Hi,
I'm interested in what software (Free or course) that people use when they
want to add a dial by voice service to their asterisk system. Meaning I
pick up the phone.. dial some extension. it prompts me for name.. I say
John Smith.. and it dials his extension and connects the call..
: [asterisk-users] Making Asterisk a Voice Router
On Mon, 2007-10-22 at 09:39 -0400, end1r wrote:
I’m interested in what software (Free or course) that people use when
they want to add a “dial by voice” service to their asterisk system.
Meaning I pick up the phone.. dial some extension… it prompts
] Making Asterisk a Voice Router
Nice job! I took the liberty to post it on AstPligg as well:
http://tinyurl.com/268bac
Thanks
l.
In data Mon, 22 Oct 2007 16:22:13 +0200, Jared Smith [EMAIL PROTECTED]
ha scritto:
On Mon, 2007-10-22 at 09:39 -0400, end1r wrote:
I’m interested in what software
Do you have any console messages?
SCCP uses a station start tone message with a value of Inside Dial Tone and
a direction of Tone Output User. and the line instance and Station Tone
Output Direction should be set to something other than 0.
SCCP runs over TCP so you should get this message,
Hi all,
Does anyone have an application/script or extensions.conf file which will do
the following?
When a new VoiceMail is left for a user, the asterisk system will place a
call to a cellphone/pstn number(via some provider). When the user answers
his cell/home phone, comedian mail will
You really need 2 packet traces from the "outside and "inside" side of the router showing the TCP process between the phone and the server. Is the connection even getting opened, is there an FTP error code?
Post traces of TCP and we can look what happening
-- Original message
Looks like the CallManager is unable to find the endpoint in its database. Make
sure asterisk trunk on the Call manager is in the same calling Search Space
as the phones are in, or make sure there is access between the calling search
spaces
-Eric
-- Original message
I know every second counts in a real 911
situation, but what about adding a pause in the call flow. Maybe a 1 second
pause before actually passing the digits to the provider. This gives the user 1
second to realize the mistake and hang up, longer than 1 seconds is a real
emergency.
I thought # to transfer didnt work if you have a t,t orr in your dial
string since asterisk remains in the media path?
but its just a guess.
-- Original message --
From: Douglas Garstang [EMAIL PROTECTED]
-Original Message-
From: Douglas Garstang
Yes sorry.. i was thinking of something else. I had a problem where I put the T
in the dial string and the media wouldnt go end-end, but thats because Asterisk
has to remain in the RTP stream to hear the #.
Much different that what your reporting... sorry to mis-lead ya...
--
To convert the phone to SIP you have to unlock and set the TFTP server to your
TFTP server address as explained below. But...
you also need to make sure you have a SIP image in the root directory of your
TFTP server and also edit the OS79XX.TXT file to contain only the SIP image
name (no
Hi,
I would like to get some Voice Recon working with a small asterisk server I
have (10 endpoints). We would have no more than 3 calls at the MOST up
trying to do recon, so volume is not a problem.
Anyone have a link to the download AND a HOWTO for Sphinx /w Asterisk?
Regards,
-Eric
Hi all,
Anyone knows of any Voice Recognition applications which use
VoIP? Preferably open source.
I am basically trying to build a Voice Call Router, something
to recognize a spoken name and then transfer the caller to the right partys
extension?
TIA.
-Eric
Looks like you have sip.conf set up to expect registrations for tycisco
since it has a D for dynamic.
You can either set up the 7960 to register with asterisk and use something
like this in sip.conf:
[tycisco]
type=friend
username= someusername
secret= somesecret
insecure=no
mailbox=757
Sounds like you have a context mis-match. Please posts your sip.conf and
extensions.conf files.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mak kwak
Sent: Monday, April 04, 2005 10:56 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] newbie - want
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