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Summary:ASTERISK-10853: No audio midcall
Reporter:ahmed984 (ahmed984)Labels:
Date Opened:2007-11-21 12:39:11.000-0600Date Closed:2011-06-07 14:08:13
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/CodecHandling
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) codecs.txt
Description:Right in the middle of a call i stop receiving audio on both ends of the call.

It seems to happen at completely random times during the day but it does happen 2-3 times a day.

I have added the output from my asterisk console under additional information.


The weird thing is i have disabled G723 on the phones as well as asterisk, i got a sip trace of the call and i do not see G723 in the sdp data on either the INVITES or the OK response so i am not sure where the G723 bit comes from.

Any ideas what is happening here?
















****** ADDITIONAL INFORMATION ******

Nov 21 10:28:41 DEBUG[24366] chan_sip.c: Outgoing Call for 07950284506
Nov 21 10:28:41 VERBOSE[24366] logger.c: We're at 77.240.48.84 port 13902
Nov 21 10:28:41 VERBOSE[24366] logger.c: Adding codec 0x8 (alaw) to SDP
Nov 21 10:28:41 VERBOSE[24366] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Nov 21 10:28:41 VERBOSE[24366] logger.c: 13 headers, 10 lines
Nov 21 10:28:41 VERBOSE[24366] logger.c: Reliably Transmitting (no NAT) to 217.14.132.178:5060:
Nov 21 10:28:41 VERBOSE[24366] logger.c:     -- Called 07950284506@ORBtw3nty04
Nov 21 10:28:49 VERBOSE[24366] logger.c:     -- SIP/ORBtw3nty04-08330030 is making progress passing it to SIP/1032-b76d02b0
Nov 21 10:28:49 VERBOSE[24366] logger.c: We're at 77.240.48.84 port 16254
Nov 21 10:28:49 VERBOSE[24366] logger.c: Adding codec 0x8 (alaw) to SDP
Nov 21 10:28:49 VERBOSE[24366] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Nov 21 10:28:49 VERBOSE[24366] logger.c: Transmitting (NAT) to 90.152.3.98:2038:
Nov 21 10:28:49 DEBUG[24366] rtp.c: RTCP NAT: Got RTCP from other end. Now sending to address 90.152.3.98:16451
Nov 21 10:28:49 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:28:49 DEBUG[24366] rtp.c: RTP NAT: Got audio from other end. Now sending to address 90.152.3.98:16450
Nov 21 10:28:49 DEBUG[24366] rtp.c: Ooh, format changed from unknown to alaw
Nov 21 10:28:49 DEBUG[24366] rtp.c: Ooh, format changed from unknown to alaw
Nov 21 10:28:54 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:28:56 VERBOSE[24366] logger.c:     -- SIP/ORBtw3nty04-08330030 answered SIP/1032-b76d02b0
Nov 21 10:28:56 DEBUG[24366] chan_sip.c: sip_answer(SIP/1032-b76d02b0)
Nov 21 10:28:56 VERBOSE[24366] logger.c: We're at 77.240.48.84 port 16254
Nov 21 10:28:56 VERBOSE[24366] logger.c: Adding codec 0x8 (alaw) to SDP
Nov 21 10:28:56 VERBOSE[24366] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Nov 21 10:28:56 VERBOSE[24366] logger.c: Reliably Transmitting (NAT) to 90.152.3.98:2038:
Nov 21 10:28:56 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:01 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:06 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:11 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:16 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:21 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:26 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:31 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:36 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:41 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:46 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:51 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:29:56 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:01 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:06 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:11 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:16 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:21 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:26 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:31 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:36 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:41 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:46 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:51 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:30:56 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:01 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:06 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:11 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:16 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:21 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:26 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:31 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:36 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:41 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:46 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:51 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:31:56 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:01 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:06 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:11 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:16 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:21 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:26 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:31 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:36 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:41 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:46 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:51 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:32:56 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:01 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:06 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:11 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:16 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:21 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:26 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:31 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:36 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:41 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:46 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:51 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:56 DEBUG[24366] rtp.c: Got RTCP report of 52 bytes
Nov 21 10:33:57 DEBUG[24366] chan_sip.c: Oooh, format changed to 1
Nov 21 10:33:57 WARNING[24366] channel.c: Unable to find a codec translation path from g723 to alaw
Nov 21 10:33:57 WARNING[24366] channel.c: Unable to find a codec translation path from g723 to alaw
Nov 21 10:33:57 WARNING[24366] translate.c: No translator path from unknown to g723
Nov 21 10:33:57 WARNING[24366] file.c: Unable to translate to format wav49, source format g723
Nov 21 10:33:57 WARNING[24366] channel.c: Failed to write data to channel monitor read stream
Nov 21 10:33:57 WARNING[24366] chan_sip.c: Asked to transmit frame type 1, while native formats is 8 (read/write = 8/8)
Nov 21 10:33:57 WARNING[24366] channel.c: No path to translate from SIP/1032-b76d02b0(1) to SIP/ORBtw3nty04-08330030(8)
Nov 21 10:33:57 WARNING[24366] channel.c: Can't make SIP/1032-b76d02b0 and SIP/ORBtw3nty04-08330030 compatible
Nov 21 10:33:57 WARNING[24366] res_features.c: Bridge failed on channels SIP/1032-b76d02b0 and SIP/ORBtw3nty04-08330030
Nov 21 10:33:57 DEBUG[24366] channel.c: Hanging up channel 'SIP/ORBtw3nty04-08330030'
Nov 21 10:33:57 DEBUG[24366] chan_sip.c: Hangup call SIP/ORBtw3nty04-08330030, SIP callid 48f4b8ac061706a81b70555e29cc0bde@77.240.48.84)
Nov 21 10:33:57 DEBUG[24366] chan_sip.c: update_call_counter(07950284506) - decrement call limit counter
Nov 21 10:33:57 VERBOSE[24366] logger.c: Scheduling destruction of call '48f4b8ac061706a81b70555e29cc0bde@77.240.48.84' in 32000 ms
Nov 21 10:33:57 VERBOSE[24366] logger.c: set_destination: Parsing <sip:217.14.132.178;ftag=as4c8bae83;lr=on> for address/port to send to
Nov 21 10:33:57 VERBOSE[24366] logger.c: set_destination: set destination to 217.14.132.178, port 5060
Nov 21 10:33:57 VERBOSE[24366] logger.c: Reliably Transmitting (no NAT) to 217.14.132.178:5060:
Nov 21 10:33:57 DEBUG[24366] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Nov 21 10:33:57 VERBOSE[24366] logger.c:   ==  agi://127.0.0.1: Total time: 301
Nov 21 10:33:57 DEBUG[24366] channel.c: Soft-Hanging up channel 'SIP/1032-b76d02b0'
Nov 21 10:33:57 VERBOSE[24366] logger.c:     -- AGI Script agi://127.0.0.1 completed, returning 0
Nov 21 10:33:57 DEBUG[24366] pbx.c: Extension 07950284506, priority 1 returned normally even though call was hung up
Nov 21 10:33:57 VERBOSE[24366] logger.c:        > cdr_odbc: Query Successful!

Nov 21 10:33:57 DEBUG[24366] channel.c: Hanging up channel 'SIP/1032-b76d02b0'
Nov 21 10:33:57 DEBUG[24366] chan_sip.c: Hangup call SIP/1032-b76d02b0, SIP callid 3c361d45c0df-cx6esjxfwk4n@snom300-000413258E23)
Nov 21 10:33:57 DEBUG[24366] chan_sip.c: update_call_counter(1032) - decrement call limit counter
Nov 21 10:33:57 VERBOSE[24366] logger.c: Scheduling destruction of call '3c361d45c0df-cx6esjxfwk4n@snom300-000413258E23' in 32000 ms
Nov 21 10:33:57 VERBOSE[24366] logger.c: set_destination: Parsing <sip:1032@10.188.48.70:2051;line=ysqmkp8e> for address/port to send to
Nov 21 10:33:57 VERBOSE[24366] logger.c: set_destination: set destination to 10.188.48.70, port 2051
Nov 21 10:33:57 VERBOSE[24366] logger.c: Reliably Transmitting (NAT) to 90.152.3.98:2038:
Nov 21 10:33:57 DEBUG[24366] res_monitor.c: monitor executing ( nice -n 19 soxmix "/var/spool/asterisk/monitor/1195640921.21366-in.WAV" "/var/spool/asterisk/monitor/1195640921.21366-out.WAV" "/var/spool/asterisk/monitor/1195640921.21366.WAV"  && rm -f "/var/spool/asterisk/monitor/1195640921.21366-"* ) &
Nov 21 12:51:45 DEBUG[24366] app_queue.c: Device 'SIP/ORBtw3nty04' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Comments:By: ahmed984 (ahmed984) 2007-11-21 12:50:46.000-0600

I have uploaded the SIP trace i got from the call.

Thanks for all your help
By: Joshua C. Colp (jcolp) 2007-11-26 10:15:25.000-0600

We are no longer accepting 1.2 bugs on the bug tracker, if this is also happening with 1.4 feel free to reopen and attach an rtp debug.