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<DIV><FONT size=2>Ok I'm trying to figure it out</FONT></DIV>
<DIV><FONT size=2>thanks</FONT></DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=anthony.minessale@gmail.com
href="mailto:anthony.minessale@gmail.com">Anthony Minessale</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">FreeSWITCH Users Help</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Saturday, April 02, 2011 3:34
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] originate
from cli</DIV>
<DIV><BR></DIV>
<P>You never get the reply to the invite trace from the thing you are calling
and see if they are getting any packets. Maybe you have iptables or
selinux on on one host or the other.</P>
<DIV class=gmail_quote>On Apr 1, 2011 10:37 PM, "Madovsky" <<A
href="mailto:infos@madovsky.org">infos@madovsky.org</A>> wrote:<BR
type="attribution">> sorry the sip trace :<BR>> after this
command:<BR>> <BR>> /usr/local/freeswitch/bin/fs_cli -x "originate
user/9999999999 1111111111"<BR>> <BR>>
------------------------------------------------------------------------<BR>>
2011-04-01 23:33:39.673581 [DEBUG] sofia.c:4754 Channel sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
entering state [calling][0]<BR>> send 1385 bytes to udp/[11.22.33.44]:52767
at 03:33:40.673952:<BR>>
------------------------------------------------------------------------<BR>>
INVITE <A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
SIP/2.0<BR>> Via: SIP/2.0/UDP
11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD<BR>> Max-Forwards:
70<BR>> From: "" <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;tag=S16mjZ06UgQSe<BR>>
To: <<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>><BR>>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5<BR>> CSeq: 10521545
INVITE<BR>> Contact: <<A
href="http://sip:mod_sofia@11.22.33.44:5080">sip:mod_sofia@11.22.33.44:5080</A>><BR>>
User-Agent: CiscoSystems-SIP-GW-UA<BR>> Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE<BR>> Supported: timer, precondition, path, replaces<BR>>
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer<BR>>
Content-Type: application/sdp<BR>> Content-Disposition: session<BR>>
Content-Length: 492<BR>> X-FS-Support: update_display<BR>>
Remote-Party-ID: <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;party=calling;screen=yes;privacy=off<BR>>
<BR>> v=0<BR>> o=FreeSWITCH 1301682069 1301682070 IN IP4
11.22.33.44<BR>> s=FreeSWITCH<BR>> c=IN IP4 11.22.33.44<BR>> t=0
0<BR>> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101
13<BR>> a=rtpmap:98 SPEEX/16000<BR>> a=rtpmap:99 G726-16/8000<BR>>
a=rtpmap:100 G726-24/8000<BR>> a=rtpmap:102 G726-32/8000<BR>>
a=rtpmap:103 G726-40/8000<BR>> a=rtpmap:104 G7221/16000<BR>> a=fmtp:104
bitrate=32000<BR>> a=rtpmap:105 iLBC/8000<BR>> a=fmtp:105
mode=20<BR>> a=rtpmap:106 L16/8000<BR>> a=rtpmap:101
telephone-event/8000<BR>> a=fmtp:101 0-16<BR>> a=ptime:20<BR>>
------------------------------------------------------------------------<BR>>
send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:42.673971:<BR>>
------------------------------------------------------------------------<BR>>
INVITE <A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
SIP/2.0<BR>> Via: SIP/2.0/UDP
11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD<BR>> Max-Forwards:
70<BR>> From: "" <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;tag=S16mjZ06UgQSe<BR>>
To: <<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>><BR>>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5<BR>> CSeq: 10521545
INVITE<BR>> Contact: <<A
href="http://sip:mod_sofia@11.22.33.44:5080">sip:mod_sofia@11.22.33.44:5080</A>><BR>>
User-Agent: CiscoSystems-SIP-GW-UA<BR>> Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE<BR>> Supported: timer, precondition, path, replaces<BR>>
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer<BR>>
Content-Type: application/sdp<BR>> Content-Disposition: session<BR>>
Content-Length: 492<BR>> X-FS-Support: update_display<BR>>
Remote-Party-ID: <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;party=calling;screen=yes;privacy=off<BR>>
<BR>> v=0<BR>> o=FreeSWITCH 1301682069 1301682070 IN IP4
11.22.33.44<BR>> s=FreeSWITCH<BR>> c=IN IP4 11.22.33.44<BR>> t=0
0<BR>> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101
13<BR>> a=rtpmap:98 SPEEX/16000<BR>> a=rtpmap:99 G726-16/8000<BR>>
a=rtpmap:100 G726-24/8000<BR>> a=rtpmap:102 G726-32/8000<BR>>
a=rtpmap:103 G726-40/8000<BR>> a=rtpmap:104 G7221/16000<BR>> a=fmtp:104
bitrate=32000<BR>> a=rtpmap:105 iLBC/8000<BR>> a=fmtp:105
mode=20<BR>> a=rtpmap:106 L16/8000<BR>> a=rtpmap:101
telephone-event/8000<BR>> a=fmtp:101 0-16<BR>> a=ptime:20<BR>>
------------------------------------------------------------------------<BR>>
send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:46.673944:<BR>>
------------------------------------------------------------------------<BR>>
INVITE <A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
SIP/2.0<BR>> Via: SIP/2.0/UDP
11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD<BR>> Max-Forwards:
70<BR>> From: "" <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;tag=S16mjZ06UgQSe<BR>>
To: <<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>><BR>>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5<BR>> CSeq: 10521545
INVITE<BR>> Contact: <<A
href="http://sip:mod_sofia@11.22.33.44:5080">sip:mod_sofia@11.22.33.44:5080</A>><BR>>
User-Agent: CiscoSystems-SIP-GW-UA<BR>> Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE<BR>> Supported: timer, precondition, path, replaces<BR>>
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer<BR>>
Content-Type: application/sdp<BR>> Content-Disposition: session<BR>>
Content-Length: 492<BR>> X-FS-Support: update_display<BR>>
Remote-Party-ID: <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;party=calling;screen=yes;privacy=off<BR>>
<BR>> v=0<BR>> o=FreeSWITCH 1301682069 1301682070 IN IP4
11.22.33.44<BR>> s=FreeSWITCH<BR>> c=IN IP4 11.22.33.44<BR>> t=0
0<BR>> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101
13<BR>> a=rtpmap:98 SPEEX/16000<BR>> a=rtpmap:99 G726-16/8000<BR>>
a=rtpmap:100 G726-24/8000<BR>> a=rtpmap:102 G726-32/8000<BR>>
a=rtpmap:103 G726-40/8000<BR>> a=rtpmap:104 G7221/16000<BR>> a=fmtp:104
bitrate=32000<BR>> a=rtpmap:105 iLBC/8000<BR>> a=fmtp:105
mode=20<BR>> a=rtpmap:106 L16/8000<BR>> a=rtpmap:101
telephone-event/8000<BR>> a=fmtp:101 0-16<BR>> a=ptime:20<BR>>
------------------------------------------------------------------------<BR>>
send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:54.673950:<BR>>
------------------------------------------------------------------------<BR>>
INVITE <A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
SIP/2.0<BR>> Via: SIP/2.0/UDP
11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD<BR>> Max-Forwards:
70<BR>> From: "" <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;tag=S16mjZ06UgQSe<BR>>
To: <<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>><BR>>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5<BR>> CSeq: 10521545
INVITE<BR>> Contact: <<A
href="http://sip:mod_sofia@11.22.33.44:5080">sip:mod_sofia@11.22.33.44:5080</A>><BR>>
User-Agent: CiscoSystems-SIP-GW-UA<BR>> Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE<BR>> Supported: timer, precondition, path, replaces<BR>>
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer<BR>>
Content-Type: application/sdp<BR>> Content-Disposition: session<BR>>
Content-Length: 492<BR>> X-FS-Support: update_display<BR>>
Remote-Party-ID: <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;party=calling;screen=yes;privacy=off<BR>>
<BR>> v=0<BR>> o=FreeSWITCH 1301682069 1301682070 IN IP4
11.22.33.44<BR>> s=FreeSWITCH<BR>> c=IN IP4 11.22.33.44<BR>> t=0
0<BR>> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101
13<BR>> a=rtpmap:98 SPEEX/16000<BR>> a=rtpmap:99 G726-16/8000<BR>>
a=rtpmap:100 G726-24/8000<BR>> a=rtpmap:102 G726-32/8000<BR>>
a=rtpmap:103 G726-40/8000<BR>> a=rtpmap:104 G7221/16000<BR>> a=fmtp:104
bitrate=32000<BR>> a=rtpmap:105 iLBC/8000<BR>> a=fmtp:105
mode=20<BR>> a=rtpmap:106 L16/8000<BR>> a=rtpmap:101
telephone-event/8000<BR>> a=fmtp:101 0-16<BR>> a=ptime:20<BR>>
------------------------------------------------------------------------<BR>>
send 1385 bytes to udp/[11.22.33.44]:52767 at 03:34:10.674077:<BR>>
------------------------------------------------------------------------<BR>>
INVITE <A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
SIP/2.0<BR>> Via: SIP/2.0/UDP
11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD<BR>> Max-Forwards:
70<BR>> From: "" <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;tag=S16mjZ06UgQSe<BR>>
To: <<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>><BR>>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5<BR>> CSeq: 10521545
INVITE<BR>> Contact: <<A
href="http://sip:mod_sofia@11.22.33.44:5080">sip:mod_sofia@11.22.33.44:5080</A>><BR>>
User-Agent: CiscoSystems-SIP-GW-UA<BR>> Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE<BR>> Supported: timer, precondition, path, replaces<BR>>
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer<BR>>
Content-Type: application/sdp<BR>> Content-Disposition: session<BR>>
Content-Length: 492<BR>> X-FS-Support: update_display<BR>>
Remote-Party-ID: <<A
href="mailto:sip%3A0000000000@11.22.33.44">sip:0000000000@11.22.33.44</A>>;party=calling;screen=yes;privacy=off<BR>>
<BR>> v=0<BR>> o=FreeSWITCH 1301682069 1301682070 IN IP4
11.22.33.44<BR>> s=FreeSWITCH<BR>> c=IN IP4 11.22.33.44<BR>> t=0
0<BR>> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101
13<BR>> a=rtpmap:98 SPEEX/16000<BR>> a=rtpmap:99 G726-16/8000<BR>>
a=rtpmap:100 G726-24/8000<BR>> a=rtpmap:102 G726-32/8000<BR>>
a=rtpmap:103 G726-40/8000<BR>> a=rtpmap:104 G7221/16000<BR>> a=fmtp:104
bitrate=32000<BR>> a=rtpmap:105 iLBC/8000<BR>> a=fmtp:105
mode=20<BR>> a=rtpmap:106 L16/8000<BR>> a=rtpmap:101
telephone-event/8000<BR>> a=fmtp:101 0-16<BR>> a=ptime:20<BR>>
------------------------------------------------------------------------<BR>>
2011-04-01 23:34:11.674022 [DEBUG] sofia.c:4754 Channel sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
entering state [terminated][408]<BR>> 2011-04-01 23:34:11.674022 [DEBUG]
switch_channel.c:2563 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
Callstate Change RINGING -> HANGUP<BR>> 2011-04-01 23:34:11.674022
[NOTICE] sofia.c:5394 Hangup sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
[CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]<BR>> 2011-04-01
23:34:11.674022 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
[KILL]<BR>> 2011-04-01 23:34:11.674022 [DEBUG] switch_core_session.c:1116
Send signal sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
[BREAK]<BR>> 2011-04-01 23:34:11.674022 [DEBUG]
switch_core_state_machine.c:320 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
Running State Change CS_HANGUP<BR>> 2011-04-01 23:34:11.674022 [DEBUG]
switch_ivr_originate.c:3506 Originate Resulted in Error Cause: 102
[RECOVERY_ON_TIMER_EXPIRE]<BR>> 2011-04-01 23:34:11.674022 [ERR]
switch_ivr_originate.c:2640 Cannot create outgoing channel of type [user]
cause: [RECOVERY_ON_TIMER_EXPIRE]<BR>> 2011-04-01 23:34:11.674022 [DEBUG]
switch_ivr_originate.c:3506 Originate Resulted in Error Cause: 102
[RECOVERY_ON_TIMER_EXPIRE]<BR>> 2011-04-01 23:34:11.675057 [DEBUG]
switch_core_state_machine.c:560 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
State HANGUP<BR>> 2011-04-01 23:34:11.675057 [DEBUG] mod_sofia.c:451
sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
Overriding SIP cause 504 with 408 from the other leg<BR>> 2011-04-01
23:34:11.675057 [DEBUG] mod_sofia.c:457 Channel sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
hanging up, cause: RECOVERY_ON_TIMER_EXPIRE<BR>> 2011-04-01 23:34:11.695371
[DEBUG] switch_core_state_machine.c:46 sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE<BR>> 2011-04-01
23:34:11.695371 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
State HANGUP going to sleep<BR>> 2011-04-01 23:34:11.695371 [DEBUG]
switch_core_state_machine.c:351 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
State Change CS_HANGUP -> CS_REPORTING<BR>> 2011-04-01 23:34:11.695371
[DEBUG] switch_core_session.c:1116 Send signal sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
[BREAK]<BR>> 2011-04-01 23:34:11.695371 [DEBUG]
switch_core_state_machine.c:320 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
Running State Change CS_REPORTING<BR>> 2011-04-01 23:34:11.695371 [DEBUG]
switch_core_state_machine.c:620 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
State REPORTING<BR>> 2011-04-01 23:34:11.695371 [DEBUG]
switch_core_state_machine.c:53 sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE<BR>> 2011-04-01
23:34:11.695371 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
State REPORTING going to sleep<BR>> 2011-04-01 23:34:11.695371 [DEBUG]
switch_core_state_machine.c:345 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
State Change CS_REPORTING -> CS_DESTROY<BR>> 2011-04-01 23:34:11.695371
[DEBUG] switch_core_session.c:1116 Send signal sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
[BREAK]<BR>> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1288
Session 5 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
Locked, Waiting on external entities<BR>> 2011-04-01 23:34:11.695371
[NOTICE] switch_core_session.c:1306 Session 5 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
Ended<BR>> 2011-04-01 23:34:11.695371 [NOTICE] switch_core_session.c:1308
Close Channel sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
[CS_DESTROY]<BR>> 2011-04-01 23:34:11.696436 [DEBUG]
switch_core_state_machine.c:449 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
Callstate Change HANGUP -> DOWN<BR>> 2011-04-01 23:34:11.696436 [DEBUG]
switch_core_state_machine.c:452 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
Running State Change CS_DESTROY<BR>> 2011-04-01 23:34:11.696436 [DEBUG]
switch_core_state_machine.c:462 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
State DESTROY<BR>> 2011-04-01 23:34:11.696436 [DEBUG] mod_sofia.c:362
sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
SOFIA DESTROY<BR>> 2011-04-01 23:34:11.696436 [DEBUG]
switch_core_state_machine.c:60 sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>
Standard DESTROY<BR>> 2011-04-01 23:34:11.696436 [DEBUG]
switch_core_state_machine.c:462 (sofia/internal/<A
href="http://sip:99999999999@11.22.33.44:52767">sip:99999999999@11.22.33.44:52767</A>)
State DESTROY going to sleep<BR>> <BR>> <BR>> a<BR>> <BR>>
----- Original Message ----- <BR>> From: Steven Ayre <BR>> To:
FreeSWITCH Users Help <BR>> Sent: Friday, April 01, 2011 5:30 PM<BR>>
Subject: Re: [Freeswitch-users] originate from cli<BR>> <BR>> <BR>>
It can also be that the other side sent FS a reply saying that *it* had timed
out. siptrace will also show if that's the case.<BR>> <BR>> <BR>>
<BR>> On 1 April 2011 22:29, Steven Ayre <<A
href="mailto:steveayre@gmail.com">steveayre@gmail.com</A>> wrote:<BR>>
<BR>> It means FS sent a message and didn't get a reply (timed
out).<BR>> <BR>> As anthm says, look at the siptrace - that'll show you
what's being sent / received.<BR>> <BR>> -Steve<BR>> <BR>>
<BR>> <BR>> On 1 April 2011 21:17, Madovsky <<A
href="mailto:infos@madovsky.org">infos@madovsky.org</A>> wrote:<BR>>
<BR>> I make some test with originate from cli.<BR>> <BR>>
/usr/local/freeswitch/bin/fs_cli -x "originate user/9999 7777 XML
public"<BR>> -ERR RECOVERY_ON_TIMER_EXPIRE<BR>> <BR>> 9999 and 7777
are ready to receive calls and I don't have any NAT.<BR>> <BR>> is
anyone can explain what it means ?<BR>> <BR>> Thanks<BR>> <BR>>
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