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thodgen
06-24-2012, 06:18 PM
I have a scenario where T1 Gateway is forwarding all calls to another PBX. When call is answered by the AA, and transfered to an extension that does not have Voicemail configured, after the incoming call timesout, the call is returned to the Quadro T1 Gateway, a recording is played that states "number dialed is temporarily unavailable, now you can make a call". After this recording, pressing the number 9, and a dial string places a call to that dial string. DOH!

I see in the log file that it's not resolving a domain name, but fear there is a security risk I need to address here before I open myself to an attack. I am assuming I need to setup a filter so these callls are made available to callers. I would love to send these calls that time out to a reorder or failure message, rather than new dial tone. Pertinent part of the log is below -

.809 SetTimer: line #64, type - FaintHopeHypnosis, interval - 8 sec, callID - 65188682906847062.
.811 RequestSIPCallNatMapping: SendCallMake, remote addr - mydomain.com, self data - [10.98.34.3:6064], callID - 65188682906847062.
15:46:28.829 <WARNING> ProcessNetworkTask: type - ResolveHostName, cannot get IP address of the host mydomain.com
.845 ProcessSIPCallNatReply: SendCallMake, result - TRUE, mapped data - [0.0.0.0(10.98.34.3):0(6064)], callID - 65188682906847062.
15:46:29.197 OnSIPCallRinging: line #64[255](e:system), callID - 65188682906847062.
.198 TS: line #64(e:system) - OffHookRinging.
.200 KillTimer: line #64, type - FaintHopeHypnosis, callID - 65188682906847062.
.200 StartSoundPlayback: line #64[0], type - RingTone, callID - 65188584122918902.
.871 <WARNING> ProcessNetworkTask: type - ResolveHostName, cannot get IP address of the host mydomain.com
15:46:30.880 <WARNING> ProcessNetworkTask: type - ResolveHostName, cannot get IP address of the host mydomain.com
15:46:31.890 <WARNING> ProcessNetworkTask: type - ResolveHostName, cannot get IP address of the host mydomain.com
15:47:28.943 EndNatSession: callID - 65188682906847062.
.944 OnSIPCallReportError: line #64[255](e:system), got error code "Timeout Occurred", callID - 65188682906847062.
.946 SendSIPCallReferResponse: result - TimeoutOccurred, callID - 65188584122957695.
.948 EraseSIPCall: callID - 65188584122957695.
.948 ExternalCallClose: callID - 65188584122957695.
.951 EndNatSession: callID - 65188584122957695.
.954 CreateFileEndpoint: {MG0}<FILE/1002> - 65188584122918902.
.955 StopSoundPlayback: line #64[0], type - RingTone, callID - 65188584122918902.
.956 StartSoundPlayback: line #64[0], type - TemporaryUnavailable, callID - 65188584122918902.
.957 StartSoundPlayback: line #64[0], {MG0}65188584122918902<4>(TOS 0xB8), FILE[/telephony/sysmessages/C/tempunavail.wav, PC: 1, a: PCMU(0)] -> ISDN[0/0].
15:47:31.420 ProcessConnectionClosed(FileOver): {MG0}65188584122918902<4>, restoring AA, line #64[0](e:system).
.421 EraseSIPCall: callID - 65188682906847062.
.421 EndMediaSession(*): {MG0} - 65188682906847062.
.422 EraseEndpoint: {MG0}<RTP/1001>[10.98.34.3:6064] - 65188682906847062.
.423 TidyMediaStatisticRequest: MG{0}, callID - 65188584122957695, sessID - 65188584122957695, remain records - 1.
.423 EndMediaSession: {MG0} - 65188682906847062, ownerID - 65188682906847062, requesting RTP statistics.
.426 <<<<<<<<<<<< Handle Map >>>>>>>>>>>>
C:
#64, ID - 65188584122918902.->ISDN(U)
S:
{MG0} - 65188584122918902.
<<<<<<<<<<<<<<<<<<>>>>>>>>>>>>>>>>>>
.427 TS: line #64(e:system) - OffHookInCall.
.428 RestoreAttendant: line #64(e:system), ISDN(U) session, callID - 65188584122918902.
.431 MapDetectionEndpoint: DTMF detection ISDN[0/0]{MG0}<0>, callID - 65188584122918902.
.434 OpenAttendantStream: line #64[0], {MG0}65188584122918902<4>, FILE[/telephony/sysmessages/C/attdial.wav, PC: 5, a: PCMU(0)] -> ISDN[0/0].
.459 ReceiveMediaStatistics: there is no CDR yet, MG{0}, sessID - 65188682906847062
-+- [L0.0.0.0:0 R0.0.0.0:0] TXP:0, TXC:N/A, TXPS:0, RXP:0, RXC:N/A, RXPS:0, RXL:0, RXJ:0, RXMD:0
15:47:34.570 SendButtonToAttendant: button - 9, callID - 65188584122918902.
.571 CloseAttendantStream: line #64[0], {MG0}65188584122918902<4>.
.572 TS: line #64(e:system) - OffHookDialing.
.573 StartSoundPlayback: line #64[0], type - ExtraDialTone, callID - 65188584122918902.
.575 SetTimer: line #64, type - Dialing, interval - 15 sec, callID - 65188584122918902.
15:47:36.129 SendButtonToAttendant: button - 2, callID - 65188584122918902.
.

Vahan
07-04-2012, 03:43 PM
Hello,

Typically callers would not be pushed back to the attendant if a call trough attendant fails. I know only one case when caller would be pushed back, it is if caller uses call relay(feature code *2) to place the call.
Do your callers use call relay by dialing *2 when on attendant? if so, they can use *1 instead, the difference is that they will not be pushed back to auto attendant if thier call fails.
Other way to go is to not use call relay at all.

Thanks.