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<p class=MsoNormal><font size=1 face=Arial><span lang=EN-US style='font-size:
9.0pt;font-family:Arial'>Hi,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=1 face=Arial><span lang=EN-US style='font-size:
9.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=1 face=Arial><span lang=EN-US style='font-size:
9.0pt;font-family:Arial'>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.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=1 face=Arial><span lang=EN-US style='font-size:
9.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Times New Roman"><span lang=EN-US
style='font-size:10.5pt'>Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Executing
Dial("Local/7188@from-internal-7036,2",
"SIP/7188|30|trWwT") in new stack<br>
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on RTP to 0<br>
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on VRTP to 0<br>
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Outgoing Call for 7188<br>
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Called 7188<br>
Jun 26 16:53:35 VERBOSE[8287] logger.c: -- Local/7188@from-internal-7036,1 is
ringing<br>
Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '2978d85271fdaf3d0ac2e5b244e78773@192.168.2.66'
Request 102: Found<br>
Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '2978d85271fdaf3d0ac2e5b244e78773@192.168.2.66'
Request 102: Found<br>
Jun 26 16:53:35 DEBUG[2957] channel.c: Avoiding initial deadlock for
'SIP/7188-6b1f'<br>
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- SIP/7188-6b1f is ringing<br>
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Acked pending invite 102<br>
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Stopping retransmission on
'2978d85271fdaf3d0ac2e5b244e78773@192.168.2.66' of Request 102: Match Found<br>
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: build_route: Contact hop: <br>
<sip:_x0037_188_x0040_192.168.2.131>Jun 26 16:53:37 VERBOSE[8290] logger.c: --
SIP/7188-6b1f answered Local/7188@from-internal-7036,2<br>
Jun 26 16:53:37 DEBUG[8287] app_queue.c: Dunno what to do with control type -1<br>
Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Local/7188@from-internal-7036,1
answered Zap/13-1<br>
Jun 26 16:53:37 DEBUG[8287] chan_zap.c: Set option TONE VERIFY, mode:
MUTECONF(1) on Zap/13-1<br>
Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Stopped music on hold on Zap/13-1<br>
Jun 26 16:53:37 DEBUG[8287] channel.c: Scheduling timer at 0 sample intervals<br>
Jun 26 16:54:02 DEBUG[8290] channel.c: Didn't get a frame from channel:
SIP/7188-6b1f<br>
Jun 26 16:54:02 DEBUG[8290] channel.c: Bridge stops bridging channels
Local/7188@from-internal-7036,2 and SIP/7188-6b1f<br>
Jun 26 16:54:02 DEBUG[8290] chan_sip.c: update_call_counter(7188) - decrement
call limit counter<br>
Jun 26 16:54:02 DEBUG[8290] app_dial.c: Exiting with DIALSTATUS=ANSWER.<br>
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10)
exited non-zero on 'Local/7188@from-internal-7036,2' in macro 'dial'<br>
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10)
exited non-zero on 'Local/7188@from-internal-7036,2' in macro 'exten-vm'<br>
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10)
exited non-zero on 'Local/7188@from-internal-7036,2'<br>
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-"* )
&<br>
Jun 26 16:54:02 DEBUG[8287] channel.c: Didn't get a frame from channel:
Local/7188@from-internal-7036,1<br>
Jun 26 16:54:02 DEBUG[8287] channel.c: Bridge stops bridging channels Zap/13-1
and Local/7188@from-internal-7036,1<br>
Jun 26 16:54:02 VERBOSE[8287] logger.c: == Spawn extension (ext-queues, 7141,
6) exited non-zero on 'Zap/13-1'<br>
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value: ON(1) on
Zap/13-1<br>
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Hangup: channel: 13 index = 0, normal =
27, callwait = -1, thirdcall = -1<br>
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Not yet hungup... Calling hangup once
with icause, and clearing call<br>
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on channel
13<br>
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option TDD MODE, value: OFF(0) on
Zap/13-1<br>
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Updated conferencing on 13, with 0
conference users<br>
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on
Zap/13-1<br>
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on channel
13<br>
Jun 26 16:54:02 VERBOSE[8287] logger.c: -- Hungup 'Zap/13-1'<o:p></o:p></span></font></p>
</sip:_x0037_188_x0040_192.168.2.131>
<p class=MsoNormal><font size=2 face="Times New Roman"><span lang=EN-US><o:p> </o:p></span></font></p>
<p class=MsoNormal><st1:PersonName w:st="on"><font size=2 face="Times New Roman"><span
lang=EN-US style='font-size:10.5pt'>Isaac Xiao</span></font></st1:PersonName><span
lang=EN-US><o:p></o:p></span></p>
<p class=MsoNormal><font size=1 face=Arial><span lang=EN-US style='font-size:
9.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
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