Displaying 20 results from an estimated 3000 matches similar to: "Showing sip subscriptions in Manager"
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"
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="
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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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
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
2017 Apr 30
2
softphone instead of desktop phones
On 30 April 2017 at 16:54, Tech Support <asterisk at voipbusiness.us> wrote:
> I thought this was a non-commercial list.
>
>
Yeah, I wouldn't mind so much if it had actually answered the original
poster's query. "Switch to our proprietary solution and we can offer you
this proprietary solution" isn't a contribution, it's an ad.
-Barry
>
>
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 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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2017 Apr 19
4
PBX selection
The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more.
You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but
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
2011 Dec 27
3
how to stop hacking of my server
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except iptables ? I want to stop on the basis
of sip.conf account only. bcoz I can't apply iptables rules on server it's
remote server of server provider and we used it for making voip call for
customers.
for the time been i have close all sip accounts. but can't stop for more
then
2012 Aug 03
1
asterisk realtime database structure
Hello,
I was wondering if there is a tool that can create the realtime database
structure for latest Asterisk version or a web resource/file containing
the sql scripts. Hope I haven't missed obvious things, I had no luck
searching on the web, in the wiki I found few pages with bits of sql or
table structures, like:
2011 May 14
10
Asterisk-cpu utilization > 60 %
Hi,
On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest.
Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly
1-2 concurrent calls. No other activity on server. Top shows asterisk on
top.
Its quad xeon server with 4 gb ram.
Any suggestion where should I start and how should I go about with my
investigation.
Thank you and have a great weekend.
Sans
2007 Oct 31
5
Druid
Is anyone out there using Druid?
After the switchbox announcement today I've been looking into some other
gui's and as I'll probably do a trial install this weekend of the free
switchvox iso but I thought I'd ask is there any other guis I should be
burning trial ISO's of as well?
Regards,
Dean Collins
Cognation Pty Ltd
dean at cognation.net
<mailto:dean at
2012 Jul 30
4
Multi-Tenant PBX with Asterisk
Hi
I came across couple of pointers on the Internet regarding solutions
available for providing hosted PBX service.
1. Multiple PBXs: Using separate hardware to host each PBX. Pretty
straightforward, but no hosting company wants to use it.
2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of
Asterisk. I.e. partitioning a single instance of Asterisk into multiple
PBXs by way
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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2017 Apr 30
3
softphone instead of desktop phones
Thirdlane Connect can be used as a softphone. It works in modern browsers (no installation is required), on Mac, Windows and Linux desktops, and on mobile phones.
Besides basic softphone functionality, it provides instant messaging, group chat (channels), voice and video conferencing, and screen sharing. It integrates with a variety of applications and CRMs such as Salesforce, Zoho, Zendesk,