[Freeswitch-users] test to 1234 is failing

Alexei Archinov archinov at earthlink.net
Sat Oct 28 10:48:24 PDT 2006


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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