even when i have a call going it is still ulaw
Thanks
Alex
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Hi
I am seeing this in my syslog
[235900.797660] dahdi: Registered tone zone 0 (United States / North
America)
I am in Australia so I would want to set them to AUS zone
I have got this though
options wctdm24xxp opermode=AUSTRALIA
thanks
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On Fri, Jun 19, 2009 at 11:08:49AM +0300, Tzafrir Cohen wrote:
On Fri, Jun 19, 2009 at 06:04:17PM +1000, Alex Samad wrote:
Hi
I am seeing this in my syslog
[235900.797660] dahdi: Registered tone zone 0 (United States / North
America)
I am in Australia so I would want to set
Hi
I am trying to setup asterisk to do a mass deploy of some snom phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where to set the notify reply.
I was hoping to not have to use dhcp options
alex
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I thought TFTP (and therefore, DHCP option 66) is the only
autoprovisioning method Asterisk supports?
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On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
Hi
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where
On Thu, Jun 18, 2009 at 07:34:38AM -0400, Alex Balashov wrote:
I thought TFTP (and therefore, DHCP option 66) is the only
autoprovisioning method Asterisk supports?
seems like the documentation from snom for V7, includes the pnp method
as well. it sends a subscribe to a multicast address
of *.
The doco seemed to suggest after I press flash I should heard a dial
tone ! which i don't
Alex
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad
Sent: Wednesday, June 17, 2009 9:41 PM
To: asterisk
On Thu, Jun 18, 2009 at 11:57:20PM +0200, Philipp Kempgen wrote:
Alex Samad schrieb:
seems like the documentation from snom for V7, includes the pnp method
as well. it sends a subscribe to a multicast address (224.0.1.75) and the
listener is
meant to respond with a notify which has
On Thu, Jun 18, 2009 at 02:21:47PM +0200, Philipp Kempgen wrote:
On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where
Hi
I have 2 digium cards (tdm410) with combination of fxs + multiple fxo
ports.
I have had a quick look at sangoma B series cards. I was wondering if
there is a card out there with
hardware echo canceller
say max 4 ports (mix of fxs/fxo)
g729 encoding onboard
Alex
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On Thu, Jun 18, 2009 at 11:24:40PM -0500, Karl Fife wrote:
After a kernel update (but before rebooting) Is there a way to recompile
Zap/Dahdi against the new kernel?
My objective is to eliminate the additional downtime that occurs while
recompiling/installing zap/dahdi after booting into
On Wed, Jun 17, 2009 at 09:34:55AM -0400, Matt Florell wrote:
On 6/17/09, Gordon Henderson gordon+aster...@drogon.net wrote:
On Wed, 17 Jun 2009, Steve Totaro wrote:
Hi,
[snip]
Gordon
The TC400B is up to 120 channels of G729a now:
hit flash
101 - dahdi/1 a uniden pots phone
Thanks
Alex
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On Sun, Jun 14, 2009 at 03:10:03PM +1000, Alex Samad wrote:
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
[snip]
The scripts for downloading the post-build firmware were moved to the
separate dahdi-firmware package (sadly it has not made it into the
archive yet
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
[snip]
Although I think I did see it download the firmware
The scripts for downloading the post-build firmware were moved to the
separate dahdi-firmware package (sadly it has not made it into the
archive yet). As the firmware
Hi
it seems like chan_dahdi.so is missing in debian asterisk 1.4.21
so I have upgraded to 1.6 and no I can load chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
[Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error
loading module 'chan_dahdi.so':
On Tue, Jun 16, 2009 at 07:03:06AM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
it seems like chan_dahdi.so is missing in debian asterisk 1.4.21
so I have upgraded to 1.6 and no I can load chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
[Jun 16 21:22:30] WARNING
On Tue, Jun 16, 2009 at 02:35:08PM +0300, Tzafrir Cohen wrote:
On Tue, Jun 16, 2009 at 09:04:37PM +1000, Alex Samad wrote:
[snip]
dahdi_genconf generates configuration. It is a tool intended to help you
and not a required step.
It defaults to using mg2[1]. You can tell it to use
/sys/module/wctdm24xxp/parameters/vpmsupport
the thing that is interesting is that dahdi_cfg -vv shows me mg2 (I have
mg2 in the /etc/dahdi/system.cfg).
Should I just leave echocanceller out fo system.conf ?
and dadhi show channel 1 still shows echo cancellation off ?
alex
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On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
some question I have now is when i do a dahdi show channel 1 i get these
interesting results
Echo Cancellation:
128 taps
currently OFF
I have a hardware echo can and I have asked
On Wed, Jun 17, 2009 at 07:16:53AM +1000, Alex Samad wrote:
On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
[snip]
Default law: ulaw
I have a alaw:1-4 in the conf file, but it doesn't seem to take
That is not valid syntax for /etc/dahdi
On Wed, Jun 17, 2009 at 01:23:19AM +0300, Tzafrir Cohen wrote:
On Wed, Jun 17, 2009 at 07:08:10AM +1000, Alex Samad wrote:
On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote:
Tzafrir Cohen wrote:
Duh. Ignore this. You asked about the hardware EC. The hardware EC can
proceeding to provide
services or consulting assistance.
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Hi
I would like the option to set the codec used on a call by call basis.
I have a tdm410 2fxs + 1fxo.
when I make calls to my vsp, they go through as ulaw, I am guessing
because I have allowed if for the vsp (g729, alaw and ulaw).
I would prefer to use g729 from the fxs to the vsp but I would
to engage digium to providing a fix for this ?
Alex
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On Mon, Jun 15, 2009 at 08:19:33PM -0500, Lyle Giese wrote:
Alex Samad wrote:
Hi
[snip]
as you can see with the interrupts the wctdm24xxp0 is above eth0 (local
lan) and eth3 (my adsl)
eth1 is wireless and not heavily used
So any one had this problems, any other possible
On Sat, Jun 13, 2009 at 11:58:40AM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 09:51:24AM +1000, Alex Samad wrote:
To get this to work can i simply
apt-get source dahdi-linux
modify debian/patches/series
to comment out no_firmware_download
then
dpkg-buildpackage
On Sat, Jun 13, 2009 at 01:10:33PM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 07:51:54PM +1000, Alex Samad wrote:
On Sat, Jun 13, 2009 at 11:58:40AM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 09:51:24AM +1000, Alex Samad wrote:
To get this to work can i simply
On Sat, Jun 13, 2009 at 05:10:34PM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 09:46:23PM +1000, Alex Samad wrote:
On Sat, Jun 13, 2009 at 01:10:33PM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 07:51:54PM +1000, Alex Samad wrote:
On Sat, Jun 13, 2009 at 11:58:40AM +0300
not tar
balls easier to maintain.
My only issue has been that because of debian rules the firmware for the
hw echo cancellor isn't provided
Alex
\erik
Date: Sat, 13 Jun 2009 09:51:24 +1000
From: Alex Samad a...@samad.com.au
Subject: Re: [asterisk-users] Help building dahdi for debian
On Sun, Jun 14, 2009 at 08:20:18AM +1000, Alex Samad wrote:
[snip]
It merely packages (most of the) the source tarball in the dahdi-source
binary package, which is later built with dahdi-linux.
that makes sense, I had a time constraint, will look at it next weekend,
when I have some
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
On Sun, Jun 14, 2009 at 12:23:41PM +1000, Alex Samad wrote:
On Sun, Jun 14, 2009 at 08:20:18AM +1000, Alex Samad wrote:
[snip]
It merely packages (most of the) the source tarball in the dahdi-source
binary
is the best way forward to recompile with hardware echo canceller
support.
Thanks
Alex
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On Fri, Jun 12, 2009 at 05:40:16PM +0300, Tzafrir Cohen wrote:
On Fri, Jun 12, 2009 at 11:58:51PM +1000, Alex Samad wrote:
Hi
I am in the process of installing a new box and using dahdi. I have a
tdm410 + hardware echo canceller.
I have just read in the read me for dadhi
On Sat, Jun 13, 2009 at 01:40:48AM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 06:57:11AM +1000, Alex Samad wrote:
any chance of getting digium to host a digium debian repo (sort of how
virtulbox doit), that way they could have a fully build package ?
Or resolve the issues
On Thu, Jun 11, 2009 at 09:02:37AM +0100, Gordon Henderson wrote:
On Wed, 10 Jun 2009, Alex Samad wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card - basically the box
On Thu, Jun 11, 2009 at 11:14:47AM -0700, Ira wrote:
At 02:01 PM 6/10/2009, you wrote:
http://www.cyberguys.com/product-search/?keyword=molex
doesn't look like it, really need a 90 degree plug and I am in OZ not
usa so postage is going to kill me
I'd buy a standard one, pull the pins,
the
card - seems like people have done this with a TDM400, unfortunately the
410 is longer
Alex
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On Wed, Jun 10, 2009 at 08:44:22AM -0400, David Backeberg wrote:
On Wed, Jun 10, 2009 at 7:17 AM, Alex Samada...@samad.com.au wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card
On Wed, Jun 10, 2009 at 05:49:22PM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
I have read up about a PWR2400b and it seems to use 2wire pin, I am
guessing to connect to P8 just behind the molex connector on the tdm410.
can any one here confirm this, or have any info
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Alex Balashov
Evariste Systems
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Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
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Hello,
How can I do bitwise AND operations on a variable? I want to check the bits
set in the HANGUPCAUSE, but can't find a way to do it.
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Alex Hermann
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Give this a whirl:
${MATH(HANGUPCAUSEbit)}
Got it here:
https://issues.asterisk.org/view.php?id=9891
Alex Hermann wrote:
Hello,
How can I do bitwise AND operations on a variable? I want to check the bits
set in the HANGUPCAUSE, but can't find a way to do it.
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Alex Balashov
Sorry, didn't carefully read my own link. :-)
I guess it should be:
${MATH(HANGUPCAUSE AND bit)}
Alex Balashov wrote:
Give this a whirl:
${MATH(HANGUPCAUSEbit)}
Got it here:
https://issues.asterisk.org/view.php?id=9891
Alex Hermann wrote:
Hello,
How can I do bitwise
to the
SPA3102b because of call waiting. Is there some way of avoiding this -
seems silly to me
Is this the way to do ?
or is there some way to create a gobalvar which is a dynamic dial string,
but I can't figure out how to modify a gobalvar on registration ?
Thanks
Alex
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I have been looking at a snom 300, which seems okay. the display goes a
bit haywire occasionally - not sure why yet.
Are the 320 worth the extra money ?
Alex
Stick with the older model snoms. So far I've seen nothing about the
820 to justify the significant extra expense
Hi
i have just recently installed asterisk 1.4 server with a digium card 410, i
used the zaptel packages in debian.
now I have notice the move to dahdi which seems to be a rename and some
changes as well.
is it a easy change from zaptel to dahdi ? any sort of gotchas to watch
out for ?
Alex
When the support for h323plus was announced for Asterisk 1.4.25, I tried
to build this support in Asterisk. For this, I checked out the h323plus
CVS from SourceForge, which reported version 1.20.beta5, and also the
ptlib-2.4.2 source RPM from Fedora 10. I finally managed to build a
chan_h323
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[context for spa3102a], can I put
reinvite here and if not where can i put it.
Thanks
Alex
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Yes, that's what SRTP is for.
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