[Freeswitch-users] test to 1234 is failing

Sat Oct 28 11:08:05 PDT 2006

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 at 66.250.68.194
not
sofia/test//1234 at 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 at 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 at 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 at 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 at 208.64.200.40"/>
>
> <param name="register-to" value="sip:1001 at 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 at 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
>
>------------------------------------------------------------------------
>
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>  
>

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