[asterisk-bugs] [Asterisk 0016292]: [patch] DTMF Not Recognized with Exchange UM
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Dec 1 15:04:40 CST 2009
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=16292
======================================================================
Reported By: rsw686
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 16292
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: major
Priority: normal
Status: ready for testing
Asterisk Version: 1.6.1.10
JIRA: SWP-463
Regression: Yes
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-11-20 09:53 CST
Last Modified: 2009-12-01 15:04 CST
======================================================================
Summary: [patch] DTMF Not Recognized with Exchange UM
Description:
I just upgraded my Asterisk 1.6.1.4 version to 1.6.1.10 and the DTMF tones
are not being recognized by the Exchange UM server. I tested out the RC
release and found that it worked in 1.6.1.7-rc1 and broke in 1.6.1.7-rc2.
What happens is Exchange UM will say to enter your mailbox number. I enter
8532 and it will say that 8 is not a valid entry. It appears Asterisk is
only detecting the first DTMF tone.
======================================================================
----------------------------------------------------------------------
(0114478) rsw686 (reporter) - 2009-12-01 15:04
https://issues.asterisk.org/view.php?id=16292#c114478
----------------------------------------------------------------------
Here is the debug output from the console. It looks like the root of the
problem is that when the channel is forwarded the constantssrc bit isn't
copied. You can see where the constantssrc flag is set just before the
forward, but not set afterwards. I'm not sure where in the code this should
be fixed.
== Using SIP RTP CoS mark 5
[Dec 1 15:55:14] DEBUG[11430]: chan_sip.c:4626 do_setnat: Setting NAT on
RTP to Off
[Dec 1 15:55:14] DEBUG[11430]: chan_sip.c:6830 sip_alloc: Allocating new
SIP dialog for ZjFmYjA3Mzg1MDc1NDIxMTdhYTJiNmY0NjcxYmUzNWQ. - INVITE (With
RTP)
[Dec 1 15:55:14] DEBUG[11430]: chan_sip.c:4626 do_setnat: Setting NAT on
RTP to Off
[Dec 1 15:55:14] DEBUG[11430]: chan_sip.c:3174 __sip_xmit: Trying to put
'SIP/2.0 401' onto UDP socket destined for 10.9.5.63:64274
[Dec 1 15:55:14] DEBUG[11430]: chan_sip.c:3709 __sip_ack: Stopping
retransmission on 'ZjFmYjA3Mzg1MDc1NDIxMTdhYTJiNmY0NjcxYmUzNWQ.' of
Response 1: Match Found
[Dec 1 15:55:14] DEBUG[11430]: chan_sip.c:4626 do_setnat: Setting NAT on
RTP to Off
ast_rtp_set_constantssrc: setting flag
[Dec 1 15:55:14] DEBUG[11430]: chan_sip.c:19021 handle_request_invite:
Checking SIP call limits for device 38678
[Dec 1 15:55:14] DEBUG[11430]: chan_sip.c:3174 __sip_xmit: Trying to put
'SIP/2.0 100' onto UDP socket destined for 10.9.5.63:64274
[Dec 1 15:55:14] DEBUG[22941]: pbx.c:3200 pbx_extension_helper: Launching
'Answer'
-- Executing [3300 at default:1] Answer("SIP/38678-0000000f", "") in new
stack
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:5772 sip_answer: SIP answering
channel: SIP/38678-0000000f
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:9736
transmit_response_with_sdp: Setting framing from config on incoming call
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:9405 add_sdp: ** Our
capability: 0x4 (ulaw) Video flag: True Text flag: True
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:9406 add_sdp: ** Our prefcodec:
0x0 (nothing)
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:3174 __sip_xmit: Trying to put
'SIP/2.0 200' onto UDP socket destined for 10.9.5.63:64274
[Dec 1 15:55:14] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 132 bytes
[Dec 1 15:55:14] DEBUG[22941]: pbx.c:3200 pbx_extension_helper: Launching
'Dial'
-- Executing [3300 at default:2] Dial("SIP/38678-0000000f",
"SIP/SIP_VM/3300,,TTr") in new stack
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:22016 sip_request_call: Asked
to create a SIP channel with formats: 0x4 (ulaw)
== Using SIP RTP CoS mark 5
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:6830 sip_alloc: Allocating new
SIP dialog for 75cf3053617b088f658353b81dda2aea at 127.0.0.1 - INVITE (With
RTP)
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:4626 do_setnat: Setting NAT on
RTP to Off
ast_rtp_set_constantssrc: setting flag
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable SIPURI.
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:5030 sip_call: Outgoing Call
for 3300
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:9405 add_sdp: ** Our
capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False Text flag:
False
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:9406 add_sdp: ** Our prefcodec:
0x4 (ulaw)
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:2897 initialize_initreq:
Initializing initreq for method INVITE - callid
7641df9072b9c28a41b566c06bcbd11f at 10.9.1.121
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:3174 __sip_xmit: Trying to put
'INVITE sip:' onto TCP socket destined for 10.9.1.13:5060
-- Called SIP_VM/3300
[Dec 1 15:55:14] DEBUG[22941]: channel.c:3178 ast_indicate_data: Driver
for channel 'SIP/38678-0000000f' does not support indication 3, emulating
it
[Dec 1 15:55:14] DEBUG[22941]: channel.c:3655 set_format: Set channel
SIP/38678-0000000f to write format slin
[Dec 1 15:55:14] DEBUG[22941]: channel.c:2377 ast_settimeout: Scheduling
timer at (50 requested / 50 actual) timer ticks per second
[Dec 1 15:55:14] DEBUG[22941]: channel.c:2490 ast_read_generator_actions:
Generator got voice, switching to phase locked mode
[Dec 1 15:55:14] DEBUG[22941]: channel.c:2377 ast_settimeout: Scheduling
timer at (0 requested / 0 actual) timer ticks per second
[Dec 1 15:55:14] DEBUG[22941]: rtp.c:3806 ast_rtp_write: Ooh, format
changed from unknown to ulaw
[Dec 1 15:55:14] DEBUG[22941]: rtp.c:3822 ast_rtp_write: Created
smoother: format: 4 ms: 20 len: 160
-- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13
-- Now forwarding SIP/38678-0000000f to
'SIP/3300::::TCP at 10.9.1.13:5067' (thanks to SIP/SIP_VM-00000010)
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:22016 sip_request_call: Asked
to create a SIP channel with formats: 0x4 (ulaw)
== Using SIP RTP CoS mark 5
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:6830 sip_alloc: Allocating new
SIP dialog for 3f58a80332343f772b91ca9b32f5cb55 at 127.0.0.1 - INVITE (With
RTP)
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:4626 do_setnat: Setting NAT on
RTP to Off
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Dec 1 15:55:14] DEBUG[22941]: channel.c:4262
ast_channel_inherit_variables: Not copying variable SIPURI.
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:5030 sip_call: Outgoing Call
for 3300
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:9405 add_sdp: ** Our
capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False Text flag:
False
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:9406 add_sdp: ** Our prefcodec:
0x4 (ulaw)
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:2897 initialize_initreq:
Initializing initreq for method INVITE - callid
60567a2b593d6fb83e2e3d2b769db223 at 10.9.1.121
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:3174 __sip_xmit: Trying to put
'INVITE sip:' onto TCP socket destined for 10.9.1.13:5067
[Dec 1 15:55:14] DEBUG[11450]: chan_sip.c:3174 __sip_xmit: Trying to put
'ACK sip:330' onto TCP socket destined for 10.9.1.13:5060
[Dec 1 15:55:14] DEBUG[22941]: channel.c:1711 ast_hangup: Hanging up
channel 'SIP/SIP_VM-00000010'
[Dec 1 15:55:14] DEBUG[22941]: chan_sip.c:5583 sip_hangup: Hangup call
SIP/SIP_VM-00000010, SIP callid
7641df9072b9c28a41b566c06bcbd11f at 10.9.1.121
-- SIP/10.9.1.13:5067-00000011 is ringing
[Dec 1 15:55:14] DEBUG[22941]: channel.c:3178 ast_indicate_data: Driver
for channel 'SIP/38678-0000000f' does not support indication 3, emulating
it
[Dec 1 15:55:14] DEBUG[22941]: channel.c:3655 set_format: Set channel
SIP/38678-0000000f to write format ulaw
[Dec 1 15:55:14] DEBUG[22941]: channel.c:3655 set_format: Set channel
SIP/38678-0000000f to write format slin
[Dec 1 15:55:14] DEBUG[22941]: channel.c:2377 ast_settimeout: Scheduling
timer at (50 requested / 50 actual) timer ticks per second
[Dec 1 15:55:14] DEBUG[22941]: channel.c:2490 ast_read_generator_actions:
Generator got voice, switching to phase locked mode
[Dec 1 15:55:14] DEBUG[22941]: channel.c:2377 ast_settimeout: Scheduling
timer at (0 requested / 0 actual) timer ticks per second
[Dec 1 15:55:14] DEBUG[11430]: chan_sip.c:3709 __sip_ack: Stopping
retransmission on 'ZjFmYjA3Mzg1MDc1NDIxMTdhYTJiNmY0NjcxYmUzNWQ.' of
Response 2: Match Found
[Dec 1 15:55:14] DEBUG[22926]: chan_sip.c:3174 __sip_xmit: Trying to put
'ACK sip:dia' onto TCP socket destined for 10.9.1.13:5067
-- SIP/10.9.1.13:5067-00000011 answered SIP/38678-0000000f
[Dec 1 15:55:14] DEBUG[22941]: channel.c:3655 set_format: Set channel
SIP/38678-0000000f to write format ulaw
[Dec 1 15:55:14] DEBUG[22941]: channel.c:2377 ast_settimeout: Scheduling
timer at (0 requested / 0 actual) timer ticks per second
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1553236498, new ssrc is 1156920430,
constantssrc is 0
[Dec 1 15:55:14] DEBUG[22941]: rtp.c:3806 ast_rtp_write: Ooh, format
changed from unknown to ulaw
[Dec 1 15:55:14] DEBUG[22941]: rtp.c:3822 ast_rtp_write: Created
smoother: format: 4 ms: 20 len: 160
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
[Dec 1 15:55:16] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 28 bytes
[Dec 1 15:55:16] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 104 bytes
[Dec 1 15:55:17] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 176 bytes
[Dec 1 15:55:17] DEBUG[22941]: rtp.c:886 send_dtmf: Sending dtmf: 56 (8),
at 10.9.5.63
[Dec 1 15:55:17] DEBUG[22941]: channel.c:4854 ast_generic_bridge: Got
DTMF begin on channel (SIP/38678-0000000f)
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1156920430, new ssrc is 276804749,
constantssrc is 0
[Dec 1 15:55:17] DEBUG[22941]: channel.c:5227 ast_channel_bridge: Bridge
stops bridging channels SIP/38678-0000000f and SIP/10.9.1.13:5067-00000011
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 276804749, new ssrc is 53644514, constantssrc
is 0
[Dec 1 15:55:17] DEBUG[22941]: rtp.c:886 send_dtmf: Sending dtmf: 56 (8),
at 10.9.5.63
[Dec 1 15:55:17] DEBUG[22941]: channel.c:4854 ast_generic_bridge: Got
DTMF end on channel (SIP/38678-0000000f)
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 53644514, new ssrc is 1602003665, constantssrc
is 0
[Dec 1 15:55:17] DEBUG[22941]: channel.c:5227 ast_channel_bridge: Bridge
stops bridging channels SIP/38678-0000000f and SIP/10.9.1.13:5067-00000011
sip_senddigit_begin: caller id: 3300
ast_rtp_set_constantssrc: setting flag
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:17] DEBUG[22941]: rtp.c:886 send_dtmf: Sending dtmf: 54 (6),
at 10.9.5.63
[Dec 1 15:55:17] DEBUG[22941]: channel.c:4854 ast_generic_bridge: Got
DTMF begin on channel (SIP/38678-0000000f)
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:17] DEBUG[22941]: channel.c:5227 ast_channel_bridge: Bridge
stops bridging channels SIP/38678-0000000f and SIP/10.9.1.13:5067-00000011
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:17] DEBUG[22941]: rtp.c:886 send_dtmf: Sending dtmf: 54 (6),
at 10.9.5.63
[Dec 1 15:55:17] DEBUG[22941]: channel.c:4854 ast_generic_bridge: Got
DTMF end on channel (SIP/38678-0000000f)
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:17] DEBUG[22941]: channel.c:5227 ast_channel_bridge: Bridge
stops bridging channels SIP/38678-0000000f and SIP/10.9.1.13:5067-00000011
sip_senddigit_begin: caller id: 3300
ast_rtp_set_constantssrc: setting flag
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:18] DEBUG[22941]: rtp.c:886 send_dtmf: Sending dtmf: 55 (7),
at 10.9.5.63
[Dec 1 15:55:18] DEBUG[22941]: channel.c:4854 ast_generic_bridge: Got
DTMF begin on channel (SIP/38678-0000000f)
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:18] DEBUG[22941]: channel.c:5227 ast_channel_bridge: Bridge
stops bridging channels SIP/38678-0000000f and SIP/10.9.1.13:5067-00000011
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:18] DEBUG[22941]: rtp.c:886 send_dtmf: Sending dtmf: 55 (7),
at 10.9.5.63
[Dec 1 15:55:18] DEBUG[22941]: channel.c:4854 ast_generic_bridge: Got
DTMF end on channel (SIP/38678-0000000f)
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:18] DEBUG[22941]: channel.c:5227 ast_channel_bridge: Bridge
stops bridging channels SIP/38678-0000000f and SIP/10.9.1.13:5067-00000011
sip_senddigit_begin: caller id: 3300
ast_rtp_set_constantssrc: setting flag
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:18] DEBUG[22941]: rtp.c:886 send_dtmf: Sending dtmf: 56 (8),
at 10.9.5.63
[Dec 1 15:55:18] DEBUG[22941]: channel.c:4854 ast_generic_bridge: Got
DTMF begin on channel (SIP/38678-0000000f)
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:18] DEBUG[22941]: channel.c:5227 ast_channel_bridge: Bridge
stops bridging channels SIP/38678-0000000f and SIP/10.9.1.13:5067-00000011
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:18] DEBUG[22941]: rtp.c:886 send_dtmf: Sending dtmf: 56 (8),
at 10.9.5.63
[Dec 1 15:55:18] DEBUG[22941]: channel.c:4854 ast_generic_bridge: Got
DTMF end on channel (SIP/38678-0000000f)
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:18] DEBUG[22941]: channel.c:5227 ast_channel_bridge: Bridge
stops bridging channels SIP/38678-0000000f and SIP/10.9.1.13:5067-00000011
sip_senddigit_begin: caller id: 3300
ast_rtp_set_constantssrc: setting flag
ast_rtp_new_source: ssrc is 49978530, new ssrc is 49978530, constantssrc
is 1
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:19] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 28 bytes
[Dec 1 15:55:19] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 128 bytes
[Dec 1 15:55:19] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 28 bytes
[Dec 1 15:55:19] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 104 bytes
[Dec 1 15:55:20] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 28 bytes
[Dec 1 15:55:20] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 104 bytes
[Dec 1 15:55:20] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 176 bytes
[Dec 1 15:55:20] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 28 bytes
[Dec 1 15:55:20] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 104 bytes
[Dec 1 15:55:20] DEBUG[22941]: rtp.c:1233 ast_rtcp_read: Got RTCP report
of 160 bytes
[Dec 1 15:55:20] DEBUG[11430]: chan_sip.c:3174 __sip_xmit: Trying to put
'SIP/2.0 200' onto UDP socket destined for 10.9.5.63:64274
[Dec 1 15:55:20] DEBUG[22941]: channel.c:4803 ast_generic_bridge: Didn't
get a frame from channel: SIP/38678-0000000f
ast_rtp_new_source: ssrc is 1602003665, new ssrc is 1602003665,
constantssrc is 1
[Dec 1 15:55:20] DEBUG[22941]: channel.c:5227 ast_channel_bridge: Bridge
stops bridging channels SIP/38678-0000000f and SIP/10.9.1.13:5067-00000011
[Dec 1 15:55:20] DEBUG[22941]: channel.c:1711 ast_hangup: Hanging up
channel 'SIP/10.9.1.13:5067-00000011'
[Dec 1 15:55:20] DEBUG[22941]: chan_sip.c:5583 sip_hangup: Hangup call
SIP/10.9.1.13:5067-00000011, SIP callid
60567a2b593d6fb83e2e3d2b769db223 at 10.9.1.121
[Dec 1 15:55:20] DEBUG[22941]: chan_sip.c:3174 __sip_xmit: Trying to put
'BYE sip:dia' onto TCP socket destined for 10.9.1.13:5067
[Dec 1 15:55:20] DEBUG[22941]: rtp.c:2113 ast_rtp_early_bridge: Channel
'<unspecified>' has no RTP, not doing anything
[Dec 1 15:55:20] DEBUG[22941]: app_dial.c:2113 dial_exec_full: Exiting
with DIALSTATUS=ANSWER.
[Dec 1 15:55:20] DEBUG[22941]: pbx.c:3806 __ast_pbx_run: Spawn extension
(default,3300,2) exited non-zero on 'SIP/38678-0000000f'
== Spawn extension (default, 3300, 2) exited non-zero on
'SIP/38678-0000000f'
[Dec 1 15:55:20] DEBUG[22941]: channel.c:1606 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/38678-0000000f'
[Dec 1 15:55:20] DEBUG[22941]: channel.c:1711 ast_hangup: Hanging up
channel 'SIP/38678-0000000f'
[Dec 1 15:55:20] DEBUG[22941]: chan_sip.c:5583 sip_hangup: Hangup call
SIP/38678-0000000f, SIP callid
ZjFmYjA3Mzg1MDc1NDIxMTdhYTJiNmY0NjcxYmUzNWQ.
asterisk*CLI>
Issue History
Date Modified Username Field Change
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2009-12-01 15:04 rsw686 Note Added: 0114478
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