[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

Isaac Xiao isaac.x at kvbkunlun.com
Mon Jun 26 07:32:14 MST 2006


Hi,

 

Does any one experience that SIP phone to SIP phone (Polycom phone)
calls can't hear each other, but Monitor application records both end's
voices. It also happens in group pickup calls. Zap calls to queue (Local
channel) also experience this problem (sometimes, our SIP phone can't
hear any voice from incoming Zap calls when pickup, sometimes this
happens after 10-50 seconds' talk). It is weird.

 

Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Executing
Dial("Local/7188 at from-internal-7036,2", "SIP/7188|30|trWwT") in new
stack
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on RTP to 0
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on VRTP to 0
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Outgoing Call for 7188
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Called 7188
Jun 26 16:53:35 VERBOSE[8287] logger.c: --
Local/7188 at from-internal-7036,1 is ringing
Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'2978d85271fdaf3d0ac2e5b244e78773 at 192.168.2.66' Request 102: Found
Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'2978d85271fdaf3d0ac2e5b244e78773 at 192.168.2.66' Request 102: Found
Jun 26 16:53:35 DEBUG[2957] channel.c: Avoiding initial deadlock for
'SIP/7188-6b1f'
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- SIP/7188-6b1f is ringing
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Acked pending invite 102
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Stopping retransmission on
'2978d85271fdaf3d0ac2e5b244e78773 at 192.168.2.66' of Request 102: Match
Found
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: build_route: Contact hop: 
Jun 26 16:53:37 VERBOSE[8290] logger.c: -- SIP/7188-6b1f answered
Local/7188 at from-internal-7036,2
Jun 26 16:53:37 DEBUG[8287] app_queue.c: Dunno what to do with control
type -1
Jun 26 16:53:37 VERBOSE[8287] logger.c: --
Local/7188 at from-internal-7036,1 answered Zap/13-1
Jun 26 16:53:37 DEBUG[8287] chan_zap.c: Set option TONE VERIFY, mode:
MUTECONF(1) on Zap/13-1
Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Stopped music on hold on
Zap/13-1
Jun 26 16:53:37 DEBUG[8287] channel.c: Scheduling timer at 0 sample
intervals
Jun 26 16:54:02 DEBUG[8290] channel.c: Didn't get a frame from channel:
SIP/7188-6b1f
Jun 26 16:54:02 DEBUG[8290] channel.c: Bridge stops bridging channels
Local/7188 at from-internal-7036,2 and SIP/7188-6b1f
Jun 26 16:54:02 DEBUG[8290] chan_sip.c: update_call_counter(7188) -
decrement call limit counter
Jun 26 16:54:02 DEBUG[8290] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial,
s, 10) exited non-zero on 'Local/7188 at from-internal-7036,2' in macro
'dial'
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial,
s, 10) exited non-zero on 'Local/7188 at from-internal-7036,2' in macro
'exten-vm'
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial,
s, 10) exited non-zero on 'Local/7188 at from-internal-7036,2'
Jun 26 16:54:02 DEBUG[8290] res_monitor.c: monitor executing ( nice -n
19 soxmix
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-in.gsm"
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-out.gsm"
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901.gsm" && rm
-f "/var/spool/asterisk/monitor/20060626-165333-1151304813.901-"* ) &
Jun 26 16:54:02 DEBUG[8287] channel.c: Didn't get a frame from channel:
Local/7188 at from-internal-7036,1
Jun 26 16:54:02 DEBUG[8287] channel.c: Bridge stops bridging channels
Zap/13-1 and Local/7188 at from-internal-7036,1
Jun 26 16:54:02 VERBOSE[8287] logger.c: == Spawn extension (ext-queues,
7141, 6) exited non-zero on 'Zap/13-1'
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value:
ON(1) on Zap/13-1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Hangup: channel: 13 index = 0,
normal = 27, callwait = -1, thirdcall = -1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Not yet hungup... Calling hangup
once with icause, and clearing call
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on
channel 13
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/13-1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Updated conferencing on 13, with
0 conference users
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value:
OFF(0) on Zap/13-1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on
channel 13
Jun 26 16:54:02 VERBOSE[8287] logger.c: -- Hungup 'Zap/13-1'

 

Isaac Xiao

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060626/092a4a5f/attachment.htm


More information about the asterisk-users mailing list