hi, since some time outgoing sofia call crashes freeswitch. this is the log woth today's trunk: 2007-04-25 02:04:40 [NOTICE] switch_core_state_machine.c:124 switch_core_standard_on_execute() Execute bridge(sofia/sip.intellectics.com/+432256815811@sip.intellectics.com) 2007-04-25 02:04:40 [NOTICE] switch_channel.c:347 switch_channel_set_name() New Chan sofia/sip.intellectics.com/+432256815811@sip.intellectics.com [8ff33714-f2c0-11db-b0ae-01ef474317ba] 2007-04-25 02:04:40 [DEBUG] mod_sofia.c:1084 sofia_outgoing_channel() sofia/sip.intellectics.com/+432256815811@sip.intellectics.com State Change CS_NEW -> CS_INIT 2007-04-25 02:04:40 [DEBUG] switch_core_state_machine.c:378 switch_core_session_run() (sofia/sip.intellectics.com/+432256815811@sip.intellectics.com) State INIT 2007-04-25 02:04:40 [DEBUG] mod_sofia.c:88 sofia_on_init() SOFIA INIT 2007-04-25 02:04:40 [DEBUG] mod_sofia.c:102 sofia_on_init() sofia/sip.intellectics.com/+432256815811@sip.intellectics.com State Change CS_INIT -> CS_RING 2007-04-25 02:04:40 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() (sofia/sip.intellectics.com/+432256815811@sip.intellectics.com) State RING 2007-04-25 02:04:40 [DEBUG] switch_ivr_originate.c:618 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2007-04-25 02:04:40 [DEBUG] mod_sofia.c:117 sofia_on_ring() SOFIA RING 2007-04-25 02:04:40 [DEBUG] switch_ivr_originate.c:48 originate_on_ring() sofia/sip.intellectics.com/+432256815811@sip.intellectics.com State Change CS_RING -> CS_HOLD 2007-04-25 02:04:40 [DEBUG] switch_core_state_machine.c:556 switch_core_session_run() (sofia/sip.intellectics.com/+432256815811@sip.intellectics.com) State HOLD 2007-04-25 02:04:40 [DEBUG] sofia.c:75 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/sip.intellectics.com/+432256815811@sip.intellectics.com 2007-04-25 02:04:40 [DEBUG] switch_core_state_machine.c:198 switch_core_standard_on_hold() Standard HOLD 2007-04-25 02:04:40 [DEBUG] sofia.c:860 sofia_handle_sip_i_state() Channel sofia/sip.intellectics.com/+432256815811@sip.intellectics.com entering state [calling] 2007-04-25 02:04:40 [DEBUG] sofia.c:75 sofia_event_callback() event [nua_r_invite] status [180][Ringing] session: sofia/sip.intellectics.com/+432256815811@sip.intellectics.com 2007-04-25 02:04:40 [DEBUG] sofia.c:75 sofia_event_callback() event [nua_i_state] status [180][Ringing] session: sofia/sip.intellectics.com/+432256815811@sip.intellectics.com 2007-04-25 02:04:40 [DEBUG] sofia.c:860 sofia_handle_sip_i_state() Channel sofia/sip.intellectics.com/+432256815811@sip.intellectics.com entering state [proceeding] 2007-04-25 02:04:40 [NOTICE] sofia.c:886 sofia_handle_sip_i_state() Ring-Ready sofia/sip.intellectics.com/+432256815811@sip.intellectics.com! 2007-04-25 02:04:40 [INFO] switch_core_session.c:345 switch_core_session_receive_message() Kill DingaLing/new [BREAK] 2007-04-25 02:04:40 [DEBUG] mod_dingaling.c:1286 channel_kill_channel() DingaLing/new CHANNEL KILL 2007-04-25 02:04:40 [DEBUG] sofia.c:860 sofia_handle_sip_i_state() Channel sofia/sip.intellectics.com/+432256815811@sip.intellectics.com entering state [ready] 2007-04-25 02:04:40 [DEBUG] sofia.c:864 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 10638 10638 IN IP4 85.126.75.146 s=session c=IN IP4 85.126.75.146 t=0 0 m=audio 10902 RTP/AVP 0 10 101 a=rtpmap:0 PCMU/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2007-04-25 02:04:40 [DEBUG] sofia_glue.c:1153 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0]/[PCMU:0] 2007-04-25 02:04:40 [INFO] sofia_glue.c:826 sofia_glue_tech_set_codec() Set Codec sofia/sip.intellectics.com/+432256815811@sip.intellectics.com PCMU/8000 20 ms 2007-04-25 02:04:40 [DEBUG] sofia_glue.c:1125 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2007-04-25 02:04:40 [INFO] sofia_glue.c:891 sofia_glue_activate_rtp() AUDIO RTP [sofia/sip.intellectics.com/+432256815811@sip.intellectics.com] 85.126.75.146:16386->85.126.75.146:10902 codec: 0 ms: 20 2007-04-25 02:04:40 [DEBUG] switch_rtp.c:533 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-04-25 02:04:40 [INFO] switch_channel.c:1174 switch_channel_perform_mark_answered() Kill DingaLing/new [BREAK] 2007-04-25 02:04:40 [DEBUG] mod_dingaling.c:1286 channel_kill_channel() DingaLing/new CHANNEL KILL 2007-04-25 02:04:40 [NOTICE] sofia.c:1121 sofia_handle_sip_i_state() Channel [sofia/sip.intellectics.com/+432256815811@sip.intellectics.com] has been answered 2007-04-25 02:04:40 [INFO] switch_core_session.c:345 switch_core_session_receive_message() Kill DingaLing/new [BREAK] 85-126-75-146:/usr/src/freeswitch.trunk# dialplan: <extension name="1004j"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^1004j"> <action application="bridge" data="sofia/sip.intellectics.com/+432256815811@sip.intellectics.com"/> </condition> </extension> sofia: <profile name="$${domain}"> <gateways> <gateway name="asterlink"> <param name="realm" value="asterisk"/> <param name="username" value="freeswitch"/> <param name="password" value="xxxx"/> <param name="proxy" value="sip.intellectics.com:5060"/> <param name="expire-seconds" value="600"/> </gateway> </gateways> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5061"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="L16,PCMU"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="false"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="auto"/> <param name="sip-ip" value="auto"/> <param name="sip-domain" value="freeswitch.intellectics.com"/> <!--Uncomment to set all inbound calls to no media mode--> <!--<param name="inbound-no-media" value="true"/>--> <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> <param name="inbound-late-negotiation" value="false"/> <!-- this lets anything register --> <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="false"/> <!--TTL for nonce in sip auth--> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <!-- on authed calls, authenticate *all* the packets not just invite --> <!--<param name="auth-all-packets" value="true"/>--> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.sipdiscount.com"/> --> <!-- <param name="ext-rtp-ip" value="sip.intellectics.com"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:10.0.1.251:5555"/>--> </settings> -- (Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden, Austria Tel. +43 2256 815811 Mobile +43 699 12532539 Fax +43 2256 815819 (Tel. US: 360-2267729 UK: 0870 4787064) sip:pekka.kurki@sip.intellectics.com skype: pkkurki, pkkurki-sip (sip gateway) gtalk: pekkis50@gtalk.com (sip gateway - experimental) gizmo: pekkis50 h323:432256815811@sip.intellectics.com email: Pekka.Kurki@intellectics.com www: http://pekka.kurki.at