make sure that your sip call is using the correct sofia profile. your sofia.conf show a profile for mydomain1.com, so ..... sofia/mydomain1.com/1234@66.250.68.194 not sofia/test//1234@66.250.68.194 Alexei Archinov wrote:
Hello,
Thank you very much for the suggestion to re-compile the project – it works. Now since it is installed and does not produce any errors on start up, I tried to call 1234 to test the switch. The attempt was not successful and the output is attached. As I see the problem is with mod_sofia configuration, while I did not make any changes to default one. The configuration I have is attached at the bottom as well.
________________________________________________
freeswitch@Archinov_IBM> pacall 1234
2006-10-28 13:38:11 [NOTICE] switch_channel.c:383 switch_channel_set_name() New
Chan PortAudio/1234-3d6c [3204a213-c70b-b04a-8399-1a2d3f9dfe04]
2006-10-28 13:38:11 [INFO] mod_portaudio.c:793 engage_device() Loaded codec L16
8000hz 20ms on PortAudio/1234-3d6c
2006-10-28 13:38:11 [DEBUG] mod_portaudio.c:878 place_call() PortAudio/1234-3d6c
State Change CS_NEW -> CS_INIT
API CALL [pacall(1234)] output:
SUCCESS:5:3204a213-c70b-b04a-8399-1a2d3f9dfe04
freeswitch@Archinov_IBM> 2006-10-28 13:38:11 [DEBUG] switch_core.c:2762 switch_c
ore_session_run() (PortAudio/1234-3d6c) State INIT
2006-10-28 13:38:11 [DEBUG] mod_portaudio.c:185 channel_on_init() PortAudio/1234
-3d6c State Change CS_INIT -> CS_RING
2006-10-28 13:38:11 [DEBUG] switch_core.c:2798 switch_core_session_run() (PortAu
dio/1234-3d6c) State RING
2006-10-28 13:38:11 [DEBUG] mod_portaudio.c:202 channel_on_ring() PortAudio/1234
-3d6c CHANNEL RING
2006-10-28 13:38:11 [DEBUG] switch_core.c:2463 switch_core_standard_on_ring() St
andard RING PortAudio/1234-3d6c
2006-10-28 13:38:11 [INFO] mod_dialplan_xml.c:278 dialplan_hunt() Processing Fre
eSwitch->1234!
------------------------------------------------------------------
2006-10-28 13:38:11 [DEBUG] mod_dialplan_xml.c:199 parse_exten() test conditions
destination_number(1234) =~ /^(18(0{2}|8{2}|7{2}|6{2})\d{7})$/
^(18(0{2}|8{2}|7{2}|6{2})\d{7})$
Length = 78 top_bracket = 2 top_backref = 0
nopartial anchored
0 50 Bra 0
3 ^
4 42 Bra 1
7 18
11 7 Bra 2
14 0{2}
18 7 Alt
21 8{2}
25 7 Alt
28 7{2}
32 7 Alt
35 6{2}
39 28 Ket
42 \d{7}
46 42 Ket
49 $
50 50 Ket
53 End
------------------------------------------------------------------
2006-10-28 13:38:11 [DEBUG] mod_dialplan_xml.c:201 parse_exten() Regex mismatch
------------------------------------------------------------------
2006-10-28 13:38:11 [DEBUG] mod_dialplan_xml.c:199 parse_exten() test conditions
destination_number(1234) =~ /^1234$/
^1234$
Length = 17 top_bracket = 0 top_backref = 0
anchored
0 13 Bra 0
3 ^
4 1234
12 $
13 13 Ket
16 End
------------------------------------------------------------------
2006-10-28 13:38:11 [DEBUG] mod_dialplan_xml.c:326 dialplan_hunt() PortAudio/123
4-3d6c State Change CS_RING -> CS_EXECUTE
2006-10-28 13:38:11 [DEBUG] switch_core.c:2834 switch_core_session_run() (PortAu
dio/1234-3d6c) State EXECUTE
2006-10-28 13:38:11 [DEBUG] mod_portaudio.c:219 channel_on_execute() PortAudio/1
234-3d6c CHANNEL EXECUTE
2006-10-28 13:38:11 [DEBUG] switch_core.c:2497 switch_core_standard_on_execute()
Standard EXECUTE
2006-10-28 13:38:11 [NOTICE] switch_core.c:2506 switch_core_standard_on_execute(
) Execute bridge(sofia/test/1234@66.250.68.194)
2006-10-28 13:38:11 [ERR] mod_sofia.c:1921 sofia_outgoing_channel() Invalid Prof
ile
2006-10-28 13:38:11 [DEBUG] mod_sofia.c:782 terminate_session() Term called from
line: 1922
2006-10-28 13:38:11 [NOTICE] switch_core.c:3012 switch_core_session_destroy() Cl
ose Channel N/A
2006-10-28 13:38:11 [ERR] switch_ivr.c:1879 switch_ivr_originate() Cannot Create
Outgoing Channel!
2006-10-28 13:38:11 [DEBUG] switch_ivr.c:2105 switch_ivr_originate() Originate R
esulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2006-10-28 13:38:11 [ERR] mod_bridgecall.c:59 audio_bridge_function() Cannot Cre
ate Outgoing Channel!
2006-10-28 13:38:11 [NOTICE] mod_bridgecall.c:61 audio_bridge_function() Hangup
PortAudio/1234-3d6c [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2006-10-28 13:38:11 [INFO] switch_channel.c:973 switch_channel_perform_hangup()
Kill PortAudio/1234-3d6c [1]
2006-10-28 13:38:12 [DEBUG] mod_portaudio.c:278 channel_kill_channel() PortAudio
/1234-3d6c CHANNEL KILL
2006-10-28 13:38:12 [DEBUG] switch_core.c:2725 switch_core_session_run() (PortAu
dio/1234-3d6c) State HANGUP
2006-10-28 13:38:12 [DEBUG] mod_portaudio.c:258 channel_on_hangup() PortAudio/12
34-3d6c CHANNEL HANGUP
2006-10-28 13:38:12 [DEBUG] switch_core.c:2453 switch_core_standard_on_hangup()
Standard HANGUP PortAudio/1234-3d6c, cause: DESTINATION_OUT_OF_ORDER
2006-10-28 13:38:12 [DEBUG] switch_core.c:3132 switch_core_session_thread() Sess
ion 6 (PortAudio/1234-3d6c) Locked, Waiting on external entities
2006-10-28 13:38:12 [INFO] switch_core.c:3139 switch_core_session_thread() Sessi
on 6 (PortAudio/1234-3d6c) Ended
2006-10-28 13:38:12 [NOTICE] switch_core.c:3012 switch_core_session_destroy() Cl
ose Channel PortAudio/1234-3d6c
________________________________________________
<configuration name="sofia.conf" description="sofia Endpoint">
<profiles>
<profile name="mydomain1.com">
<registrations>
<!-- <registration name="asterlink">
<param name="register-scheme" value="Digest"/>
<param name="register-realm" value=""/>
<param name="register-username" value="1001"/>
<param name="register-password" value="nhy65tgb"/>
<param name="register-from" value="sip:1001@208.64.200.40"/>
<param name="register-to" value="sip:1001@66.250.68.194"/>
<param name="register-proxy" value="sip:66.250.68.194:5060"/>
<param name="register-frequency" value="20"/>
</registration> -->
</registrations>
<settings>
<param name="debug" value="1"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="PCMU@20i"/>
<param name="codec-ms" value="20"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="rtp-ip" value="192.168.1.20"/>
<param name="sip-ip" value="mydomain1.com"/>
<!-- 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="true"/>
<!--<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.server.com"/>-->
<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
<!-- 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>
</profile>
</profiles>
</configuration>
Thank you,
Alexei Archinov
------------------------------------------------------------------------
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- SCO: hey, where are you guys goin? OS/2: Where did you go yesterday? Microsoft: Where do you want to go today? Linux: Where do you want to go tomorrow? FreeBSD: Are you guys coming or what?