[Freeswitch-users] outgoing sofia call crashes freeswitch on slug (linksys n2slu)
Pekka K. Kurki
pekka.kurki at intellectics.com
Tue Apr 24 17:18:23 PDT 2007
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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at sip.intellectics.com
2007-04-25 02:04:40 [DEBUG] sofia.c:860 sofia_handle_sip_i_state()
Channel sofia/sip.intellectics.com/+432256815811 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at sip.intellectics.com
skype: pkkurki, pkkurki-sip (sip gateway)
gtalk: pekkis50 at gtalk.com (sip gateway - experimental)
gizmo: pekkis50
h323:432256815811 at sip.intellectics.com
email: Pekka.Kurki at intellectics.com
www: http://pekka.kurki.at
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