Displaying 20 results from an estimated 8000 matches similar to: "No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?"
2010 Aug 21
7
Opensource Speech recognition for Asterisk
Hi Everyone,
Has anyone got any opensource speech recognition software to work with
Asterisk? Please only list WORKING ones. Not the "theoretically" should work
ones!
Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100821/4d11d6c0/attachment.htm
2010 Sep 22
5
OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Hi Everyone,
I have setup an OpenVPN tunnel between Server A (running Asterisk) and
Server B suppling it's SIP Phones with DHCP pool of IPs.
So, the tunnel is established nicely and everyone can ping others. "sip show
peers" shows the local subnet of the SIP Phones registered (192.168.100.0/24
).
But there is the old bad one-way audio. Calls also drop after few seconds.
In the SIP
2010 Nov 07
2
Any good guides for installing Asterisk on Embedded systems like Alix boards?
Hi Everyone,
Knowing that running Asterisk on an embedded board like the Alix2d3 requires
some fine tuning. Do you know of any good guides out there that does this
from beginning to end? Looking to run this in a small office environment.
Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2010 Sep 14
9
Speech To Text on linux with asterisk
Hi,
Is it possible to record say 30 seconds of audio and then have LumenVox
convert to text ?
or any available tool open source for speech to text .
Regards
Dhaval
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm
2010 Jul 23
1
Why does a bridged channel stay open for 4 hours?
Hi Everyone,
Using a PRI with Sangoma A101D and Asterisk 1.4.2.x.
I notice that occasionally after a call is disconnected and both the phone
devices and the the channel is down but the bridge stays open for hours.
Channel Location State Application(Data)
Local/9054445555 at fro (None) Up Bridged Call(Zap/4-1)
9054445555 is the inbound DID on the
2010 Jul 29
2
Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1
Hi Everyone,
This is probably more related to Linux than to Asterisk. Analogue channels
on a system were un-responsive on Monday morning. Apparently something
happened over the weekend and the router went off or it lost it's DSL
connection.
[Jul 23 22:50:01] VERBOSE[12437] logger.c: -- Remote UNIX connection
[Jul 23 22:50:01] VERBOSE[27087] logger.c: -- Remote UNIX connection
2013 Feb 24
3
Asterisk AMI - Create a daemon (background process)
I wanted to create a daemon (background process) in PHP. A daemon will use
socket to connect with Asterisk AMI to send events and listen the actions.
A daemon will also listen the commands from agents via HTTP, for example:
A agent pressed a hang up button on a browser - it will send http command
to a daemon. A daemon received a command and will then send Hang Up Action
to AMI.
How should a
2010 Jul 01
3
Originate multiple channels
Hello,
Is it possible to use the asterisk manager interface to originate
multiple channels?
like
Action: Originate
Channel: SIP/101&SIP/102
So that both extensions 101 and 102 rings simultaneously.
I am using asterisk manager interface over http.
Thanks
2010 Sep 20
3
Extension continues ringing after caller hanged up
Hi,
I use asterisk with sip3000 device with "sip-aho" connected to PSTN and
"sip-ahi" connected to a phone.
When call arrives from PSTN, the *phone continues ringing even after caller
hanged up*.
The dialplan contains the following lines:
[from-pstn]
...
exten => 99,n,Dial(SIP/sip-ahi,30,g)
exten => 99,n,Hangup()
The asterisk properly detects hangup of the caller as I
2010 Jun 21
3
Create Conference and exit myself
Hi,
I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4
I am looking for the following functionality:
``````````````````````````````````````````````````````````````````````````````````````````````
I receive a call from Mr. A.
I put Mr. A on hold.
I dial Mr. B
I connect Mr. A's call (which was on hold) to Mr. B and I get out of the
call.
Mr. A & Mr. B are in
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
-------------- next part --------------
An HTML attachment was scrubbed...
2012 Sep 26
1
Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
Hello,
I'm having issues connecting throu PRI with the following error "Requested
transfer capability: 0x00 - SPEECH"
Below are the logs:
== Using SIP RTP CoS mark 5
-- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003",
"CALLERID(num)=xxxxxxxxx") in new stack
-- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",
2010 Oct 21
10
Asterisk 1.80-rc5
Just done a clean install of rc5 on a totally new machine and found the
following:-
/etc/init.d/asterisk start
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
Dave Cotton
2010 Oct 23
2
Just Take dCAP at Astricon?
Since it is Saturday evening (7PM EST) I am asking this on the list in case
someone who knows sees it and can answer.
Astricon is in my back yard for the first time, and I could hit you with a
rock. I would always like to attend, and spoke at the 2007 Astricon in
Phoenix but don't have the idle cycles.
Question: Can I just go to Astricon and take the dCAP exam only? In and
out? Cost?
I
2012 Jun 11
1
Differences between PBX and SBC
Hello,
I would like to know the difference between encrypt the rtp and signaling
between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm
trying to understand whether an SBC could fit an Asterisk deployment
Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2010 Jul 24
4
getting some segmentation faults with 1.8
I downloaded the latest 1.8 (27922) but got some segmentation faults.
The first one was when it loaded cdr_odb, and so I changed menuselect
not to compile that one, but the second one was when it tried to load
chan_agent and so I stopped there to see if anyone else was seeing
this. The agents.conf is all commented out except for [general] .
Anyone know what is happening?
Thanks.
P.S. I deleted
2010 Oct 14
6
Audiocodes firmware
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body text="#000000" bgcolor="#ffffff">
<font size="+1">Does anyone have links to the most recent audiocodes
2010 Oct 22
5
dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook?
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm