Hi Sergio and Amit, when i send a participant request from mcuWeb to the softphone that i registered into the asterisk .i am getting this log.
[#|2011-09-23T14:59:49.972+0530|WARNING|sun-glassfish-comms-server1.5|global|_ThreadID=32;_ThreadName=Thread-29;_RequestID=16cd3197-093e-47e0-a94f-fed259ea4e19;|SimpleProxyServlet:doResponse Got request: SIP/2.0 408 Request Timeout From: <sip:mcu@null>;tag=gswyycgc-i3m Cseq: 1 INVITE To: <sip:200@192.168.115.53>;tag=gswyz15g-i3q Server: Glassfish_SIP_1.0.0 Call-Id: 192.168.115.24_4_3192961169039426897 |#] [#|2011-09-23T14:59:56.484+0530|WARNING|sun-glassfish-comms-server1.5|global|_ThreadID=33;_ThreadName=Thread-35;_RequestID=a7ba4f75-a2f8-4e35-a549-8b219d4126e8;|SimpleProxyServlet:doResponse Got request: SIP/2.0 408 Request Timeout From: <sip:mcu@null>;tag=gswyyhh8-i3p Cseq: 1 INVITE To: <sip:100@192.168.115.53>;tag=gswyz66c-i3r Server: Glassfish_SIP_1.0.0 Call-Id: 192.168.115.24_5_1892418549266906116 when i dial to mcuWeb using the dial rule ,i get this log Sep 23 14:56:06 fstl-desktop asterisk[9005]: NOTICE[11870]: pbx_ael.c:180 in pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. Sep 23 14:56:06 fstl-desktop asterisk[9005]: NOTICE[11870]: pbx_ael.c:187 in pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. Sep 23 14:56:06 fstl-desktop asterisk[9005]: NOTICE[11870]: pbx_ael.c:192 in pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. Sep 23 14:56:06 fstl-desktop asterisk[9005]: NOTICE[11870]: pbx_ael.c:195 in pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. Sep 23 14:56:06 fstl-desktop asterisk[9005]: NOTICE[11870]: app_queue.c:6484 in reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. Sep 23 14:56:06 fstl-desktop asterisk[9005]: NOTICE[11870]: chan_skinny.c:7209 in config_load: Configuring skinny from skinny.conf Sep 23 14:57:12 fstl-desktop asterisk[9005]: WARNING[9018]: chan_sip.c:3620 in retrans_pkt: Retransmission timeout reached on transmission 1f9119e162c1866814b1b9b0040f99d2@192.168.115.53:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions#012Packettimed out after 31999ms with no response Sep 23 14:57:12 fstl-desktop asterisk[9005]: WARNING[9018]: chan_sip.c:3649 in retrans_pkt: Hanging up call 1f9119e162c1866814b1b9b0040f99d2@192.168.115.53:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Sep 23 14:57:44 fstl-desktop asterisk[9005]: WARNING[9018]: chan_sip.c:3620 in retrans_pkt: Retransmission timeout reached on transmission 393b93de45b2b9fc794edadf70700a66@192.168.115.53:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions#012Packettimed out after 32000ms with no response Sep 23 14:57:50 fstl-desktop asterisk[9005]: WARNING[9018]: chan_sip.c:3620 in retrans_pkt: Retransmission timeout reached on transmission 1f74cfb04b20e6d40ae4c5a228f2a5c0@192.168.115.53:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions#012Packettimed out after 32000ms with no response my main intention is to evaluate the performance of the mixer ,whats the cpu usage for 2 or more participants and the quality of mixing . please help me with the sip.conf and extenison.conf so that i can get over this signalling part and concentrate on the Video mixer.please attach the sip.conf and extension.conf files. Thanks, N Sivaramkrishna.
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