Displaying 20 results from an estimated 4000 matches similar to: "Dialplan Problem - Outgoing"
2008 Mar 19
0
How configure Voice mail for multi users.
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,.......and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf ..... settings below..
[voicemail]
exten => _X.,1,Dial(SIP/${EXTEN})
exten =>
2008 Mar 19
3
How to configure Voice mail for multi users.
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,.......and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf ..... settings below..
[voicemail]
exten => _X.,1,Dial(SIP/${EXTEN})
exten =>
2007 Aug 28
1
calls being forwarded to neighbor?? please help, thx :)
hi ppl :D
my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2,
i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a
grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have
a machine (machine 1), which functions as my router and machine 2 and sip device are behind it,
grandstream box
2005 Feb 22
0
PSTN tones with ISDN4Linux
Hi all,
I'm playing with Asterisk and I've already configured all needed .conf
files.
It works quite well, but now I need your help to tune the system: when I
place a call from a softphone to the PSTN, I can't hear directly Telco's
tones and I can't use its services, e.g. a mobile's answering machine.
I don't know if I have to modify the dialplan or if it depends on my
2007 Jun 26
1
CDR Records "s" as dst
I am using VoiceOne http://voiceone.it/ as my management interface.
I am not 100% sure when it started, but my CDR is now full of "s" as
the DST instead of the actual dialed number.
As I understand it - it is because it is being recorded in the CDR
while in a macro (as below).
Is there any work around so that I can record the actual dialed number?
[macro-dialout]
exten =
2007 Oct 30
1
G.729 transcoder beetween asterisk to avaya
Dear all
I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine.
I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download
2006 Dec 28
0
Say who is using the PSTN?
I use an SPA3000 to connect to the PSTN (SIP/pstn). Since I only have one line,
if it is in use, and someone else tries to dial out, they get the
all-outgoing-lines-unavailable message played.
I'd like to find a way to instead tell them which extension is using the PSTN
line. I know that info is available in the manager API, but I have no idea how
to get access to it from either the
2006 Nov 12
1
outgoing works, incoming fails on asterisk passthrough to inter-tel
Hi everybody,
Well, I've finally got asterisk to to talk nicely with my Intertel pbx.
Currently there is a outside T1 line (e&m wink start, esf, b8zs)
connected to asterisk, and then asterisk connected similarly to my
Intertel pbx. For right now all asterisk is doing is passing calls
between the two.
When I call out from the pbx, I can connect perfectly to the outside
world. When I
2008 Feb 15
1
DialPlan help with Analog Fax Machine
I'm struggling to get my dialplan to work with a simple analog fax
machine.
I have TDM400B zaptel card with an FXO and FXS port. I have the FXO
port connected to the POTS machine and the FAX machine connected to the
FXS port.
The FAX machine itself works fine, I can FAX outgoing messages fine. I
can also dial the FAX extension from the internal context, the FAX
machine answers and I
2007 Feb 15
0
SIP Redirect from Asterisk behind a NAT
SetUp:
- Asterisk behind a NAT,
- Red Hat 9.0
- Asterisk 1.2.14
My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my
dial plan set up so that when outside callers dial the DiD, the call is
answered by my auto-attendant. The caller can then select who they'd like
to speak to and the call is transferred to the external line associated
with that person (usually a mobile
2007 Jul 01
0
Transfer outgoing call - macro
Dear All,
I have a problem with call transfer. When I dial a number and then I want to transfer current call to an extension, I'm getting disconnected. Transfering incoming call works fine. I'm using macro for dialing.
extensions.conf:
[from-internal]
ignorepat => 9
exten => 200,1,Macro(stdexten,200,SIP/dzalewski)
[macro-stdexten]
exten => s,1,Set(temp=${DB(CFU/${ARG1})})
2009 Feb 12
4
Asterisk Queue and URL Calling
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the call
center the agent or extension which answers the call , before pickup the
phone and talk to customer , view his/her information if it is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new
2006 Oct 31
1
Asterisk does not bridge zap channels on outgoing calls
Hello... I have a big problem with asterisk. Every time i make a call
asterisk does not bridge the zap channels. The zap channel from which
i'm calling remains in state:ring and applicaton:dial and the zap
channel with the external line configured remains in state:dialling an
Application:AppDial.
Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None)
Zap/9-1 int_omg 09399 5 Ring
2009 Jun 29
0
FW: re: Asterisk Outbound with Failover, alarm notification, dial status and hangupcause capturing to CDR from Dialplan
Managed to implement this on asterisk v1.4.24.1,
Also, Hangupcause updating to user field.
However, this only works on the edge of my voice network (demarcation
point)
It does not work on my internal routing boxes as I use IAX to route
between remote sites.
I was thinking of using some sort of SIP variables to transport these
results over the IAX trunk..
Any bright ideas folks???
2006 Mar 06
0
No ring when doing blind transfer.
Hi,
I have an odd problem when doing a blind transfer. The transfer is
intiated and the transferred caller hears nothing until the timeout. I
have tried setting the 'r' and the 'm' variables in the dial command.
Nothing happens when I use the 'r' variable when I use the 'm' variable
I briefly hear music on hold and then it stops until the timeout for no
answer
2009 Jun 28
0
Recommendation / doubt about building of dialplan
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi all!
Now that I have a little more time, I was debugging my dialplan and it
was of the following way:
- -------------------------------------------------------------------------
; DGB - 20090615
[macro-dial]
exten => s,1,Dial(${ARG1},15)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@voicemail,u)
2005 Feb 12
0
Outbound calls on a busy Zap/1: BUSY vs. CHANUNAVAIL
Hi,
I'm using the macro below in extensions.conf for most of my outbound
calls. One issue with my current configuration is that when I make an
outbound call it doesn't properly detect that my PSTN line (Zap/1) is
busy with another call and then overflow to my outbound IAX
connections. I think the root cause is that DIALSTATUS gets reported
as BUSY. The debug output is below. My desired
2008 Jan 02
2
Asterisk dialplan date and time operations
Hi all,
Im using Asterisk 1.4.11 and I want to proceed some time and date operations
in my dial plan. (for a time shifted callback).
Should look like:
CURRENT TIME + x minutes.
Of course it should increase the hours for example in this case:
10.59 + 5 minutes = 11.04
I guess I've to use the math function in 1.4 but how can I manage easily the
time operations?
Kind Regards,
Erik
2006 Mar 30
0
BUG: FOP reports incorrect (duplicate) IP address until restarted
Hi,
This problem may be related to a configuration problem but I believe it
is a bug in the FOP since restarting the FOP server clears the problem.
Here is the scenario: Using AgentCallBackLogin and have four agents
logged in a call is made to one of the agents directly from an internal
phone. Okay so far. Call is hung up and the same extension is used to
call another agent okay again, no
2004 Nov 30
0
No voice when I dial out
I can dial from 601 to a public number.
The public number rings. I pickup and hear nothing, while on 601 it keeps ringing.
(BTW, is it right to say "ringing" on the active phone?)
The *CLI> doesn't show me anything useful:
Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack
Executing SetGlobalVar("SIP/601-8238",