Displaying 10 results from an estimated 10 matches for "busyton".
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busytone
2003 Oct 23
0
FW: Voicetronix
...c
Index: chan_vpb.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_vpb.c,v
retrieving revision 1.9
diff -r1.9 chan_vpb.c
100,102c100,102
< static VPB_TONE Dialtone = {440, 440, 440, 0, 0, 0, 5000, 0 };
< static VPB_TONE Busytone = {440, 0, 0, 0, -100, -100, 500,
500};
< static VPB_TONE Ringbacktone = {440, 0, 0, 0, -100, -100, 100,
100};
---
> static VPB_TONE Dialtone = {440, 440, 440, -1, -1, -1, 5000, 0
};
> static VPB_TONE Busytone = {440, 0, 0, -1, -100, -100,
500, 500};
&...
2014 Sep 01
1
SIP Calls Not Working
...text=exten-100
[101]
type=friend
username=101
secret=101
host=dynamic
port=5060
dtmfmode=rfc2833
fromdomain=dynamic
nat=no
canreinvite=false
context=exten-101
The extensions.conf contains
========================
[exten-100]
exten => 101,1,Dial(SIP/101)
;exten => echo,1,Echo()
;exten => busytone,1,Playback(moh)
;exten => 101,n,Hangup()
exten => 100,1,Answer()
exten => 100,n,Hangup()
[exten-101]
exten => 101,1,Answer()
exten => 101,n,Hangup()
exten => 100,1,Dial(SIP/100)
;exten => _x.,1,Playback(moh)
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2004 Apr 26
1
troubles working with Voicetronix Openswitch12
...t;s,1,Wait
exten =>s,2,Answer
exten =>s,3,Background(demo-congrats)
exten =>s,4,Background(vm-repeat)
exten =>s,4,Hangup
i have also change the DBlevels to -5 and -12 repectively in the
/channels/chan_vpb.c i have
static VPB_TONE Dialtone = 440,440,440,-12,-12,12,5000,0
static VPB_TONE Busytone= 440,0,0,-12,-12,-12,500,500
static VPB_TONE Ringback= 440,0,0,-12,-12,-12,100,100
with no change.
i have also tried to use the tonetrain program in the VPB doc, running
this tests on the ports i get
export VPB_TONE=5,P6,312,100,2000,578,100,10,10 ; for the busytone
export VPB_TONE=5,C,437,1...
2011 May 10
2
About X100P and TDM400P analog card in China
...quot;The call has been
answered".
Actually it is still dialing and my mobile is ringing because I didn't
answer the call.. The music was played by ISP
4. whether I answered the call or refuse the call. No more prints on
asterisk console.
But on phone end, when I refuse the call, instead of busytone, I hear the
voice "The phone you're dialing is busy now. Please try again later.".
So the whole thing is, during the whole call process, only before dialing,
we can hear the phone tone, for all other time, Dialing, refused, the ISP
will play music/voice instead of providing the tone....
2005 May 10
3
MGCP : chan_mgcp.c:1509 find_subchannel
...rIdMethod: 0xc0019e60
DNS1IP: 0.0.0.0
DNS2IP: 0.0.0.0
Domain: .
NumTxFrames: 2
TOS: 0x000068b8
OpFlags: 0x00000002
VLANSetting: 0x0000002b
Polarity: 0x00000000
FXSInputLevel: 0
FXSOutputLevel: -4
SigTimer: 0x00000064
RingCadence: 2,4,25
DialTone: 2,31538,30831,1380,1740,1,0,0,1000,0,0
BusyTone: 2,30467,28959,1191,1513,0,4000,4000,0,0,0
ReorderTone: 2,30467,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0
RingBackTone: 2,30831,30467,1943,2111,0,16000,32000,0,0,0
CallWaitTone: 1,30831,0,5493,0,0,2400,2400,4800,0,0
AlertTone: 1,30467,0,5970,0,0,480,480,1920,0,0
NPrintf: 0.0.0.0.0
TraceF...
2004 Apr 23
1
newbie install problems
...h) files
for some codecs are missing. Is that ok?
2. Zaptel and Libpri did make and make install without noticeable warnings,
but when I try to dial out thru X100P card
(or best, as soon as asterisk initiate diallling with X100P) I get a kind
of "fast beep" from asterisk (like a rapid busytone) that
persists until both ends hangup. Any tip on what is going on? Is that normal ?
3. After a dial out (or dial in/answer) the X100P card hangs and do not
respond to any command until asterisk is finished (stop now/gracefully)
and the drivers are reload (rmmod and modprobe again a few times)....
2004 Oct 07
1
dial out
Good day all
I'm getting this error while trying to dial out on my asterisk server
using a openline4 card
"exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872"
Please Help me
2006 Apr 13
0
Hangupcause to handle Called party disconnect ? PSTN----E1----OldPBX---E1--Asterisk
Hi,
I've been debuging the call disconnection problem in our architecture:
PSTN---E1---OldPBX---E1---Asterisk
This is our problem:
-SIP user agent "A" calls a pstn phone "B".
-"B" hangs up the call.
-SIP user agent "A" starts listenning busytones... But the call still on.
(and being payed).
- Call only ends when it is correctly hanged up in the SIPphone.
I've been tracing the communications between the OldPBX (NETWORK) and
Asterisk (USER SIDE) and i found this:
M03 PROGRESS
I08 Cause
Coding Std=CCITT
Location=
Private net-remote
Ca...
2006 Apr 11
1
E1 Disconnection Asterisk behind an old PBX
...tone indications and
Asterisk doesn't hangsup the call automatically.
4- The call only gets finished when the Sip User Agent "A" hangs the call on
the SIP phone.
5- This way asterisk CDR could be reporting 30 minutes call duration , even
if i have 1 minute of speech and 29 minutes of busytone indications....
Could be the OldPBX that doesn't send the disconnect ?
Any tips?
Best regards,
Marco Mouta
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2003 Dec 16
2
Help! VoiceTronix Multi FXO/FXS Problem
Hello, Hacker
I install VoiceTronix OpenSwitch 12 port PCI Telephone Card,
and setting vpb.conf, extensions.conf following
My problem is:
When i dial to fxo(channel 9-12), it is ok,
but when i continue press exten 102, the channel crach with error messages
following
exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872
Do i ignore some setting for VoiceTronix OpenSwitch12