On Tue, Apr 9, 2013 at 7:56 PM, Olivier oza_4...@yahoo.fr wrote:
Hello,
What about having a new DB_IFEXISTS function in Asterisk ?
It would accept two arguments : one being family/Key just as DB function,
and the other being a default value which would be returned if family/key
value does
2013/4/10 Satish Barot satish4aster...@gmail.com
On Tue, Apr 9, 2013 at 7:56 PM, Olivier oza_4...@yahoo.fr wrote:
Hello,
What about having a new DB_IFEXISTS function in Asterisk ?
It would accept two arguments : one being family/Key just as DB function,
and the other being a default
Please, excuse me but I'm not sure I got your suggestion and I'm
realizing I didn't correctly describe my lab set up.
At the moment, the router between both servers provides Internet access
to server1.
That means it has one WAN interface eth0 which is on server2 side and
one eth1 LAN
On 04/09/2013 09:14 PM, Joseph wrote:
On 04/09/13 19:27, Joseph wrote:
exten = 4,n,Set(goaway=${CALLERID(number):0:2})
exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1)
But the first line suppose to compare the caller ID to the one
On 04/10/13 08:29, Matthew Jordan wrote:
On 04/09/2013 09:14 PM, Joseph wrote:
On 04/09/13 19:27, Joseph wrote:
exten = 4,n,Set(goaway=${CALLERID(number):0:2})
exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1)
But the first line
Is anyone using something to log SIP results (connected/not, latency) that
they really like? We do some logging using simple scripts writing the
results of sip show peers to a text file if customers report issues, but it
would be nice to have a tool that logs all the time and lets us do some
On Wed, 10 Apr 2013, Carlos Alvarez wrote:
Is anyone using something to log SIP results (connected/not, latency)
that they really like? We do some logging using simple scripts writing
the results of sip show peers to a text file if customers report issues,
but it would be nice to have a tool
On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards
asterisk@sedwards.comwrote:
dumpcap can capture all of the SIP (and RTP) packets into a series of
files without a huge performance hit.
A cron job can pbzip2 the files and delete if over x days old.
That's completely different. We
http://www.artifact-software.com/?page_id=1666
Would this help?
Put a JasperReport graph or two in a report step.
Ron
On 10/04/2013 2:02 PM, Steve Edwards wrote:
On Wed, 10 Apr 2013, Carlos Alvarez wrote:
Is anyone using something to log SIP results (connected/not, latency)
that they really
Hi,
I am working on a small inbound call center solution that uses an ACD system. I
might add an IVR system later on. I only have 2 extensions set up (extensions
1000 and 1001), I want the system to put new calls in a queue if both
extensions are busy. I am currently subscribed with a SIP
On 13-04-10 04:08 PM, Tommy Cooper wrote:
Hi,
I am working on a small inbound call center solution that uses an ACD system. I
might add an IVR system later on. I only have 2 extensions set up (extensions
1000 and 1001), I want the system to put new calls in a queue if both
extensions are
This line :
exten = *DID number*,2,Dial(SIP/1000) is redundant and useless when you
are already using Queues. So just remove it and it should work.
What happen is, your dial-plan executes at 2nd priority DIAL a SIP
extension 1000 .. produce a call and at hang-up finishes no Queue/ACD
Howdy,
I'm building a webapp to allow my techs to do minor dialplan edits and
trigger a reload on my PBX's running 1.8
I have no problem triggering a 'reload pbx_config.so' via manager, The
problem is how can I see the results of my reload?
For example a missing close parenthesis which would
Hello Brynjolfur Thorvardsson,
Can I take a look at you CDR reporting tool?
I'm planning on using it on Postgresql but MySQL could be used too.
Thank you!
Elder D. Arohuanca
dCAP
Lima - Peru
On Fri, Feb 10, 2012 at 11:55 AM, asterisk jobs asteriskcod...@gmail.comwrote:
No, that doesn't do
Thank you for your prompt reply. I removed that redundant line and now
everything seems to work fine. Except outgoing calls that is, whenever i try to
call an outside number the phone rings, the user can even answer back but then
it hangs up after about 5 sec.
extensions.conf:
[sip-phone]
Hi,
We have recently implemented asterisk as the office PBX, and as a GUI, we
chose Elastix. All seems quite easy and am very pleased with the way
Asterisk is running and handling calls. To summarize briefly, our setup
looks like this:
LAN - Asterisk PBX (Elastix) - Asterisk TDM (basic CLI
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I am trying to set the CDR(userfield) to a certain vaule using the
application map of features.conf but I am not able to do it. When I
receive a call I would like to tag it with a client code (3 digit
numeric) so I can referenci it later from
On On Wed, 10 Apr 2013, Carlos Alvarez wrote:
Is anyone using something to log SIP results (connected/not, latency) that
they really like? We do some logging using simple scripts writing the
results of sip show peers to a text file if customers report issues, but it
would be nice to
*Hi pal.*
*
*
*Maybe you forgot to enclose the near parenthesis found? 'Goto(7124,1' in
extension file, Firstly correct this syntax then try again.*
*
*
*BRs*
*/***
*Nguyen Trung (Mr)*
*Trust me I'm an Engineer!*
* *
* VEGA
Hi,
You can check extension status using chanisavail function. And extension is
not free, you can divert your call to queue.
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail
Regards,
Bharat Lalcheta
On Thu, Apr 11, 2013 at 1:38 AM, Tommy Cooper tomcoope...@yahoo.com wrote:
Hi,
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