Unable to make call from SipML5 to Softphone.
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Anurag Rana
May 12, 2014, 3:51:16 AM5/12/14
to doub...@googlegroups.com
Hi,
I am trying to call from SipML5 to Softphone (twinkle), User gets connected to FreeSwitch server but is unable to make a call. Status being displayed is "Call in progress..." (Or sometimes "remote ringing"). Call is not received at softphone end.
However call from softphone to SipML5 is successfully made.
No log is produced in FreeSwitch or WebRTC2Sip server log. JavaScript log of SipML5 is paste here. (It either hangs at line # 210 or line # 244)
Please have a look and suggest what could be wrong with setup.
let me know if more info is required.
Thanks
Anurag Rana
I am trying to call from SipML5 to Softphone (twinkle), User gets connected to FreeSwitch server but is unable to make a call. Status being displayed is "Call in progress..." (Or sometimes "remote ringing"). Call is not received at softphone end.
However call from softphone to SipML5 is successfully made.
No log is produced in FreeSwitch or WebRTC2Sip server log. JavaScript log of SipML5 is paste here. (It either hangs at line # 210 or line # 244)
Please have a look and suggest what could be wrong with setup.
let me know if more info is required.
Thanks
Anurag Rana
Anurag Rana
May 13, 2014, 2:20:57 PM5/13/14
to doub...@googlegroups.com
Hi,
I tried after editing the codec list in conf.xml and this time call was successfully made from Sipml5 to softphone (twinkle) but when I tried in reverse order i.e from softphone to sipml5 , call was not made. It says "the user 1003 is not available"
please let me know what should be done in order to make call succesfull in both directions.
------------------------------------------------------------------------------------------
Log of Freeswitch is (green line : call from sipml5 to softphone - success , Red line : call from softphone to sipml5 - failed) ->
2014-05-13 23:33:48.591718 [CONSOLE] switch_core.c:2160
[This app Best viewed at 160x60 or more..]
2014-05-13 23:34:33.885039 [CONSOLE] mod_voicemail.c:4066 Event Thread Started
2014-05-13 23:36:46.685051 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/10...@192.168.49.170 [cacec487-6678-4f71-ae03-9f0c6021422c]
2014-05-13 23:36:46.785052 [INFO] mod_dialplan_xml.c:558 Processing 1003 <1003>->1001 in context default
2014-05-13 23:36:46.785052 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *1 execute_extension::dx XML features
2014-05-13 23:36:46.785052 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1003.2014-05-13-23-36-46.wav
2014-05-13 23:36:46.785052 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *3 execute_extension::cf XML features
2014-05-13 23:36:46.785052 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *4 execute_extension::att_xfer XML features
2014-05-13 23:36:46.805050 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/sip:10...@192.168.49.170:5075 [e40dde6b-1b18-4815-a359-d9e7a1a989e0]
2014-05-13 23:36:46.805050 [NOTICE] sofia.c:6287 Ring-Ready sofia/internal/sip:10...@192.168.49.170:5075!
2014-05-13 23:36:46.825062 [INFO] switch_ivr_originate.c:1191 Sending early media
2014-05-13 23:36:46.825062 [WARNING] switch_core_media.c:2548 NO candidate ACL defined, Defaulting to wan.auto
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.49.170:50562
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 2 proto: udp type: host addr: 192.168.49.170:50563
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2581 Choose audio Candidate cid: 1 proto: udp type: srflx addr: 103.25.231.2:17678
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2581 Choose audio Candidate cid: 2 proto: udp type: srflx addr: 103.25.231.2:50447
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2710 setting remote audio ice addr to 103.25.231.2:17678 based on candidate
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2730 setting remote rtcp audio addr to 103.25.231.2:50447 based on candidate
2014-05-13 23:36:46.825062 [INFO] switch_core_media.c:5010 Activating Audio ICE
2014-05-13 23:36:46.825062 [NOTICE] switch_rtp.c:3775 Activating RTP audio ICE: 2eF89HydvIpyT8K:FOh0s7UFCmf2hSsZ 103.25.231.2:17678
2014-05-13 23:36:46.825062 [INFO] switch_rtp.c:3151 Activating Audio Secure RTP SEND
2014-05-13 23:36:46.825062 [INFO] switch_rtp.c:3129 Activating Audio Secure RTP RECV
2014-05-13 23:36:46.825062 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/10...@192.168.49.170!
2014-05-13 23:36:46.845043 [NOTICE] switch_rtp.c:1132 Auto Changing stun/rtp/dtls port from 103.25.231.2:17678 to 192.168.49.170:50562
2014-05-13 23:36:53.585110 [NOTICE] sofia.c:7025 Channel [sofia/internal/sip:10...@192.168.49.170:5075] has been answered
2014-05-13 23:36:53.585110 [NOTICE] switch_ivr_originate.c:3493 Channel [sofia/internal/10...@192.168.49.170] has been answered
2014-05-13 23:37:14.365051 [NOTICE] sofia.c:927 Hangup sofia/internal/sip:10...@192.168.49.170:5075 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2014-05-13 23:37:14.365051 [NOTICE] switch_ivr_bridge.c:1603 Hangup sofia/internal/10...@192.168.49.170 [CS_EXECUTE] [NORMAL_CLEARING]
2014-05-13 23:37:14.365051 [NOTICE] switch_core_session.c:1622 Session 2 (sofia/internal/sip:10...@192.168.49.170:5075) Ended
2014-05-13 23:37:14.365051 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/sip:10...@192.168.49.170:5075 [CS_DESTROY]
2014-05-13 23:37:14.365051 [NOTICE] switch_core_session.c:1622 Session 1 (sofia/internal/10...@192.168.49.170) Ended
2014-05-13 23:37:14.365051 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/10...@192.168.49.170 [CS_DESTROY]
2014-05-13 23:37:18.545060 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/10...@192.168.49.170 [84d1fbef-ec98-4ed5-925a-6253d70605ec]
2014-05-13 23:37:18.565059 [INFO] mod_dialplan_xml.c:558 Processing 1001 <1001>->1003 in context default
2014-05-13 23:37:18.565059 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *1 execute_extension::dx XML features
2014-05-13 23:37:18.565059 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1001.2014-05-13-23-37-18.wav
2014-05-13 23:37:18.565059 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *3 execute_extension::cf XML features
2014-05-13 23:37:18.585055 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *4 execute_extension::att_xfer XML features
2014-05-13 23:37:18.585055 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/sip:10...@192.168.49.170:10060 [98288ea7-86f1-4340-8ccd-731f233ee125]
2014-05-13 23:37:18.585055 [NOTICE] sofia.c:7080 Hangup sofia/internal/sip:10...@192.168.49.170:10060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
2014-05-13 23:37:18.605061 [NOTICE] switch_core_session.c:1622 Session 4 (sofia/internal/sip:10...@192.168.49.170:10060) Ended
2014-05-13 23:37:18.605061 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/sip:10...@192.168.49.170:10060 [CS_DESTROY]
2014-05-13 23:37:18.605061 [NOTICE] switch_ivr_originate.c:2707 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]
2014-05-13 23:37:18.605061 [INFO] mod_dptools.c:3234 Originate Failed. Cause: INCOMPATIBLE_DESTINATION
2014-05-13 23:37:18.605061 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/10...@192.168.49.170!
2014-05-13 23:37:18.605061 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/10...@192.168.49.170] has been answered
2014-05-13 23:37:19.625059 [NOTICE] switch_channel.c:1054 New Channel loopback/app=voicemail:default 192.168.49.170 1003-a [dcc761ec-5b42-4d56-bdfb-7777aa71430a]
2014-05-13 23:37:19.625059 [NOTICE] switch_channel.c:1052 Rename Channel loopback/app=voicemail:default 192.168.49.170 1003-a->loopback/voicemail-a [dcc761ec-5b42-4d56-bdfb-7777aa71430a]
2014-05-13 23:37:19.625059 [NOTICE] switch_channel.c:1054 New Channel loopback/voicemail-b [731e1315-e9fd-4fc1-80d6-69c1dcf7e6b5]
2014-05-13 23:37:19.625059 [NOTICE] mod_loopback.c:947 Pre-Answer loopback/voicemail-a!
2014-05-13 23:37:19.645085 [NOTICE] mod_dptools.c:1293 Pre-Answer loopback/voicemail-b!
2014-05-13 23:37:34.425055 [NOTICE] sofia.c:927 Hangup sofia/internal/10...@192.168.49.170 [CS_EXECUTE] [NORMAL_CLEARING]
2014-05-13 23:37:34.425055 [NOTICE] switch_ivr_bridge.c:751 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL]
2014-05-13 23:37:34.425055 [NOTICE] mod_loopback.c:553 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL]
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1622 Session 5 (loopback/voicemail-a) Ended
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1626 Close Channel loopback/voicemail-a [CS_DESTROY]
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1622 Session 6 (loopback/voicemail-b) Ended
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1626 Close Channel loopback/voicemail-b [CS_DESTROY]
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1622 Session 3 (sofia/internal/10...@192.168.49.170) Ended
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/10...@192.168.49.170 [CS_DESTROY]
2014-05-13 23:37:43.665050 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/10...@192.168.49.170 [6544be75-2183-415e-8cf0-223c4a5e604c]
-----------------------------------------------------------------------------------
Content of Config.xml file ->
<?xml version="1.0" encoding="utf-8" ?>
<!-- Please check the technical guide (http://webrtc2sip.org/technical-guide-1.0.pdf) for more information on how to adjust this file -->
<config>
<debug-level>ERROR</debug-level>
<transport>udp;*;10060</transport>
<transport>ws;*;10060</transport>
<transport>wss;*;10062</transport>
<!--transport>tcp;*;10063</transport-->
<!--transport>tls;*;10064</transport-->
<enable-rtp-symetric>yes</enable-rtp-symetric>
<enable-100rel>yes</enable-100rel>
<enable-media-coder>yes</enable-media-coder>
<enable-videojb>yes</enable-videojb>
<video-size-pref>vga</video-size-pref>
<rtp-buffsize>65535</rtp-buffsize>
<avpf-tail-length>100;400</avpf-tail-length>
<srtp-mode>mandatory</srtp-mode>
<srtp-type>sdes;dtls</srtp-type>
<dtmf-type>rfc4733</dtmf-type>
<codecs>speex;pcma;pcmu;ilbc;opus;amr</codecs>
<codec-opus-maxrates>48000;48000</codec-opus-maxrates>
<stun-server>stun.l.google.com;19302;stun...@doubango.org;stun-password</stun-server>
<enable-icestun>yes</enable-icestun>
<max-fds>-1</max-fds>
<!--nameserver>66.66.66.6</nameserver-->
<!--ssl-certificates>
C:/Projects/ssl/priv.pem;
C:/Projects/ssl/pub.pem;
C:/Projects/ssl/ca-cert.pem;
</ssl-certificates-->
<!-- ***CLICK-TO-CALL SERVICE*** -->
<transport>c2c;*;10070</transport>
<transport>c2cs;*;10072</transport>
<database>sqlite;*</database>
<!--account-mail>smtps;*;*;auth.smtp.1and1.fr;465;nor...@example.com;nor...@example.com;mysecret</account-mail-->
<!--account-sip-caller>*;sip:a...@example.com;a;example.com;mysecret</account-sip-caller-->
</config>
--------------------------------------------------------------
Webrtc2sip Log ->
SSL is enabled :)
DTLS supported: yes
DTLS-SRTP supported: yes
***ERROR: function: "_sdp_pcfgs_from_sdp()"
file: "src/tdav_session_av.c"
line: "2140"
MSG: Failed to find 'acap' with tag=1
***ERROR: function: "_sdp_pcfgs_from_sdp()"
file: "src/tdav_session_av.c"
line: "2140"
MSG: Failed to find 'acap' with tag=2
***ERROR: function: "_sdp_pcfgs_from_sdp()"
file: "src/tdav_session_av.c"
line: "2140"
MSG: Failed to find 'acap' with tag=1
***ERROR: function: "_sdp_pcfgs_from_sdp()"
file: "src/tdav_session_av.c"
line: "2140"
MSG: Failed to find 'acap' with tag=2
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "928"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
warning: The VAD has been replaced by a hack pending a complete rewrite
***ERROR: function: "_tnet_ice_ctx_fsm_GatheringComplet_2_ConnChecking_X_ConnCheck()"
file: "src/ice/tnet_ice_ctx.c"
line: "1478"
MSG: ConnCheck timedout, have_nominated_answer=false, have_nominated_offer=false
warning: The VAD has been replaced by a hack pending a complete rewrite
***ERROR: function: "tdav_session_av_set_ro()"
file: "src/tdav_session_av.c"
line: "1589"
MSG: SRTP negotiation failed
***ERROR: function: "tdav_session_audio_set_ro()"
file: "src/audio/tdav_session_audio.c"
line: "623"
MSG: tdav_session_av_set_ro(audio) failed
***ERROR: function: "tdav_session_av_set_ro()"
file: "src/tdav_session_av.c"
line: "1589"
MSG: SRTP negotiation failed
***ERROR: function: "tdav_session_audio_set_ro()"
file: "src/audio/tdav_session_audio.c"
line: "623"
MSG: tdav_session_av_set_ro(audio) failed
I tried after editing the codec list in conf.xml and this time call was successfully made from Sipml5 to softphone (twinkle) but when I tried in reverse order i.e from softphone to sipml5 , call was not made. It says "the user 1003 is not available"
please let me know what should be done in order to make call succesfull in both directions.
------------------------------------------------------------------------------------------
Log of Freeswitch is (green line : call from sipml5 to softphone - success , Red line : call from softphone to sipml5 - failed) ->
2014-05-13 23:33:48.591718 [CONSOLE] switch_core.c:2160
[This app Best viewed at 160x60 or more..]
2014-05-13 23:34:33.885039 [CONSOLE] mod_voicemail.c:4066 Event Thread Started
2014-05-13 23:36:46.685051 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/10...@192.168.49.170 [cacec487-6678-4f71-ae03-9f0c6021422c]
2014-05-13 23:36:46.785052 [INFO] mod_dialplan_xml.c:558 Processing 1003 <1003>->1001 in context default
2014-05-13 23:36:46.785052 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *1 execute_extension::dx XML features
2014-05-13 23:36:46.785052 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1003.2014-05-13-23-36-46.wav
2014-05-13 23:36:46.785052 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *3 execute_extension::cf XML features
2014-05-13 23:36:46.785052 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *4 execute_extension::att_xfer XML features
2014-05-13 23:36:46.805050 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/sip:10...@192.168.49.170:5075 [e40dde6b-1b18-4815-a359-d9e7a1a989e0]
2014-05-13 23:36:46.805050 [NOTICE] sofia.c:6287 Ring-Ready sofia/internal/sip:10...@192.168.49.170:5075!
2014-05-13 23:36:46.825062 [INFO] switch_ivr_originate.c:1191 Sending early media
2014-05-13 23:36:46.825062 [WARNING] switch_core_media.c:2548 NO candidate ACL defined, Defaulting to wan.auto
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.49.170:50562
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 2 proto: udp type: host addr: 192.168.49.170:50563
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2581 Choose audio Candidate cid: 1 proto: udp type: srflx addr: 103.25.231.2:17678
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2581 Choose audio Candidate cid: 2 proto: udp type: srflx addr: 103.25.231.2:50447
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2710 setting remote audio ice addr to 103.25.231.2:17678 based on candidate
2014-05-13 23:36:46.825062 [NOTICE] switch_core_media.c:2730 setting remote rtcp audio addr to 103.25.231.2:50447 based on candidate
2014-05-13 23:36:46.825062 [INFO] switch_core_media.c:5010 Activating Audio ICE
2014-05-13 23:36:46.825062 [NOTICE] switch_rtp.c:3775 Activating RTP audio ICE: 2eF89HydvIpyT8K:FOh0s7UFCmf2hSsZ 103.25.231.2:17678
2014-05-13 23:36:46.825062 [INFO] switch_rtp.c:3151 Activating Audio Secure RTP SEND
2014-05-13 23:36:46.825062 [INFO] switch_rtp.c:3129 Activating Audio Secure RTP RECV
2014-05-13 23:36:46.825062 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/10...@192.168.49.170!
2014-05-13 23:36:46.845043 [NOTICE] switch_rtp.c:1132 Auto Changing stun/rtp/dtls port from 103.25.231.2:17678 to 192.168.49.170:50562
2014-05-13 23:36:53.585110 [NOTICE] sofia.c:7025 Channel [sofia/internal/sip:10...@192.168.49.170:5075] has been answered
2014-05-13 23:36:53.585110 [NOTICE] switch_ivr_originate.c:3493 Channel [sofia/internal/10...@192.168.49.170] has been answered
2014-05-13 23:37:14.365051 [NOTICE] sofia.c:927 Hangup sofia/internal/sip:10...@192.168.49.170:5075 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2014-05-13 23:37:14.365051 [NOTICE] switch_ivr_bridge.c:1603 Hangup sofia/internal/10...@192.168.49.170 [CS_EXECUTE] [NORMAL_CLEARING]
2014-05-13 23:37:14.365051 [NOTICE] switch_core_session.c:1622 Session 2 (sofia/internal/sip:10...@192.168.49.170:5075) Ended
2014-05-13 23:37:14.365051 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/sip:10...@192.168.49.170:5075 [CS_DESTROY]
2014-05-13 23:37:14.365051 [NOTICE] switch_core_session.c:1622 Session 1 (sofia/internal/10...@192.168.49.170) Ended
2014-05-13 23:37:14.365051 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/10...@192.168.49.170 [CS_DESTROY]
2014-05-13 23:37:18.545060 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/10...@192.168.49.170 [84d1fbef-ec98-4ed5-925a-6253d70605ec]
2014-05-13 23:37:18.565059 [INFO] mod_dialplan_xml.c:558 Processing 1001 <1001>->1003 in context default
2014-05-13 23:37:18.565059 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *1 execute_extension::dx XML features
2014-05-13 23:37:18.565059 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1001.2014-05-13-23-37-18.wav
2014-05-13 23:37:18.565059 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *3 execute_extension::cf XML features
2014-05-13 23:37:18.585055 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *4 execute_extension::att_xfer XML features
2014-05-13 23:37:18.585055 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/sip:10...@192.168.49.170:10060 [98288ea7-86f1-4340-8ccd-731f233ee125]
2014-05-13 23:37:18.585055 [NOTICE] sofia.c:7080 Hangup sofia/internal/sip:10...@192.168.49.170:10060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
2014-05-13 23:37:18.605061 [NOTICE] switch_core_session.c:1622 Session 4 (sofia/internal/sip:10...@192.168.49.170:10060) Ended
2014-05-13 23:37:18.605061 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/sip:10...@192.168.49.170:10060 [CS_DESTROY]
2014-05-13 23:37:18.605061 [NOTICE] switch_ivr_originate.c:2707 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]
2014-05-13 23:37:18.605061 [INFO] mod_dptools.c:3234 Originate Failed. Cause: INCOMPATIBLE_DESTINATION
2014-05-13 23:37:18.605061 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/10...@192.168.49.170!
2014-05-13 23:37:18.605061 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/10...@192.168.49.170] has been answered
2014-05-13 23:37:19.625059 [NOTICE] switch_channel.c:1054 New Channel loopback/app=voicemail:default 192.168.49.170 1003-a [dcc761ec-5b42-4d56-bdfb-7777aa71430a]
2014-05-13 23:37:19.625059 [NOTICE] switch_channel.c:1052 Rename Channel loopback/app=voicemail:default 192.168.49.170 1003-a->loopback/voicemail-a [dcc761ec-5b42-4d56-bdfb-7777aa71430a]
2014-05-13 23:37:19.625059 [NOTICE] switch_channel.c:1054 New Channel loopback/voicemail-b [731e1315-e9fd-4fc1-80d6-69c1dcf7e6b5]
2014-05-13 23:37:19.625059 [NOTICE] mod_loopback.c:947 Pre-Answer loopback/voicemail-a!
2014-05-13 23:37:19.645085 [NOTICE] mod_dptools.c:1293 Pre-Answer loopback/voicemail-b!
2014-05-13 23:37:34.425055 [NOTICE] sofia.c:927 Hangup sofia/internal/10...@192.168.49.170 [CS_EXECUTE] [NORMAL_CLEARING]
2014-05-13 23:37:34.425055 [NOTICE] switch_ivr_bridge.c:751 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL]
2014-05-13 23:37:34.425055 [NOTICE] mod_loopback.c:553 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL]
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1622 Session 5 (loopback/voicemail-a) Ended
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1626 Close Channel loopback/voicemail-a [CS_DESTROY]
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1622 Session 6 (loopback/voicemail-b) Ended
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1626 Close Channel loopback/voicemail-b [CS_DESTROY]
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1622 Session 3 (sofia/internal/10...@192.168.49.170) Ended
2014-05-13 23:37:34.445046 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/10...@192.168.49.170 [CS_DESTROY]
2014-05-13 23:37:43.665050 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/10...@192.168.49.170 [6544be75-2183-415e-8cf0-223c4a5e604c]
-----------------------------------------------------------------------------------
Content of Config.xml file ->
<?xml version="1.0" encoding="utf-8" ?>
<!-- Please check the technical guide (http://webrtc2sip.org/technical-guide-1.0.pdf) for more information on how to adjust this file -->
<config>
<debug-level>ERROR</debug-level>
<transport>udp;*;10060</transport>
<transport>ws;*;10060</transport>
<transport>wss;*;10062</transport>
<!--transport>tcp;*;10063</transport-->
<!--transport>tls;*;10064</transport-->
<enable-rtp-symetric>yes</enable-rtp-symetric>
<enable-100rel>yes</enable-100rel>
<enable-media-coder>yes</enable-media-coder>
<enable-videojb>yes</enable-videojb>
<video-size-pref>vga</video-size-pref>
<rtp-buffsize>65535</rtp-buffsize>
<avpf-tail-length>100;400</avpf-tail-length>
<srtp-mode>mandatory</srtp-mode>
<srtp-type>sdes;dtls</srtp-type>
<dtmf-type>rfc4733</dtmf-type>
<codecs>speex;pcma;pcmu;ilbc;opus;amr</codecs>
<codec-opus-maxrates>48000;48000</codec-opus-maxrates>
<stun-server>stun.l.google.com;19302;stun...@doubango.org;stun-password</stun-server>
<enable-icestun>yes</enable-icestun>
<max-fds>-1</max-fds>
<!--nameserver>66.66.66.6</nameserver-->
<!--ssl-certificates>
C:/Projects/ssl/priv.pem;
C:/Projects/ssl/pub.pem;
C:/Projects/ssl/ca-cert.pem;
</ssl-certificates-->
<!-- ***CLICK-TO-CALL SERVICE*** -->
<transport>c2c;*;10070</transport>
<transport>c2cs;*;10072</transport>
<database>sqlite;*</database>
<!--account-mail>smtps;*;*;auth.smtp.1and1.fr;465;nor...@example.com;nor...@example.com;mysecret</account-mail-->
<!--account-sip-caller>*;sip:a...@example.com;a;example.com;mysecret</account-sip-caller-->
</config>
--------------------------------------------------------------
Webrtc2sip Log ->
SSL is enabled :)
DTLS supported: yes
DTLS-SRTP supported: yes
***ERROR: function: "_sdp_pcfgs_from_sdp()"
file: "src/tdav_session_av.c"
line: "2140"
MSG: Failed to find 'acap' with tag=1
***ERROR: function: "_sdp_pcfgs_from_sdp()"
file: "src/tdav_session_av.c"
line: "2140"
MSG: Failed to find 'acap' with tag=2
***ERROR: function: "_sdp_pcfgs_from_sdp()"
file: "src/tdav_session_av.c"
line: "2140"
MSG: Failed to find 'acap' with tag=1
***ERROR: function: "_sdp_pcfgs_from_sdp()"
file: "src/tdav_session_av.c"
line: "2140"
MSG: Failed to find 'acap' with tag=2
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "928"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
warning: The VAD has been replaced by a hack pending a complete rewrite
***ERROR: function: "_tnet_ice_ctx_fsm_GatheringComplet_2_ConnChecking_X_ConnCheck()"
file: "src/ice/tnet_ice_ctx.c"
line: "1478"
MSG: ConnCheck timedout, have_nominated_answer=false, have_nominated_offer=false
warning: The VAD has been replaced by a hack pending a complete rewrite
***ERROR: function: "tdav_session_av_set_ro()"
file: "src/tdav_session_av.c"
line: "1589"
MSG: SRTP negotiation failed
***ERROR: function: "tdav_session_audio_set_ro()"
file: "src/audio/tdav_session_audio.c"
line: "623"
MSG: tdav_session_av_set_ro(audio) failed
***ERROR: function: "tdav_session_av_set_ro()"
file: "src/tdav_session_av.c"
line: "1589"
MSG: SRTP negotiation failed
***ERROR: function: "tdav_session_audio_set_ro()"
file: "src/audio/tdav_session_audio.c"
line: "623"
MSG: tdav_session_av_set_ro(audio) failed
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