Displaying 20 results from an estimated 3000 matches similar to: "Voice recognition recommendations?"
2011 May 12
1
Higher CPU usage on 1.6.1 than 1.4?
Hello,
We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now
experiencing higher CPU utilization on their server. I can't see anything
wrong, so is this just expected with 1.6? Can anyone help explain it?
Thanks for any advice.
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
2015 Mar 12
2
WebRTC demo phones
Hello,
Can anyone recommend a particular online WebRTC phone for testing with
Asterisk?
We tried:
- JsSIP, but even with the "enable video" checkbox disabled it sends video
options in the INVITE SDP and Asterisk rejects it with "Rejecting secure
video stream without encryption details".
- sipML5, but it won't register, perhaps something to do with not using the
Asterisk
2015 Mar 12
0
WebRTC demo phones
Sipml5 works. You need to have TLS enabled on asterisk web socket.
Mitul
On 12-Mar-2015 12:47 PM, "David Cunningham" <dcunningham at voisonics.com>
wrote:
> Hello,
>
> Can anyone recommend a particular online WebRTC phone for testing with
> Asterisk?
>
> We tried:
>
> - JsSIP, but even with the "enable video" checkbox disabled it sends video
>
2011 Jan 20
1
Introducing easySysAdmin - automated security and telecom fraud protection
Hello all,
Voisonics is pleased to introduce easySysAdmin, an automated
support/security platform, designed to save your engineer's time and prevent
hacking attempts and telecom fraud.
It comprises of an online service run by us, and a lightweight and
easy-to-install client on your side. Specifically of interest to Asterisk
users is the monitoring of SIP registrations, and automatic blocking
2014 May 22
1
maxsecs not working
Hello,
We have servers running Asterisk 1.8.20.1 and 11.7.0, and on both setting
maxsecs in voicemail.conf doesn't seem to have any effect. A voicemail
keeps recording after the specified time, and when the caller hangs up the
voicemail is saved in the mailbox.
Are we doing something really silly?
Here's the voicemail.conf. We have tried 'voicemail reload' and restarting
2013 Jan 03
3
faxdetect on/off on the fly?
Hello,
We want the ability to choose from an AGI script whether or not to enable
faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone
suggest a workaround?
Thanks for any advice.
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
-------------- next part --------------
An HTML attachment was
2011 Jan 19
0
progressinband, how much extra CPU load?
Hi everyone,
We have an Asterisk 1.4.17 user who has problems with sometimes not getting
a ring tone on the calling phone.
We're considering setting progressinband = yes, but would like to know how
much extra CPU load this will require? If anyone can give something even
roughly specific (eg "30% increase") that would be great, rather than just
"lots".
Also, are there any
2011 Jun 09
0
Change to pickups in Asterisk 1.8 - not working on local channels?
Hello all,
We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups
which previously worked fine have stopped working.
Can anyone advise if there has been a change in how pickups work?
Here is an example where 1000101 is trying to pick up a call to 1000103:
<SIP/product-local-00000005>AGI Rx << EXEC Dial
"Local/1000103 at product-pickup
2015 Aug 18
2
Asterisk 13 chan_sip trunk appending @string to dialled number
Yes indeed.
Do you have the dialplan, eg from /etc/asterisk/extensions.conf?
Something is getting this OUT_3_SUFFIX variable and including it in a Dial
to 172.22.4.12.
On 18 August 2015 at 16:21, Brendan Ord <bord at staff.onthenet.com.au> wrote:
> Starting to make sense when I saw this line:
>
>
>
> [2015-08-18 15:01:33] DEBUG[19366][C-00001cfc]: pbx.c:3785
>
2016 Jan 21
2
Mixing PJSIP realtime and flat files
Hello,
Is it possible to mix PJSIP realtime and flat file configuration in
pjsip,conf?
What we want is to set up endpoints in the ps_endpoints table with some
columns set but most being NULL, and then allow end-customers to optionally
add configuration by adding a pjsip.conf section.
For example, in ps_endpoinds might be an endpoint with id "asterisk-1" with
the transport, aors, auth,
2014 Jan 20
3
Asterisk not receiving call from VPN address
Hi,
We have a Kamailio and Asterisk cluster, both machines being on a real
103.x IP address and also on a 172.x OpenVPN address.
The problem is that when Kamailo receives a call from the VPN and forwards
it to the Asterisk server on it's 103.x address, Asterisk never sees the
call.
If Kamailio receives a call from the VPN and forwards the call to the
Asterisk server on it's 172.x
2015 Aug 18
2
Asterisk 13 chan_sip trunk appending @string to dialled number
Hello,
So, I found this line under macro-dialout-trunk, in extensions_additional.conf (FreePBX, so it controls the conf files mostly);
exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})
If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing..
Here's a paste of a few things out of the two files that I
2011 Nov 21
2
Continue AGI after Dial() following caller hang up?
Hello,
We would like to continue a Perl AGI after a Dial() it has done completes
following caller hangup. We would like to do this in the same AGI, and not
using a new AGI from the 'h' extension. It works fine when the called party
hangs up and the 'g' option is used, but not for caller hangup.
Is this possible?
If not a confirmation that this is the case would be very helpful.
2015 Aug 18
2
Asterisk 13 chan_sip trunk appending @string to dialled number
David,
I should also note;
246 is my extension, it has IP 172.22.3.238.
172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway.
The trunk is called ?testing? at the moment. The route that selects this trunk uses a 9 prefix.
This system is in semi-production, so there might be fluff in the log from other active calls.
Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
2014 Oct 27
0
Detect hangup due to RTP timeout
Hello,
Is there a way to tell in the dialplan, Asterisk AGI, or Asterisk AMI, when
a call has been hung up because the SIP rtptimeout has been reached?
Thank you,
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2010 May 19
2
Cause and cure for "Exceptionally long voice queue length queuing to Local"?
Hello,
We're seeing lots of warnings like the following, running Asterisk
1.6.1.12. Does anyone know the cause or cure?
One explanation I've come across is that the peer is congested and
sending RTCP messages asking us to slow the RTP down. Is there any way
we can verify this?
[May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice
queue length queuing to Local/12126412121
2010 May 13
1
What does Asterisk give to reject a re-invite?
Hello,
If you have canreinvite=no and a peer sends you a re-invite, what will
Asterisk reply with?
Thanks,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
2010 Dec 07
1
'Bookmarking' a place in a sound file
Hi all,
Is it possible to somehow 'bookmark' a place in a sound file? That is, the
user presses a key while a sound file is playing and that point is saved,
and some time in the future we can play the same sound file and tell it to
start playing from that point.
This would be done within a perl AGI program.
Thanks for any advice!
--
David Cunningham, Voisonics
http://voisonics.com/
US
2020 Oct 30
0
Multiple IP addresses and using same IP for outbound calls as inbound
Hi Dovid,
We can change the SDP in Kamailio, but Asterisk will still send its RTP
from its default address. The remote end is strict about accepting RTP from
the specified source and won't accept it. Have you any suggestions to solve
that problem?
Thank you.
On Fri, 30 Oct 2020 at 14:49, Dovid Bender <dovid at telecurve.com> wrote:
> Why not use OpenSips/Kamailoo in between?
2016 Dec 12
2
AMI version of CONNECTEDLINE
Hello,
Is there any equivalent of the CONNECTEDLINE function which can be called
from an application using the AMI?
Thanks for any ideas.
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
Australia: +61 (0) 2 8063 9019
-------------- next part --------------
An HTML attachment was scrubbed...
URL: