Hi Sergio and Amit,<br><br>when i send a participant request from mcuWeb to the softphone that i registered into the asterisk .i am getting this log.<br><br>[#|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:<br>
SIP/2.0 408 Request Timeout<br>From: <sip:mcu@null>;tag=gswyycgc-i3m<br>Cseq: 1 INVITE<br>To: <<a href="mailto:sip%3A200@192.168.115.53">sip:200@192.168.115.53</a>>;tag=gswyz15g-i3q<br>Server: Glassfish_SIP_1.0.0<br>
Call-Id: 192.168.115.24_4_3192961169039426897<br><br>|#]<br><br>[#|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:<br>
SIP/2.0 408 Request Timeout<br>From: <sip:mcu@null>;tag=gswyyhh8-i3p<br>Cseq: 1 INVITE<br>To: <<a href="mailto:sip%3A100@192.168.115.53">sip:100@192.168.115.53</a>>;tag=gswyz66c-i3r<br>Server: Glassfish_SIP_1.0.0<br>
Call-Id: 192.168.115.24_5_1892418549266906116<br><br><br>when i dial to mcuWeb using the dial rule ,i get this log<br>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'.<br>
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'.<br>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'.<br>
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'.<br>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.<br>
Sep 23 14:56:06 fstl-desktop asterisk[9005]: NOTICE[11870]: chan_skinny.c:7209 in config_load: Configuring skinny from skinny.conf<br>Sep 23 14:57:12 fstl-desktop asterisk[9005]: WARNING[9018]: chan_sip.c:3620 in retrans_pkt: Retransmission timeout reached on transmission <a href="http://1f9119e162c1866814b1b9b0040f99d2@192.168.115.53:5060">1f9119e162c1866814b1b9b0040f99d2@192.168.115.53:5060</a> for seqno 102 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions#012Packet">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions#012Packet</a> timed out after 31999ms with no response<br>
Sep 23 14:57:12 fstl-desktop asterisk[9005]: WARNING[9018]: chan_sip.c:3649 in retrans_pkt: Hanging up call <a href="http://1f9119e162c1866814b1b9b0040f99d2@192.168.115.53:5060">1f9119e162c1866814b1b9b0040f99d2@192.168.115.53:5060</a> - no reply to our critical packet (see <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).<br>
Sep 23 14:57:44 fstl-desktop asterisk[9005]: WARNING[9018]: chan_sip.c:3620 in retrans_pkt: Retransmission timeout reached on transmission <a href="http://393b93de45b2b9fc794edadf70700a66@192.168.115.53:5060">393b93de45b2b9fc794edadf70700a66@192.168.115.53:5060</a> for seqno 102 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions#012Packet">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions#012Packet</a> timed out after 32000ms with no response<br>
Sep 23 14:57:50 fstl-desktop asterisk[9005]: WARNING[9018]: chan_sip.c:3620 in retrans_pkt: Retransmission timeout reached on transmission <a href="http://1f74cfb04b20e6d40ae4c5a228f2a5c0@192.168.115.53:5060">1f74cfb04b20e6d40ae4c5a228f2a5c0@192.168.115.53:5060</a> for seqno 102 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions#012Packet">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions#012Packet</a> timed out after 32000ms with no response<br>
<br><br>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 .<br>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.<br>
<br>Thanks,<br>N Sivaramkrishna.<br><br><br>