[asterisk-bugs] [Asterisk 0018129]: [patch] Oneway audio from SIP phone to FXS port after FXS port gets a CallWaiting pip

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18129 
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Reported By:                alecdavis
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18129
Category:                   Channels/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-2367 
Regression:                 No 
Reviewboard Link:           https://reviewboard.asterisk.org/r/978/ 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 290865 
Request Review:              
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Date Submitted:             2010-10-13 04:31 CDT
Last Modified:              2010-11-04 03:00 CDT
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Summary:                    [patch] Oneway audio from SIP phone to FXS port
after FXS port gets a CallWaiting pip
Description: 
Internal SIP phone initiates a call with FXS port on TDM800P.
FXS connected phone has to have FSK CIDCW support to fail, as it will send
back a DTMF 'A' or 'D' when it's ready to receive CallerID.
A normal phone with no CID never fails. 

External call comes in on FXO port, and attempts to ring FXS ports.
Call waiting beep is heard at FXS port, also no outbound audio to SIP
phone.

Then FXS port hook flashes, the FXO port is then connected to FXS as
expected.
But SIP device should hear MOH, but has dead air.

If the FXS port hook flashes to the SIP device, the FXO call then does
hear MOH.
Hook flash again back to FXO call, SIP hears nothing.  
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---------------------------------------------------------------------- 
 (0128599) alecdavis (manager) - 2010-11-04 03:00
 https://issues.asterisk.org/view.php?id=18129#c128599 
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<u><<<<<<<<<BT100 in conversation with FXS port>>>>>>>>>></u>
<u><<LAPTOP dials FXS PORT>></u>
  == Using SIP RTP CoS mark 5
    -- Executing [89 at phones:1] Dial("SIP/laptop-00000002", "DAHDI/35,60")
in new stack
[2010-11-04 20:44:25.647975] NOTICE[20179]: chan_dahdi.c:5038
save_conference: Disabled conferencing
[2010-11-04 20:44:25.648056] WARNING[20179]: chan_dahdi.c:1876
my_callwait: CIDCW + CAS
    -- Called 35
    -- DAHDI/35-2 is ringing
[2010-11-04 20:44:26.127122] NOTICE[20178]: chan_sip.c:6225
sip_senddigit_begin: SIP_DTMF_RFC2833 SIP/bt100black-00000001 D
<u><<BT100 has heard beginning of FXS phone's CAS DTMF 'D' and is now
begin sent DTMF RTP packets>></u>
[2010-11-04 20:44:26.207062] NOTICE[20178]: sig_analog.c:1576
analog_handle_dtmfup: Got some DTMF, but it's for the CAS
[2010-11-04 20:44:26.726994] NOTICE[20178]: chan_dahdi.c:5054
restore_conference: Restored conferencing
[2010-11-04 20:44:35.627104] NOTICE[20178]: chan_dahdi.c:5038
save_conference: Disabled conferencing
[2010-11-04 20:44:35.627187] WARNING[20178]: chan_dahdi.c:5155
dahdi_callwait: CIDCW no CAS
<u><<Hangup 2nd call>></u>
    -- Hanging up on 'DAHDI/35-2'
    -- Hungup 'DAHDI/35-2'
  == Spawn extension (phones, 89, 1) exited non-zero on
'SIP/laptop-00000002'
<u><<FXS port about to dial '1' to restore voice with BT100>></u>
[2010-11-04 20:44:51.407313] NOTICE[20178]: chan_sip.c:6225
sip_senddigit_begin: SIP_DTMF_RFC2833 SIP/bt100black-00000001 1
[2010-11-04 20:44:51.527518] NOTICE[20178]: chan_sip.c:6251
sip_senddigit_end: SIP_DTMF_RFC2833 SIP/bt100black-00000001 1

The previous approach to disable the DTMF detector, seems like it should
be disabled for all other parties, except for the FXS port which is being
called. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-11-04 03:00 alecdavis      Note Added: 0128599                          
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