Displaying 20 results from an estimated 80000 matches similar to: "Mixing HASH() and LOCAL()"
2016 Sep 15
2
Tricking asterisk to think the call has ended, but it was continuing on the other side
No, there is no Music On Hold starting and the bad thing is the call
duration reported by asterisk was just few seconds while the call duration
reported by the provider was few thousand seconds, the max allowed. So they
will be able to terminate the call on the asterisk side and have it run on
the provider side.
Leandro
2016-09-15 19:18 GMT+02:00 Max Grobecker <max.grobecker at
2016 Sep 15
3
Tricking asterisk to think the call has ended, but it was continuing on the other side
I am banging my head over a simple asterisk trick I was seeing on one
asterisk server.
An extension dials an international premium number, the called number
answers, then the extension hangups, but the call continue to run on the
international number side, generating an high profit for the premium number
company and a big loss for the asterisk owner.
I think some sort of "transfer"
2008 Dec 02
1
func_odbc and hash problem
Hello,
Now I'm testing func_odbc and hash. My configurations are:
func_odbc.conf
[GETNUMBER]
dsn=sqlserver
;mode=multirow
;rowlimit=10
readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers
WHERE number=${SQL_ESC(${ARG1})}
extensions.conf
exten => s,1,Ringing
exten => s,n,Wait(4)
exten => s,n,Answer
exten => s,n,Set(NUMERIS=37037210602)
exten =>
2015 Jan 15
2
Showing sip subscriptions in Manager
Hello,
almost any useful CLI command has an analogue on Asterisk Manager
Interface, but I cannot find a way to get the list of subscriptions using
AMI. Which is the command, if any? The CLI command is "sip show
subscriptions"
Leandro
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2011 Apr 16
5
Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: <iq from="+
17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="
2013 Feb 24
2
AEL Macro are evil :-)
I just discover an "hidden" problem with AEL macro I want to have your
feedback. If you use a macro to dial out, like &dialout(${EXTEN}), the leg
extension will became ~~~~s~~~~ and if it happens you transfer the call,
that will be the callerid appearing on the other phone display.
I am just rewriting all the dialplan getting rid of the macro and using
gosub, even if asterisk is
2015 Mar 03
2
Dialing multiple channels with confirm
I'd like to dial two extensions (or external number) and ask for
confirmation to accept the call.
Dialing an extension, asking for confirmation and then dialing a second
extension if the call has not been accepted is easy by using the dial
option U(...), but if I dial two extensions at once, when the first
answers, the other stops ringing.
Any idea to make the first continue to ring until
2016 Jul 06
3
Impossible to use any recent asterisk version with chan_sip
This is a great news, thank you. I have open the issue,
https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the
relevant files, let me know if you need more info.
Leandro
2016-07-06 21:46 GMT+02:00 Joshua Colp <jcolp at digium.com>:
> Leandro Dardini wrote:
>
>> Hello,
>> I'd like to know if anyone of you is finding my same problems using any
>>
2016 Jul 21
0
Asterisk 13.10.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.10.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.10.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
2014 Jan 15
2
Asterisk ignoring nat settings
Hello,
I have an asterisk box with a peer configured with nat=force_rport,comedia,
but asterisk keeps sending the audio to the private IP address and ignoring
the client peer nat settings.
If I check the "sip show peer extension", I see both symmetric RTP and
Force Rport are set to yes, but asterisk seems ignoring them.
Force rport : Yes
Symmetric RTP: Yes
Asterisk is behind a
2016 Jul 02
3
Registration server with PJSIP
Hello,
I am moving from realtime chan_sip to pjsip and one of the problem I am
facing is the lack of "sipregs". With chan_sip, when an extension
registers, the server where it has registered to is stored in sipregs.
Is there something similar in pjsip? How can I find on which server the
pjsip extension has registered to?
Leandro
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2012 Aug 04
1
Suggestion of Server Specifications for Asterisk
What the minimum Server Specifications do I need to run
200 concurrent channels at the time with .WAV recording (MixMonitor)?
It will be connected via VOIP sip account.
Codec will be ulaw.
Which UK dedicated server provider do you recommend and how much bandwidth
do I need?
Thanks
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2016 Jul 21
2
Asterisk 13.10.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.10.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.10.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
2015 Jan 15
0
Showing sip subscriptions in Manager
You can use "Command" command, and "sip show subscriptions" as a parameter
--
Alex Epshteyn
email: alex at thirdlane.com
web: www.thirdlane.com
phone +1 415.261.6601
----- Original Message -----
> From: "Leandro Dardini" <ldardini at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at
2016 Sep 01
0
Asterisk 13.11.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.11.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.11.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New
2013 Nov 14
1
Queue linear "unordered" feature when using realtime
Hello,
I was trying to use a queue in linear order and to provide the exact order
of members to dial by adjusting the uniqueid value. Obviously it doesn't
work and it seems an old problem:
https://issues.asterisk.org/jira/browse/ASTERISK-18480
Realtime configuration can't identify "orders" in the list of results, so
the members for the queue are returned in random order.
2013 Nov 29
2
Answering agent
Hello friends,
when a call arrives in the queue, a CDR record is created, but there is no
info about which agent has picked up the call. I can find that info only in
queue_log.
Is there a way to have that info in the CDR or maybe in a variable in the
"h" context, when the call is ended?
Leandro
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2014 Oct 05
1
Voicemail message number off by one when using ODBC storage
Hello,
have you noticed the message num (VM_MSGNUM) is off by one?
For example, I receive the following message:
"Just wanted to let you know you were just left a 0:03 long message (number
7)"
but in attach there is the msg0006.wav
Leandro
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2014 Nov 14
1
SLA (Shared Line Appearance) and realtime
Hello,
do you know if it is possible to define the SLA configuration in the
database for realtime usage with asterisk?
Leandro
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2013 Oct 03
1
Disable the Connected Line info
When you set sendrpid=yes in sip.conf, a very nice feature is activated.
When dialing an extension, the callerid of the dialed extension is returned
back on the display of the calling phone. So if you call extension 100, you
can see you are calling Ann (for example).
I want to selectively disable the transmission of this information back to
the caller. How can I do it?
I tried setting