2011/9/23 amit anand onewaytoconn...@gmail.com
Hi
you cannt do itin default CDR of the asterisk, To do so you can use Mysql
and do it from dialplan
How ?
Shall I simply append a new record into MySQL CDR table ?
On Thu, Sep 22, 2011 at 18:53, Olivier oza_4...@yahoo.fr wrote:
On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote:
I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been
reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy
should be used to provide an interface for Asterisk to get kernel
timing. - espescially if using
Hi list
I have 2 servers with a TDM400 card, port 1 populated by an FXO (red)
module), port 4 populated with an FXS module. I am using dahdi
linux and tools 2.5.0.1. The servers are running CentOS 4 and the other
box CentOS 6.
Both modules have been working fine but recently stopped
On 09/23/2011 02:50 AM, Ishfaq Malik wrote:
On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote:
I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been
reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy
should be used to provide an interface for Asterisk to get
Add match_auth_username=yes in the [general] section of sip.conf
Remove from each peer any insecure entry
Usually I add also auth, defaultuser and username to the peer
definition, but some of these entries are not needed.
Leandro
2011/9/23 David Björkevik da...@dynamore.se
Dear list,
We
Hi,
I have the following setup:
Asterisk - Nat - Internet - Nat - 2 x SIP endpoints
With directmedia=no I can make a call between the two SIP endpoints; the RTP
stream being passed through the Asterisk box.
Obviously, this is sub-optimal. I attempted to enable bridging of the call
between the
Leandro,
Thank you for your input!
I tried this and it's still the same.
(although I still have _unrelated_ peers with the insecure entry)
/David
On 2011-09-23 14:24, Leandro Dardini wrote:
Add match_auth_username=yes in the [general] section of sip.conf
Remove from each peer any insecure
Please check no other peers with insecure entry are registered from the
same IP. Asterisk takes some shortcut and try authenticating peers by IP
address before authenticating them by username/password.
Leandro
2011/9/23 David Björkevik da...@dynamore.se
Leandro,
Thank you for your input!
I
Hi All,
I am new in asterisk. In my office we have purchased ISDN pri
line with 30 channels. we have more than 60 soft phone nodes and the
internal asterisk connectivity between extensions are working with soft
phones. Can anybody tell me which pci or pci express digium card can be
compare the prices between sangoma and digium pri boards!
Sangoma's oards here in Germany are cheaper as the ones from digium.
if you need detailed help, you can contact me, and I can workout for you
something as well as helping you setting up your pbx!
Tamer
Am 23.09.2011 15:01, schrieb
Kevin P. Fleming kpflem...@digium.com wrote:
On 09/23/2011 02:50 AM, Ishfaq Malik wrote:
On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote:
I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been
reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy
should
Dear List,
Thank you for your suggestions. This turned out to be an issue with our
SIP provider, and has now been resolved.
Regards,
David
On 2011-09-23 14:44, David Björkevik wrote:
Leandro,
Thank you for your input!
I tried this and it's still the same.
(although I still have
On Fri, Sep 23, 2011 at 06:31:16PM +0530, michael k wrote:
Can anybody tell me which pci or pci express digium card can be used
to connect my asterisk server and the ISDN pri line with 30 channels ?
You need to search for an E1 card (32 channels total, 30 voice):
Hi,
I tried to sendfax a text file, it was received successfully and the context
were in ascii format (readable form). As I tried to send a fax in .tiff
format (converted from pdf format using ghostscript), the context I received
in fax is in binary form. The dial plan is listed below;
exten =
Suppose I have two IP aliases on one asterisk box.
I have to talk to SIP friend A using IP x.x.x.x and I have to talk to
SIP friend B using IP y.y.y.y.
(In case you're wondering, the reason is that we have two accounts with
a service provider and different features and rates are tied to the
Hello,
I have a question regarding a usb hub which is connected with a usb
bluetooth adapter
I setup asterisk16 with chan_mobile.Is working good.
1). When I use the bluetooth adapter into computer usb port is working
voice 2 ways without delay : test OK
2). When I use the bluetooth adapter into
So I was hoping I would be able to set the source IP that we use when
talking to the two different SIP friends. I see externip in general
options, but is there nothing equivalent that can be set per user/peer?
Hi,
as far as I know, you cant do this on a per peer basis.
I suppose you run two
Hello,
I have an AGI script that occasionally disappears without completing its action
and asterisk logs the following.
Local/0123456@context-f46e;1AGI Script script.php completed, returning 4
Spawn extension (context, 0123456, 2) exited non-zero on
'Local/0123456@context-f46e;1'
I
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mehmet
Avcioglu
Sent: Friday, September 23, 2011 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI Problem
Hello,
I have an AGI script
So I was hoping I would be able to set the source IP that we use when
talking to the two different SIP friends. I see externip in general
options, but is there nothing equivalent that can be set per user/peer?
Hi,
as far as I know, you cant do this on a per peer basis.
I suppose you run two
On Sep 23, 2011, at 8:07 PM, Danny Nicholas wrote:
Just a WAG - 4 is the error level returned by your php script, where it
normally returns 0.
Yes would thing so. But at no place in my script I intentionally exit with 4. I
believe 4 is SIGILL (Illegal Instruction) so my script might be seg
Currently if asterisk loses its connection to the postgresql it does not
attempt to reconnect. I have searched all over for a setting that would have
asterisk attempt to reconnect but I can not find anything. Is there something I
am missing?
Thanks!
-Eric
The Asterisk Development Team announces the release of Asterisk 1.8.7.0.
This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.7.0 resolves several issues reported by the
community and would have not been possible
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Hiller
Sent: Friday, September 23, 2011 1:29 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Postgresql Reconnect on connection failure
Currently if asterisk loses
On 09/23/2011 12:16 PM, Mehmet Avcioglu wrote:
On Sep 23, 2011, at 8:07 PM, Danny Nicholas wrote:
Just a WAG - 4 is the error level returned by your php script, where it
normally returns 0.
Yes would thing so. But at no place in my script I intentionally exit with 4. I
believe 4 is SIGILL
Are there any free DID in Illinois 708-839 or area?
--
Joseph
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On Fri, Sep 23, 2011 at 09:30:59AM -0400, cov...@ccs.covici.com wrote:
Kevin P. Fleming kpflem...@digium.com wrote:
On 09/23/2011 02:50 AM, Ishfaq Malik wrote:
On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote:
So am I correct in assuming dahdi_dummy isn't needed/useful
anymore?
Hi All;
I noticed in the queues.conf the configuration for recording the calls in the
queuing, and regarding to the filename (or any other parameter), it is written
that I can determine the filename using the command:
Set(MONITOR_FILENAME=foo)
But it should be called from the dialing plan,
Shaun Ruffell sruff...@digium.com wrote:
On Fri, Sep 23, 2011 at 09:30:59AM -0400, cov...@ccs.covici.com wrote:
Kevin P. Fleming kpflem...@digium.com wrote:
On 09/23/2011 02:50 AM, Ishfaq Malik wrote:
On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote:
So am I correct in assuming
On Sun, Sep 18, 2011 at 07:51:43PM -0400, Zeeshan A Zakaria wrote:
This DTMF problem has always been there and there is no real solution
for it, other than using those expensive PBX systems like that from
Avaya, Cisco, etc. This problem happens when you are sending inband
DTMF tones. Via
In Trunk, or earlier, is it possible to execute an AGI or any piece of
the Diaplan when a new peer registers successfully?
Venefax
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
On 09/23/2011 09:59 PM, CDR wrote:
In Trunk, or earlier, is it possible to execute an AGI or any piece of
the Diaplan when a new peer registers successfully?
No.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax:
One way of doing something when a peer registers is to use AMI to monitor
events and when a register event occurs do what you want.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 23, 2011, at 7:08 PM, Alex Balashov wrote:
On 09/23/2011 09:59 PM, CDR wrote:
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