Displaying 20 results from an estimated 600 matches similar to: "How can I dial (0 + telephone number)"
2008 Oct 09
2
Menu for call forwarding or voicemail
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten => 101,1,Dial(SIP/101,20)
exten => 101,n,Voicemail(101 at default)
;This automatically
2008 Oct 19
6
adding a second extension
I'm trying to add a second extension to my setup. The second device is
able to successfully connect to the Asterisk server. I am unable to
contact extension 101 from 102 and vise-versa. Also are my context
setup logically or is there a better fashion to organize them? My
error is at the bottom.
Here is the extension.conf
[default]
;
; By default we include the demo. In a production system,
2008 Oct 04
5
Vitelity Asterisk configuration help
I have a Asterisk server setup and I am able to connect to the server
using a soft client 'x-lite' and call and leave a message on my second
extension 102. I have setup a Vitelity account and add what I believe
to be the correct information to my sip.conf and extension.conf. I
would like to setup incoming and outgoing calls with voicemail
support. I've searched all over but many of the
2003 Oct 24
4
Context restrictions
Can someone please explain what I am doing wrong here? I only want the
extensions listed in long-users to be able to access the longdistance
context.
If I do this, I get a congestion tone no matter what I dial. If I add a
[default] context and include => longdistance, then the local callers
can call the long distance number fine, which is not what I want, but I
still want long-users to be
2004 May 18
3
Free Softphone Recomendations
Does anyone have any recomendations for a free Windows softphone, SIP or IAX that supports the following features:
* Message Waiting Indicator
* Consultative Transfers
* Speed Dials
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2008 Oct 10
2
Configuring Bandwidth.com SIP trunks to prevent one-way audio
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
2004 May 20
4
x100p card + dailing out
I think I have it configured properly. ztcfg -vv shows it as channel 1 and
zttool shows it as OK. But I can't dial out.
When I try, it shows it arrive in teh right stack, but then issues the
following errors:
channel.c:1676 ast_request: No channel type registered for '{PSTN-1}'
app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}'
= = Everyone is busy at
2005 Sep 29
4
Calling voicemail from external phone.
Hey.
How would I set up my dialplan if a user wants to call its voicemail
from an external phone?
I'm thinking of getting the user to enter its mailbox number.
Something like this:
1. User calls the dedicated voicemail number.
2. Phone prompts for mailbox number.
3. Voicemail(${mailboxnr}@context)
Thanks.
2009 Aug 05
2
original & reformat extension
Question:
Naturally there are times when need to I reformat an extension in a context as such:
;Reformat add CC1
exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1)
-or-
;Reformat 011 with with +CC
exten => _011X. ,1,Goto(+${EXTEN:3},1)
It's a helpful trick, BUT there are times when I want to send the call to another context in its original un-reformatted state. Naturally the ${EXTEN}
2013 Apr 18
5
ODBC dialplan looping problem
All,
Thank you in advance for any help.
I have a customer in need of a conferencing system. A requirement is for
users to each have their own PIN for the same bridge.
So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC
connector to parse the table.
Asterisk is connected and reads the rows as expected. The problem is that
if a user enters a PIN that is NOT in the table,
2010 Dec 22
1
Simplifying dial-plan
Is there a way to include:
_NXXNXXXXXX
_NXXXXXX
_011.
_911
into my current plan:
2009 Jun 13
1
1.6.0.10: core restart on ReceiveFax()
For our internal fax machines, I'm checking if the faxes are going to
branch offices. If they are, I want to capture and email them to the
branches. I've set up extension 8447 to test this.
A fax machines is connected via an SPA 2102 on 173. Any calls from 173
are sent to:
[outbound-fax]
exten => 8447,1,Answer()
exten => 8447,n,GoSub(Capture-Fax,s,1)
exten
2004 May 24
2
Newbie extensions.conf I need to include [SMS] context.
I want to include a new context in my exensions.conf
I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan
and I can sort of follow it?!
I have a context [local] that I know zapata.conf points to, I have edited
extensions.conf and put in my phone, sip and iax extensions. I want to add
an sms context.
I understand that all calls go through my [local] context and I have
2010 Aug 30
2
help with dialplan
Todd
How do you have the context in the phones sip configs set?
Bryant
From: "Todd Reese" treese65 at gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk
2006 Dec 21
2
Insert 1+areacode for VOIP calls
Greetings,
Currently my asterisk box is using Voicepulse. It works fine with the
exception that people need to enter the 1+area code for local calls.
I'd like to get around this if possible. The following is what I have
in my extensions.conf..
exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=6162997590)
exten => _1NXXNXXXXXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN})
exten
2010 May 27
2
Pattern matching - how to ignore numbers after 10 digits
All:
Yesterday I discovered something interesting. I dialed 1800ANCESTRY
from the asterisk system I am testing and got the number doesn't exist
message. I then dialed the same number from our old system and it went
through.
I realized that the "Y" in ancestry made the number too long, and went
back to my dialplan.
How do I ignore numbers that are too long? Obviously,
2006 Jan 27
7
AAH out bound routing problem
Hi all
I have installed AAH 2.2 in my P4 PC
following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
and made as per the guide says
and downloaded SJ Phone, and registered user
and when i try to dial the 19197543700
i get message that, all circuits are busy now, please try your call later
and when i see in the console i get this mesage
any help
Called easycall/19197543700
2004 Jan 19
3
Residential services
Hi folks,
The obligatory newbie disclaimer:
"Hi, I'm new to Asterisk and I have a couple questions..."
OK, now that that's over with:
I've just started working for a small CLEC, and I'm trying to sell * to
my boss as a replacement for some expensive/inflexible/closed-source
software he's been using to provide residential dialtone with for a
couple years now.
2005 Oct 11
1
call to a particular 800 numbernevershowsanswered on Zap channel
> Watch the output of 'pri debug span 1' on the Asterisk server while
> placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468)
> might be relevant.
Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:
2004 May 07
2
quadBRI & ISDN telephone
Hello,
We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a
ISDN telephone to this nothings happen.
What can I do?
My config files are this:
Zaptel.conf:
loadzone=es
defaultzone=es
# qozap span definitions
# most of the values should be bogus because we are not really zaptel
span=1,1,3,ccs,ami
span=2,1,3,ccs,ami
span=3,1,3,ccs,ami
span=4,1,3,ccs,ami
bchan=1,2
dchan=3