Displaying 20 results from an estimated 20000 matches similar to: "migration"
2009 Nov 07
6
Location
Where is everyone located?
I am in Washington DC.
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2009 Dec 19
5
sendmail
Anyone have a cookbook on configuring sendmail to work with Asterisk?
Or,a few config examples.
2011 Jan 02
2
incoming
Is it possible to have
Calls incoming to different DIDs?
I want an AA that handles 100s of businesses.
[Incoming-pizza]
Exten => 4045551212,1,Goto(pizza,s,1)
[Incoming-hvac]
Exten => 8085551212,1,Goto(hvac,s,1)
[Incoming-gutter]
Exten => 6175551212,1,Goto(gutter,s,1)
2011 May 22
5
call files .vbs
This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses
but I want to know in any case!
Can a vb script run somehow on a Linux machine or does it only work on
Windows?
If I were to build a call file script (described in this link
http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then
how does it work if my Asterisk machine is running on Centos 5.5?
I simply
2010 Jan 30
8
MATH
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten => s,n,Read(NUMBER,,1000)
exten => s,n,SayDigits(${NUMBER})
2010 Feb 08
2
IVR Demo / Create file / Move file / Demo all
Do you see any syntax errors?
Positive comments welcomed.
The short version of the logic is as follows:
create a file based on the NUMBER
create a an audio file
move to a new extension (label) and play the results
exten => 621,1,Answer()
exten => 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5)
; create a variable from a DTMF entry / 12 characters long
exten =>
2010 Feb 06
6
Dial script
Does anyone have a Dial script or a hint on how I can dial 10000
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined digit, the call gets bridged to me.
2009 Nov 21
1
Verification number / code
I want to distribute a random number to each of the first 100 callers to my
IVR.
This random number will be matched to their telephone number.
Where in Asterisk can I do this? And, how?
Logic.
Call arrives.
Context for announcement begins.
You will receive a authentication code at the end of the message.
Then, if they press a certain digit to confirm then I simply pass a code to
them.
These
2013 Apr 08
3
extensions.conf / test DID
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI> sip show registry
Host dnsmgr Username Refresh
State Reg.Time
sip3.voipvoip.com:5060
2009 Nov 06
1
Question on peering two Asterisk servers
I have two servers each with one TDM card from Digium and three POTS
lines going into each server (each POTS line is one individual
number). and I want to have all incoming calls into server B be
directed to the IVR on server A and to be able to fail over to Server
B in case Server A has an issue (it can be a manual failover and I
actually prefer it to be a manual process).
I've
2010 Dec 06
1
no audio
Any reason why I don't get audio on the channel after it rings and the
end user picks up.
Here are my files.
CONSOLE=Console/dsp ; Console interface for demo
OUTBOUNDTRUNK=SIP/callwithus
[default]
include => stdexten
exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks))
exten => s,n,Wait(2)
2010 Dec 08
1
debug audio or channel
Does anyone have any short answers on how I can fix this problem:
A calls B.
B rings
Says connected.
But the call is not bridged and therefor no audio passes.
very simple dial plan.
Frustrated.
v 1.8
2013 Apr 06
1
sip registration
I have a very lite layout and attempting to get the SIP configuration set
up initially before proceeding into other areas.
VMware is running my Asterisk 11 on Ubuntu 12.
Shouldnt I be able to at least ping the SIP provider IP?
I run command "sip show registry" and do not see it set up.
I run sip show peers and I do see an entry.
I have not configured anything other then entries in the
2009 Aug 03
4
single port voip gateways
I have used the handytone 488 from grandstream in the past....
However I need to be able to send a number to a unit like the 488 and
have it dial out.
Is there a unit like this available? Basically a 488 unit that can place
a call out.
Jerry
2010 Mar 04
1
Remote Agents
I'm trying to setup a situation where I have agents on POTS lines at remote
locations. I want to allow them to call a DID, log into the Asterisk
system, and be an agent. Ultimately I'd like Asterisk to call them at the
number they were at when they logged in.
Does this functionality exist in Asterisk?
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2010 Jul 01
1
call file question
I am sure this is simple, but have been struggling. I want to create a
call file that dials out a particular Dahdi channel to enable call
forwarding on a POTS line. I have this in extensions.conf:
[custom-callfwd]
exten => s,1,Answer
exten => s,n,Dial(DAHDI/4-1/*717157750)
exten => s,n,Verbose(${DIALSTATUS})
exten => s,n,Hangup
[custom-callfwdcanc]
exten => s,1,Answer
exten
2015 Jun 18
3
setting outbound caller ID
Thanks very much for all the responses. I now have a few more things to try.
I should have noted that I am using IAX2 rather than SIP to connect to my
provider. I do have some internal phones that use SIP to connect to my
asterisk box, as well as some corded phones connected through a Digium
DAHDI-driven card.
I am certain that the old number that is showing up as my caller ID is not
present in
2008 Jan 16
2
[IAX] Up-to-date list of soft- and hardphones?
Hello
There's a lot of information on VoIP at www.voip-info.org ...
but there's also a lot of outdated information there as well :-/
Since SIP is a pain to use when NAT is involved, especially when both
the Asterisk server and the remote phones are behind NAT... I'd like
to try IAX to see how it works and if it solves the issue.
I'd like to start with a softphone (Windows
2018 May 10
2
SIP Codec negotiation
I receive an INVITE/SDP containing:
m=audio 11310 RTP/AVP 3 0 101
which I interpret as gsm, ulaw, rfc2833.
and I reply with an OK/SDP containing:
m=audio 15884 RTP/AVP 0 3 101
which I interpret as ulaw, gsm, rfc2833.
How can I tell which codec was actually used for the call?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards