Displaying 20 results from an estimated 4000 matches similar to: "No ring tone on inbound call - but channel connects fine"
2010 Dec 15
2
Two asterisk servers, two different service providers
All:
I am looking to install another asterisk server in an office located in
a different part of the country.
I think I can configure the sip and extension conf files, so that the
internal phones at the two locations can call each other.
My question is this, how do I properly configure the sip file for a
different provider at the new location? Can I use a different register
statement for
2011 Feb 18
3
Assigning an extension to a roaming phone
Hi,
I'm trying to automatically have the dialplan assign an extension to a
roaming phone on my network.
I tried the following without success:
exten => 3001,1(readop),BackGround(beep)
exten => 3001,n,Read(digito,vm-youhave,3)
exten => 3001,n,SayDigits(${digito})
exten => 3001,n,Set(ROAM=${digito})
exten => 3001,n,Set(DB(roam/ext)=${digito})
exten =>
2005 Feb 04
3
No ring tone on Outgoing calls
Hi there i have some problems with some clients of my asterisk box, i
have some cases when a client tried to make a call and there is no
ring back only a silence and then the call hung up. I dont know why
this is happening. I have the following stable asterisk version:
CVS-v1-0-01/18/05-19:49:31
I did the an update a few days ago, the version that i had installed
before was:
2006 Oct 29
2
Incorrect Ring tone. Getting a US tone when it should be AU tone
For some reason Asterisk is producing a US ring tone when it should be
an Australian ring tone. I am using ztdummy and do not have any cards
installed. My configuration is as follows. I am using Trixbox 1.2.2.
Can someone please guide me into the right direction?
zaptel.conf
loadzone = au
defaultzone = au
zapata.conf
[channels]
language=au
indications.conf
[general]
2006 Mar 15
3
Double-ring tone
I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works
fine. Except that when I make an outbound call, I get a double-ring
sound. I also found that if the target number is engaged, I get a ring
sound and at the same time get a busy sound.
If I revert back to 7-4, there is no problem.
Anyone else had this, or any clues on how to fix it ? All of our other
phones are still on
2006 Jan 26
0
Aastra 9112i or 9133i Ring Tones
I've been looking at the two phones Aastra 9112i and 9133i for our offices
now. I was just curious if anyone could tell me about the ring tones.
Specifically:
1) How many ring tones do they come with?
2) Can you upload your own ring tones?
3) If so, what format and how?
(wav, mp3, gsm, etc. and TFTP, web management, flash, etc.)
Thanks for any help.
--
Tim DeBaillie 812-476-2721
2005 Jan 04
2
integrating with panasonic td-1232
Hi,
Anybody have an idea how to integrate * with a Panasonic td-1232?
We one at the main office, and are installing * in a branch office.
We'd like to be able to make calls from * extensions to Panasonic
extensions and the other way around.
Making outgoing calls from extensions one one side to lines on the other
would be nice too.
I can put another * machine at the main office, but what is
2007 Jan 30
1
No intercom splash tone?
Environment:
Asterisk 1.2.14, FreePBX 2.2.0, Aastra 480i IP telephones firmware
version 1.4.1.1077.
Problem:
Intercom feature: the dialed phone does not play the splash tone when
auto-answering an intercom call. Otherwise, intercom works perfectly.
Questions:
What is the extensions.conf syntax to trigger a splash tone in Asterisk
1.2.14 (from the documentation and posts I've found, it has
2007 Oct 18
1
Ring Groups
Here's what I'm looking to do....
exten => 10,1,Dial(SIP/1000&SIP/1001,15,wW)
exten => 10,2,Voicemail(u1000)
This should ring both phones and they should keep ringing for the
alloted time before moving on. However, it appears that if one of the
phones is Busy, it returns with a busy and moves on without really
ringing the second phone.
Short of checking if the channels are
2003 Apr 17
1
timeout music on hold or ring tone
Is any way to limit music on hold (or ringtones) to specified time ? I
need it to play it ~ for 7 seconds .
How to do this ?
in dial plan i have:
exten => _021XXXXXX,4,Dial,Zap/1/BYEXTENSION||r
when go to this extension it rings once!
and then asterisk say :
-- Zap/1-1 answered Modem[i4l]/ttyI0
and it stop ringing ;) becouse mean that other end is ringing :) ..
BUT when the other
2003 Sep 18
0
no ring tone analog Zap --> I4L
Hi all,
i have noticed that i can't hear a ring tone if i make a call from my TDM40B
to a chan_modem_i4l endpoint.
I had a look in the code from chan_modem_i4l and there is a function calling
"i4l_handle_escape" that gives a AST_CONTROL_RINGING frame back. But this
seems not work ...(or i4l is not signaling it ?)
Til now i have used the Dail app like
Dial, Zap/g1:XXXXXX|60|r
so it
2004 Jul 22
1
no incoming pstn ring tone
Hi All,
I recently upgraded from a very old stable to HEAD. For some reason,
incoming callers don't hear ring tones when calling in. Everything else
is working fine. Where should I look for a fix?
ISDN --> E100P --> asterisk --> sipphones.
Thanks
Johan
2004 Sep 01
0
Ring tone when busy in trunk scenario
Hi,
I have an asterisk box connected to a PRI LINE, some extensions are
trunked by IAX to another box that's connected via ISDN BRI to a PBX.
That's what's happening
call comes in via PRI to the first box and is sent to the other box
exten => _N.,1,Dial(IAX2/sip:pippo@trunk/*${EXTEN})
the other box rings a pbx (simulating an ISDN call from a BRI line)
exten =>
2006 Feb 20
0
SIP ATA gives no ring tone on IAX2 route
Hello everybody,
I have this problem where I can't get a ring tone when
SIP devices dial an IAX2 route. I get the ring tone
using IAX2 devices to dial the same route. The call
completes normally in both cases...
Facts:
- Asterisk 1.0.9
- The Dial command is within an AGI.
- ATA (grandstream) and firefly (SIP mode) would not give me the ring tone
at all
- Switching to a SIP route works ok
-
2006 May 02
1
SIP trunk ring tone
Hi,
I'm wondering what I need to change to get the "swedish" type ring on a
SIP-trunk. When I make an inbound call i still have the "US"-type of
ring on my SIP trunks. I need help on changing this.
However I've successfully changed this on the Zap interface for all
inbound calls.
Thanks in advance!
Regards,
Jan
2007 Jan 16
1
Ring tone too loud on IAX channel
Hi,
We are using MozIAX as a softphone with a USB headset and are making
outbound calls using IAX with ulaw encoding to our voip provider.
We're running asterisk 1.4
Users are complaining that the ring tone generated by asterisk is much
louder than the voice call once connected. They are having to turn the
volume down to avoid being deafened by the ring tone, but then have an
unacceptably
2008 Oct 26
1
Strange ring tone: Long-Short-Short
I'm using Linksys SPA3102 adapter and have a strange ring tone:
Long-Short-Short or Long-Long-Short-Short
Does anybody know which setting adjust this ring tone on SPA3102
Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab
--
#Joseph
2009 Apr 16
2
TDM2400P dial tone is not present on phones, but the phone ring with incoming calls
Hi,
I have a problem with TDM2400P card. The card is detected ok, I can make a call but only with pulse dialing (not tone dialing) without hear sounds from the headset. When I receive a call, I can to establish a communication, but without hear sounds from the headset. When I dial any phone key, I can hear dtmf tone.
I'm using Elastix 1.5.2. These are my configuration files:
2009 Jun 13
1
Dial with r option doesn't use 'ring' tone as defined in indications.conf
Hi,
Just noticed Asterisk is not playing 'ring' tone as defined in
indications.conf when Dial command is used with 'r' option.
For example:
[test]
exten => 123,1,PlayTones(ring)
exten => 123,n,Wait(5)
exten => 123,n,Playback(demo-congrats)
exten => 123,n,Hangup()
exten => 321,1,Dail(LOCAL/123 at test/n,60,r)
When I now dial with a SIP phone - 123 I can hear nice
2009 Aug 14
2
no ring tone
how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring.
Edit sip_nat.conf for proper NAT:
localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here)
externrefresh=10
fromdomain=DOMAIN.com (Set your external domain name here)
nat=yes
qualify=yes
canreinvite=no
Add extra codecs to