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