Displaying 20 results from an estimated 700 matches similar to: "Incoming call #2 sent to VM immediately when already on phone with incoming."
2006 Mar 03
2
Does an entry in AstDB stay after reboot?
I set up a call forwarding script in extensions.conf which uses the
AstDB but I'm wondering if I reboot the server, will the entry in
AstDB still reside?
What the script does is when a call comes in, it check to see if there
is a null value or a call forward number. If null, it will call the
local office connections. If there is a number, it calls that. Now I
just need to know if I reboot
2005 Aug 09
0
Incoming call #2 sent to VM immediately whenalready on phone with incoming.
I have been wanting something similar. I paid some money for a busy
detect routine from newman telecom, but it is not yet done.
We'll see what happens.
Greg
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Min Hwan
Chang
Sent: Tuesday, August 09, 2005 6:57 PM
To: Asterisk-Users@lists.digium.com
Subject:
2006 Apr 15
3
FreePBX in Production systems?
Is anyone using FreePBX in production level systems because I'm just
wondering if its stable enough to use. Currently I'm editing my own *.conf
scripts but it sure would be nice if there were some sort of web interface
for other people to use. The only thing holding me back is the stability of
the FreePBX package... Any comments on this? Thanks in advance.
Regards,
Min Chang
2005 Sep 13
1
wctdm, issue w/outbound calls
Hi all,
I've been running Asterisk with a TDM400P for about 6months, no problems.
2 in/outgoing analog lines, one analog phone. Recently I was messing with
the XTEN client, got to finagling with things, and not knowing what was
wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was
testing various things, and found everything worked except outgoing calls.
So I checked
2006 Mar 28
2
Problems Configuring Cisco 12SP+
Hi,
After reading this valuable forum and the voip-info wiki and follow
all the steps , but my Cisco 12SP+ remains unregistered.
These are my config files:
skinny.conf
[general]
port = 2000 ; Port to bind to, default tcp/2000
bindaddr = 172.20.1.1 ; Address to bind to
dateFormat = D-M-Y ; M,D,Y in any order (5 chars max)
keepAlive = 120
languaje=es
allow = all
; disallow
2005 Mar 18
2
No sound when calling in from pstn
I am just starting out with * so bear with me please.
I have tdm400p with 4 fxo modules on it. When I call into the asterisk
box from my mobile, I can see the asterisk console picks the call up
and routes it to my computer with x-lite. There was no sound coming
from either - just silence. I then decided to route it directly to
voice mail to see if that would narrow the problem down, but it
2005 Jan 04
1
Displaying incoming e.164 callers number - how?
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
phone number on my phone. In the log is the following - which displayed
'601' on my phone. The caller was +886288097680 - am I getting the wrong
ClID because of my end or the caller end?
2005 Mar 22
1
No recorded messages
I have installed my first Asterisk implementation using the Asterisk@home
ISO. I am using the SJPhone software. Using the setup page, I have been able
to configure two extensions. Whne I dial from one to the other, the other
does not answer even though it is registered. Watching the log in the CLI, I
can see that recorded messages are being played;:
== No one is available to answer at this time
2007 Jan 30
3
musiconhold restarts for every extension
Hello!
I've upgraded from 1.2.9 to 1.2.14 recently but experience an
unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was
playing continuous on sequential extensions after a
timeout, it is restarted for every extension in 1.2.14:
;music starts
exten => 902,1,Dial(SIP/phone1@proxy.com|5|m(mymusic))
;music starts again
exten =>
2003 Oct 13
1
AGI solution to Grandstream BT102 call waiting problem
I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification.
PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)
2006 Apr 13
1
Ztmonitor shows RX is always on.
Details:
Asterisk 1.0.9
Zaptel 1.0
Dell P3 1ghz with X100P Clone
Location: India
This is an interesting issue where when I open up ZTMonitor, it shows the RX
as being on. It seems that Zaptel doesn't know to hang up the line so after
a couple of hours when the telecom cuts the line, everythign stops working.
Things I've tried include playing with the zaptel.conf, trying zaptel
v1.2(with
2005 Mar 17
3
Newbie can't dial out to pstn
Hi,
I have just put in a tdm400p with 4 fxo modules and am trying to dial
out from x-lite to dial my mobile phone just to test.
The output in the asterisk console is like this
Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack
-- Goto (mobile,61400039953,1)
-- Executing Goto("SIP/2002-239b", "localcall|61400039953|1") in
new
2004 May 12
1
Musical interruptions
Whilst on a call, I'm getting the following...
-- Started music on hold, class 'default', on SIP/phone3-a7d5
-- Playing 'pbx-transfer' (language 'en')
-- Unable to find extension '#' in context 'default'
-- Playing 'pbx-invalid' (language 'en')
ie - without anyone pushing keys - I hear the music on Hold - as does
the
2006 May 31
0
Incoming IAX going to wrong context
I have (more than 1) provider that I receive calls from using IAX, and I
have 2 IAX deskphones, all work fine except for some reason with 1
provider, when the call comes in, it doesn't match up with the
incomingcall context. (A bit worrying, since I don't want people to be
able to relay calls off me.)
in iax.conf I have:
[ipcomms]
type=user
nat=yes
dtmfmode=rfc2833
host=71.16.179.149
2004 Dec 04
5
BLOCKING incoming FAXES on voice line.
At time to time somebody is trying "their luck" and send me most likely
a junk fax on my voice line. During normal working hours is not a
problem I just pickup the line and hangup the call but after-hours my
voice mailbox is intercepting the call and recording those
"beeps" (waisting my CPU cycles).
Is there a way to block call / issue hangup command if the incoming call
is a
2004 May 14
3
SoftPhone to SoftPhone with No Voice
Hello
I Installed Asterisk on RedHat 9. I am currently try to configure minimum with
two softphone talking to each other over the LAN. I am using X-Lite softphones
from xten.com site. I defined 3 phones in sip.conf and also specifies in
extensions.conf file. I am able to ring other computer but there is no voice
exchange ( i can't hear any think except ring). Here is the portion of sip.conf
2005 Jun 22
3
indexing tables for dialing
Hello
I would like to know how can I manage to implement a table which translates
an extension number into a phone number. Let see an example:
If I dial an extension like 3021, Asterisk has to Dial an agent (our
employees) located at San Francisco using the following telephone number:
415 541 XXXX. If it does not work we can also use his/her mobile number.
We need to manage more than 180
2005 Jul 16
2
beginners question about extension context
Hi, all
I have couple of SIP phones and they are in [from-sip] context.
I also have an IAX2 phone. I have put this one in [iax-user] context.
I want to make calls between SIP and IAX2 phones. If I put them all in same
context all is fine, however when they are in different contexts they will
not call each other and I will get message (in * CLI) that particular
extension does not exist in a
2005 Mar 07
1
Custom Development
Hey guys,
I'm looking for a programming or Development Team/Company to do some custom
coding for Asterisk. What we need is not exactly simple. In fact, I'm not
sure the extent of the coding as far as technical terms go at all.
Currently we have a "call center" with 4 phones. There will be a total of 8
people using the phones. Obviously, no more than 4 people will use
2003 Jun 26
0
Kphone not working with Asterisk?
I'm trying to get two linux machines with kphone-3.11 two communicate with
each other over asterisk. I have them configured correctly on asterisk to use
sip channels, but when I call from one phone to the other I don't any voice
communication between the phones. According to the phones I'm connected, but
according to asterisk, I get the following message:
-- Executing