search for: busytone

...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}; &g...

...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) -------------- next part -------------- An HTML attachment was scrubb...

...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,437,10...

...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....

...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 TraceFl...

...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)....

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

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 Cau...

...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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060411/209df3a9/attachment.htm

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