Unable to register sip user in freeswitch after installing bigbluebutton

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Prashanth Francis

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Jun 23, 2014, 3:46:51 AM6/23/14
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I have made a fresh install of bigbluebutton on my ubuntu 10.04 using http://code.google.com/p/bigbluebutton/wiki/InstallationUbuntu . In order to register a sip user using a softphone(zoiper) into this server I've created a xml file for  user 1100 in the location /opt/freeswitch/conf/directory/default . This is what it looks like

<include>
  <user id="1100">
    <params>
      <param name="password" value="pass1100"/>
      <param name="vm-password" value="1100"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1100"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="1100"/>
      <variable name="effective_caller_id_number" value="1100"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

And this is how my vars.xml in /opt/freeswitch/conf looks like

<include>
  <!-- Preprocessor Variables
       These are introduced when configuration strings must be consistent across modules.
       NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.
      
       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
      
       YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
       toll fraud in the future.  It's your responsibility to secure your own system.
      
       This default config is used to demonstrate the feature set of FreeSWITCH.
      
       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
  -->
  <X-PRE-PROCESS cmd="set" data="default_password=1234"/>
  <!-- Did you change it yet? -->

  <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>

  <!--
      This setting is what sets the default domain FreeSWITCH will use if all else fails.
     
      FreeSWICH will default to $${local_ip_v4} unless changed.  Changing this setting does
      affect the sip authentication.  Please review conf/directory/default.xml for more
      information on this topic.
  -->
  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.15"/>

  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
  <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
  <X-PRE-PROCESS cmd="set" data="use_profile=internal"/>

  <!--
      Enable ZRTP globally you can override this on a per channel basis
     
      http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
  -->
  <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>

  <!--
       Examples of codec options: (module must be compiled and loaded)
      
       codecname[@8000h|16000h|32000h[@XXi]]
      
       XX is the frame size must be multples allowed for the codec
       FreeSWITCH can support 10-120ms on some codecs.
       We do not support exceeding the MTU of the RTP packet.


       iLBC@30i         - iLBC using mode=30 which will win in all cases.
       DVI4@8000h@20i   - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10)
       DVI4@16000h@40i  - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10)
       speex@8000h@20i  - Speex 8kHz using 20ms ptime.
       speex@16000h@20i - Speex 16kHz using 20ms ptime.
       speex@32000h@20i - Speex 32kHz using 20ms ptime.
       BV16             - BroadVoice 16kb/s narrowband, 8kHz
       BV32             - BroadVoice 32kb/s wideband, 16kHz
       G7221@16000h     - G722.1 16kHz (aka Siren 7)
       G7221@32000h     - G722.1C 32kHz (aka Siren 14)
       CELT@32000h      - CELT 32kHz, only 10ms supported
       CELT@48000h      - CELT 48kHz, only 10ms supported
       GSM@40i          - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms)
       G722             - G722 16kHz using default 20ms ptime. (multiples of 10)
       PCMU             - G711 8kHz ulaw using default 20ms ptime. (multiples of 10)
       PCMA             - G711 8kHz alaw using default 20ms ptime. (multiples of 10)
       G726-16          - G726 16kbit adpcm using default 20ms ptime. (multiples of 10)
       G726-24          - G726 24kbit adpcm using default 20ms ptime. (multiples of 10)
       G726-32          - G726 32kbit adpcm using default 20ms ptime. (multiples of 10)
       G726-40          - G726 40kbit adpcm using default 20ms ptime. (multiples of 10)
       AAL2-G726-16     - Same as G726-16 but using AAL2 packing. (multiples of 10)
       AAL2-G726-24     - Same as G726-24 but using AAL2 packing. (multiples of 10)
       AAL2-G726-32     - Same as G726-32 but using AAL2 packing. (multiples of 10)
       AAL2-G726-40     - Same as G726-40 but using AAL2 packing. (multiples of 10)
       LPC              - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)
       L16              - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.
      
       These are the passthru audio codecs:
      
       G729             - G729 in passthru mode. (mod_g729)
       G723             - G723.1 in passthru mode. (mod_g723_1)
       AMR              - AMR in passthru mode. (mod_amr)
      
       These are the passthru video codecs: (mod_h26x)
      
       H261             - H.261 Video
       H263             - H.263 Video
       H263-1998        - H.263-1998 Video
       H263-2000        - H.263-2000 Video
       H264             - H.264 Video
      
       RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for.

       96  - AMR
       97  - iLBC (30)
       98  - iLBC (20)
       99  - Speex 8kHz, 16kHz, 32kHz
       100 -
       101 - telephone-event
       102 -
       103 -
       104 -
       105 -
       106 - BV16
       107 - G722.1 (16kHz)
       108 -
       109 -
       110 -
       111 -
       112 -
       113 -
       114 - CELT 32kHz, 48kHz
       115 - G722.1C (32kHz)
       116 -
       117 - SILK 8kHz
       118 - SILK 12kHz
       119 - SILK 16kHz
       120 - SILK 24kHz
       121 - AAL2-G726-40 && G726-40
       122 - AAL2-G726-32 && G726-32
       123 - AAL2-G726-24 && G726-24
       124 - AAL2-G726-16 && G726-16
       125 -
       126 -
       127 - BV32

  -->

        <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@16000h@20i,speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM" />
        <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@16000h@20i,PCMU,PCMA,GSM" />
    <!-- BigBlueButton: To use mlaw , change the above two lines with data set to these values.
        data="global_codec_prefs=PCMU,G722,PCMA,GSM"
        data="outbound_codec_prefs=PCMU,G722,PCMA,GSM"
    -->

  <!--
      xmpp_client_profile and xmpp_server_profile
      xmpp_client_profile can be any string.
      xmpp_server_profile is appended to "dingaling_" to form the database name
      containing the "subscriptions" table.
      used by: dingaling.conf.xml enum.conf.xml
  -->

  <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
  <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
  <!--
       THIS IS ONLY USED FOR DINGALING

       bind_server_ip

       Can be an ip address, a dns name, or "auto".
       This determines an ip address available on this host to bind.
       If you are separating RTP and SIP traffic, you will want to have
       use different addresses where this variable appears.
       Used by: dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>

  <!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
      
       If you're going to load test FreeSWITCH please input real IP addresses
       for external_rtp_ip and external_sip_ip
  -->

  <!-- external_rtp_ip
       Can be an one of:
           ip address: "12.34.56.78"
           a stun server lookup: "stun:stun.server.com"
           a DNS name: "host:host.server.com"
       where fs.mydomain.com is a DNS A record-useful when fs is on
       a dynamic IP address, and uses a dynamic DNS updater.
       If unspecified, the bind_server_ip value is used.
       Used by: sofia.conf.xml dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>

  <!-- external_sip_ip
      Used as the public IP address for SDP.
       Can be an one of:
           ip address: "12.34.56.78"
           a stun server lookup: "stun:stun.server.com"
           a DNS name: "host:host.server.com"
       where fs.mydomain.com is a DNS A record-useful when fs is on
       a dynamic IP address, and uses a dynamic DNS updater.
       If unspecified, the bind_server_ip value is used.
       Used by: sofia.conf.xml dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>

  <!-- unroll-loops
       Used to turn on sip loopback unrolling.
  -->
  <X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>

  <!-- outbound_caller_id and outbound_caller_name
       The caller ID telephone number we should use when calling out.
       Used by: conference.conf.xml and user directory for default
       outbound callerid name and number.
  -->
  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>

  <!-- various debug and defaults -->
  <X-PRE-PROCESS cmd="set" data="call_debug=false"/>
  <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
  <X-PRE-PROCESS cmd="set" data="default_areacode=918"/>
  <X-PRE-PROCESS cmd="set" data="default_country=US"/>

  <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/>
  <X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440.0,480.0)"/>
  <X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440.0,0.0)"/>
  <X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425.0,0.0)"/>
  <X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425,0)"/>
  <X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
  <X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
  <!--
      Setting up your default sip provider is easy.
      Below are some values that should work in most cases.
     
      These are for conf/directory/default/example.com.xml
  -->
  <X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_password=password"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/>
  <!-- true or false -->
  <X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>

  <!--
      SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls
  -->
  <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5090"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>

  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>
</include>

And this is a copy of  /usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties



# The address of your FreeSWITCH/asterisk server
sip.server.host=192.168.0.15
sip.server.port=5070
sip.server.username=bbbuser
sip.server.password=secret

# The start/stop RTP port the application is going to use
# for the media stream.
# NOTE: This will also be used as SIP users to REGISTER with
# Asterisk. Therefore, make sure you have this range of users
# in your bbb_sip.conf.
# See http://code.google.com/p/bigbluebutton/source/browse/#svn/trunk/bbb-voice-conference/config/asterisk
# create-sip-users.sh script to create the users.
startAudioPort=15000
stopAudioPort=16383

# An extension pattern, in case your asterisk extensions.conf
# uses a naming convetion for your meeting rooms
# e.g. conf-85115 instead of just 85115       
callExtensionPattern={0}

# If you want mjsip stack (red5/log/*access*.log) to minimize the amount of logs it
# generates, set this to a lower value (e.g. 3).
sipStackDebugLevel=3





As I said I'm unable to register the user from my internal network which is in 192.168.0.0 range, I've even tried from the ubuntu machine itself after installing a client on it. I've also  disabled the firewall of the server but its all the same. Is ther any other configuration changes required to get this connected? Please help.

Richard Alam

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Jun 23, 2014, 4:06:34 PM6/23/14
to BigBlueButton-dev
On Mon, Jun 23, 2014 at 3:46 AM, Prashanth Francis <prash...@telenovanetworks.com> wrote:
I have made a fresh install of bigbluebutton on my ubuntu 10.04 using http://code.google.com/p/bigbluebutton/wiki/InstallationUbuntu . In order to register a sip user using a softphone(zoiper) into this server I've created a xml file for  user 1100 in the location /opt/freeswitch/conf/directory/default . This is what it looks like

<include>
  <user id="1100">
    <params>
      <param name="password" value="pass1100"/>
      <param name="vm-password" value="1100"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1100"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="1100"/>
      <variable name="effective_caller_id_number" value="1100"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>


<snip>
 

As I said I'm unable to register the user from my internal network which is in 192.168.0.0 range, I've even tried from the ubuntu machine itself after installing a client on it. I've also  disabled the firewall of the server but its all the same. Is ther any other configuration changes required to get this connected? Please help.

How did you configure Zoiper? Have you configured it to register as user 1100?

Richard

 

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Prashanth Francis

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Jun 24, 2014, 1:58:13 AM6/24/14
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Thank you Richard for replying, these are the values which I gave

user name - 11...@192.168.0.15
password - pass1100
domain - 192.168.0.15:5060

It didn't work, but later on when I tried to register into a zoiper installed on the same server where bigbluebutton is installed, it got registered with these

username - 11...@192.168.0.15
password- pass1100
domain - 127.0.0.1:5060

I'm still not able to log in from my internal network, not to mention that I haven't even started to try and make it work from outside my network. Please help.


Richard Alam

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Jun 24, 2014, 11:03:25 AM6/24/14
to BigBlueButton-dev
Ok...let's back up for a minute.

Which fails to register to FS after you made the changes? BBB SIP or Zoiper? Do you have firewall installed in your server? Check iptables.


On Mon, Jun 23, 2014 at 3:46 AM, Prashanth Francis <prash...@telenovanetworks.com> wrote:
I have made a fresh install of bigbluebutton on my ubuntu 10.04 using http://code.google.com/p/bigbluebutton/wiki/InstallationUbuntu . In order to register a sip user using a softphone(zoiper) into this server I've created a xml file for  user 1100 in the location /opt/freeswitch/conf/directory/default . This is what it looks like

<include>
  <user id="1100">
    <params>
      <param name="password" value="pass1100"/>
      <param name="vm-password" value="1100"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1100"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="1100"/>
      <variable name="effective_caller_id_number" value="1100"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

The above looks ok for Zoiper account.
 

And this is how my vars.xml in /opt/freeswitch/conf looks like

<include>
  <!-- Preprocessor Variables
       These are introduced when configuration strings must be consistent across modules.
       NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.
      
       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
      
       YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
       toll fraud in the future.  It's your responsibility to secure your own system.
      
       This default config is used to demonstrate the feature set of FreeSWITCH.
      
       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
  -->
  <X-PRE-PROCESS cmd="set" data="default_password=1234"/>
  <!-- Did you change it yet? -->

  <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>

  <!--
      This setting is what sets the default domain FreeSWITCH will use if all else fails.
     
      FreeSWICH will default to $${local_ip_v4} unless changed.  Changing this setting does
      affect the sip authentication.  Please review conf/directory/default.xml for more
      information on this topic.
  -->
  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.15"/>

This should only be the only change you need to make in this file.
 

  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
  <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
  <X-PRE-PROCESS cmd="set" data="use_profile=internal"/>

And this is a copy of  /usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties



# The address of your FreeSWITCH/asterisk server
sip.server.host=192.168.0.15
sip.server.port=5070
sip.server.username=bbbuser
sip.server.password=secret

# The start/stop RTP port the application is going to use
# for the media stream.
# NOTE: This will also be used as SIP users to REGISTER with
# Asterisk. Therefore, make sure you have this range of users
# in your bbb_sip.conf.
# See http://code.google.com/p/bigbluebutton/source/browse/#svn/trunk/bbb-voice-conference/config/asterisk
# create-sip-users.sh script to create the users.
startAudioPort=15000
stopAudioPort=16383

# An extension pattern, in case your asterisk extensions.conf
# uses a naming convetion for your meeting rooms
# e.g. conf-85115 instead of just 85115       
callExtensionPattern={0}

# If you want mjsip stack (red5/log/*access*.log) to minimize the amount of logs it
# generates, set this to a lower value (e.g. 3).
sipStackDebugLevel=3



What BBB version do you have installed? This config looks old.

Here is the 0.81 release.

 
As I said I'm unable to register the user from my internal network which is in 192.168.0.0 range, I've even tried from the ubuntu machine itself after installing a client on it. I've also  disabled the firewall of the server but its all the same. Is ther any other configuration changes required to get this connected? Please help.


Restart BBB "sudo bbb-conf --restart" and make sure you can call in using BBB by clicking on the headset icon.

Try with Zoiper, it should be able to register. If not, enable SIP message tracing on FreeSWITCH if you even receive the REGISTER message.

Richard
 

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Prashanth Francis

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Jun 25, 2014, 1:53:20 AM6/25/14
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The only change made in vars.xml is the one you mentioned and the iptables are disabled. Should I update this version of bigbluebutton? Because when I click on the headset icon it shows an option to share the microphone, nothing else. Does the registration have anything to do with the external.xml and internal.xml files?

Please take a look at these files.


/opt/freeswitch/conf/sip_profiles/external.xml



<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <!--
    <alias name="outbound"/>
    <alias name="nat"/>
    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
    <!-- <param name="shutdown-on-fail" value="true"/> -->
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="5060"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!--<param name="enable-100rel" value="true"/>-->
    <!-- This could be set to "passive" -->
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>

    <!-- used to share presence info across sofia profiles
     manage-presence needs to be set to passive on this profile
     if you want it to behave as if it were the internal profile
     for presence.
    -->
    <!-- Name of the db to use for this profile -->
    <!--<param name="dbname" value="share_presence"/>-->
    <!--<param name="presence-hosts" value="$${domain}"/>-->
    <!--<param name="force-register-domain" value="$${domain}"/>-->
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
    <!-- ************************************************* -->

    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <!--
    DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
    -->
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="192.168.0.15"/>
    <param name="ext-rtp-ip" value="$${local_ip_v4}"/>
    <param name="ext-sip-ip" value="192.168.0.15"/>
    <!--
       Set the RTP timeout for 3 hours to prevent FS from hanging up bbb clients
       who are muted and not sending any audio.
    -->
    <param name="rtp-timeout-sec" value="10800"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <!--<param name="enable-3pcc" value="true"/>-->

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${external_ssl_enable}"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
</profile>



/opt/freeswitch/conf/sip_profiles/internal.xml


<profile name="internal">
  <!--
      This is a sofia sip profile/user agent.  This will service exactly one ip and port.
      In FreeSWITCH you can run multiple sip user agents on their own ip and port.
     
      When you hear someone say "sofia profile" this is what they are talking about.
  -->
 
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!--aliases are other names that will work as a valid profile name for this profile-->
  <aliases>
    <!--
    <alias name="default"/>
    -->
  </aliases>
  <!-- Outbound Registrations -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
  </gateways>
 
  <domains>
    <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
    <!--<domain name="$${domain}" parse="true"/>-->
    <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
    <!--<domain name="all" alias="true" parse="true"/>-->
    <domain name="all" alias="true" parse="false"/>
  </domains>
 
  <settings>
    <!--
    When calls are in no media this will bring them back to media
    when you press the hold button.
    -->
    <!--<param name="media-option" value="resume-media-on-hold"/> -->
    <!--
    This will allow a call after an attended transfer go back to
     bypass media after an attended transfer.
    -->
    <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
    <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
    <param name="debug" value="0"/>
    <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
    <!-- <param name="shutdown-on-fail" value="true"/> -->
    <param name="sip-trace" value="no"/>
    <param name="log-auth-failures" value="true"/>
    <param name="context" value="public"/>
    <param name="rfc2833-pt" value="101"/>
    <!-- port to bind to for sip traffic -->
    <param name="sip-port" value="5060"/>
    <param name="dialplan" value="XML"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
    <param name="sip-ip" value="192.168.0.15"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="apply-nat-acl" value="nat.auto"/>

    <!-- extended info parsing -->
    <!-- <param name="extended-info-parsing" value="true"/> -->

    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <!--
    There are known issues (asserts and segfaults) when 100rel is enabled.
    It is not recommended to enable 100rel at this time.
    -->
    <!--<param name="enable-100rel" value="true"/>-->
    <!-- Enable Compact SIP headers. -->
    <!--<param name="enable-compact-headers" value="true"/>-->
    <!--
    enable/disable session timers
    -->
    <!--<param name="enable-timer" value="false"/>-->
    <!--<param name="minimum-session-expires" value="120"/>-->
    <param name="apply-inbound-acl" value="domains"/>
    <!--
    This defines your local network, by default we detect your local network
    and create this localnet.auto ACL for this.
    -->
    <param name="local-network-acl" value="localnet.auto"/>
    <!--<param name="apply-register-acl" value="domains"/>-->
    <!--<param name="dtmf-type" value="info"/>-->


    <!-- 'true' means every time 'first-only' means on the first register -->
    <!--<param name="send-message-query-on-register" value="true"/>-->
   
     

    <!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
    <!-- Remote-Party-ID header -->
    <!--<param name="caller-id-type" value="rpid"/>-->

    <!-- P-*-Identity family of headers -->
    <!--<param name="caller-id-type" value="pid"/>-->

    <!-- neither one -->
    <!--<param name="caller-id-type" value="none"/>-->



    <param name="record-path" value="$${recordings_dir}"/>
    <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
    <!--enable to use presence -->
    <param name="manage-presence" value="true"/>
    <!--<param name="manage-shared-appearance" value="true"/>-->
    <!-- used to share presence info across sofia profiles -->
    <!-- Name of the db to use for this profile -->
    <!--<param name="dbname" value="share_presence"/>-->
    <!--<param name="presence-hosts" value="$${domain}"/>-->
    <!-- ************************************************* -->
   
    <!-- This setting is for AAL2 bitpacking on G726 -->
    <!-- <param name="bitpacking" value="aal2"/> -->
    <!--max number of open dialogs in proceeding -->
    <!--<param name="max-proceeding" value="1000"/>-->
    <!--session timers for all call to expire after the specified seconds -->
    <!--<param name="session-timeout" value="1800"/>-->
    <!-- Can be 'true' or 'contact' -->
    <!--<param name="multiple-registrations" value="contact"/>-->
    <!--set to 'greedy' if you want your codec list to take precedence -->
    <param name="inbound-codec-negotiation" value="generous"/>
    <!-- if you want to send any special bind params of your own -->
    <!--<param name="bind-params" value="transport=udp"/>-->
    <!--<param name="unregister-on-options-fail" value="true"/>-->

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${internal_ssl_enable}"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
    <param name="tls-sip-port" value="$${internal_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
    <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

    <!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
     (reduces delay on latent connections default true, must be disabled explicitly)-->
    <!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
   
    <!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
    <!--<param name="rtp-rewrite-timestamps" value="true"/>-->
    <!--<param name="pass-rfc2833" value="true"/>-->
    <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
   
    <!--Uncomment to set all inbound calls to no media mode-->
    <!--<param name="inbound-bypass-media" value="true"/>-->

    <!--Uncomment to set all inbound calls to proxy media mode-->
    <!--<param name="inbound-proxy-media" value="true"/>-->
   
    <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
    <!--<param name="inbound-late-negotiation" value="true"/>-->
   
    <!-- 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"/> -->

    <!-- accept any authentication without actually checking (not a good feature for most people) -->
    <!-- <param name="accept-blind-auth" value="true"/> -->
   
    <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
    <!-- <param name="suppress-cng" value="true"/> -->
   
    <!--TTL for nonce in sip auth-->
    <param name="nonce-ttl" value="60"/>
    <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
    that the originator is using-->
    <!--<param name="disable-transcoding" value="true"/>-->
    <!-- Handle 302 Redirect in the dialplan -->
    <!--<param name="manual-redirect" value="true"/> -->
    <!-- Disable Transfer -->
    <!--<param name="disable-transfer" value="true"/> -->
    <!-- Disable Register -->
    <!--<param name="disable-register" value="true"/> -->
    <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
    <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
    <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
    <param name="auth-calls" value="$${internal_auth_calls}"/>
    <!-- Force the user and auth-user to match. -->
    <param name="inbound-reg-force-matching-username" value="true"/>
    <!-- on authed calls, authenticate *all* the packets not just invite -->
    <param name="auth-all-packets" value="false"/>

   
    <!-- external_sip_ip
      Used as the public IP address for SDP.
      Can be an one of:
           ip address            - "12.34.56.78"
           a stun server lookup  - "stun:stun.server.com"
           a DNS name            - "host:host.server.com"
           auto                  - Use guessed ip.
           auto-nat              - Use ip learned from NAT-PMP or UPNP
       -->
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>

    <!-- rtp inactivity timeout -->
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <!-- 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"/>-->
    <!--
    These are enabled to make the default config work better out of the box.
    If you need more than ONE domain you'll need to not use these options.

    -->
    <!--all inbound reg will look in this domain for the users -->
    <param name="force-register-domain" value="$${domain}"/>
    <!--force the domain in subscriptions to this value -->
    <param name="force-subscription-domain" value="$${domain}"/>
    <!--all inbound reg will stored in the db using this domain -->
    <param name="force-register-db-domain" value="$${domain}"/>

    <!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
    <!--<param name="rtcp-audio-interval-msec" value="5000"/>-->
    <!--<param name="rtcp-video-interval-msec" value="5000"/>-->

    <!--force suscription expires to a lower value than requested-->
    <!--<param name="force-subscription-expires" value="60"/>-->
    <!-- disable register and transfer which may be undesirable in a public switch -->
    <!--<param name="disable-transfer" value="true"/>-->
    <!--<param name="disable-register" value="true"/>-->

    <!--
     enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
     right away, proxy waits until the call has been answered then sends accepts
    -->
    <!--<param name="enable-3pcc" value="true"/>-->
   
    <!-- use at your own risk or if you know what this does.-->
    <!--<param name="NDLB-force-rport" value="true"/>-->
    <!--
    Choose the realm challenge key. Default is auto_to if not set.
   
    auto_from  - uses the from field as the value for the sip realm.
    auto_to    - uses the to field as the value for the sip realm.
    <anyvalue> - you can input any value to use for the sip realm.

    If you want URL dialing to work you'll want to set this to auto_from.
   
    If you use any other value besides auto_to or auto_from you'll loose
    the ability to do multiple domains.
   
    Note: comment out to restore the behavior before 2008-09-29

    -->
    <param name="challenge-realm" value="auto_from"/>
    <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
    <!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
    <!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
    <!-- on outbound calls set the callid to match the uuid of the session -->
    <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
    <!-- set to false disable this feature -->
    <!--<param name="rtp-autofix-timing" value="false"/>-->

    <!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
    <!--<param name="pass-callee-id" value="false"/>-->

    <!-- clear clears them all or supply the name to add or the name prefixed with ~ to remove
     valid values:

     clear
     CISCO_SKIP_MARK_BIT_2833
     SONUS_SEND_INVALID_TIMESTAMP_2833

    -->
    <!--<param name="auto-rtp-bugs" data="clear"/>-->

     <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
     <!--<param name="disable-srv" value="false" />-->
     <!--<param name="disable-naptr" value="false" />-->

    <!-- The following can be used to fine-tune timers within sofia's transport layer
         Those settings are for advanced users and can safely be left as-is -->
       
    <!-- Initial retransmission interval (in milliseconds).
        Set the T1 retransmission interval used by the SIP transaction engine.
        The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G.      -->
    <!-- <param name="timer-T1" value="500" /> -->
   
    <!--  Transaction timeout (defaults to T1 * 64).
        Set the T1x64 timeout value used by the SIP transaction engine.
        The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
        The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
    <!-- <param name="timer-T1X64" value="32000" /> -->
   
   
    <!-- Maximum retransmission interval (in milliseconds).
        Set the maximum retransmission interval used by the SIP transaction engine.
        The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
        Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
        until the timer B fires.  -->
    <!-- <param name="timer-T2" value="4000" /> -->
       
    <!--
        Transaction lifetime (in milliseconds).
        Set the lifetime for completed transactions used by the SIP transaction engine.
        A completed transaction is kept around for the duration of T4 in order to catch late responses.
        The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
    <!-- <param name="timer-T4" value="4000" /> -->
   
  </settings>
</profile>










Prashanth Francis

unread,
Jun 25, 2014, 2:07:54 AM6/25/14
to bigblueb...@googlegroups.com

Richard, thank you very much! It got registered after restarting bbb, So how do I join a conference after I register a user on a softphone?

Prashanth Francis

unread,
Jun 25, 2014, 5:30:47 AM6/25/14
to bigblueb...@googlegroups.com
I created two extensions and registered on two softphones(x-lite) installed on 2 pcs in my network and tried calling,but it wasnt going through(firewalls on both pcs are turned off). This is the debug information from fs_cli

2014-06-25 14:22:24.714795 [NOTICE] switch_channel.c:808 New Channel sofia/external/11...@192.168.0.15 [0708870a-fc46-11e3-b398-d9722d5f6f3e]
2014-06-25 14:22:24.715793 [DEBUG] sofia.c:4624 Channel sofia/external/11...@192.168.0.15 entering state [received][100]
2014-06-25 14:22:24.715793 [DEBUG] sofia.c:4635 Remote SDP:
v=0
o=- 13048159638044127 1 IN IP4 192.168.0.19
s=X-Lite 4 release 4.5.5  stamp 71236
c=IN IP4 192.168.0.19
t=0 0
m=audio 55876 RTP/AVP 125 9 0 8 100 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2014-06-25 14:22:24.715793 [DEBUG] switch_core_state_machine.c:320 (sofia/external/11...@192.168.0.15) Running State Change CS_NEW
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [opus:125:48000:20:0]/[SPEEX:99:16000:20:42200]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [opus:125:48000:20:0]/[SPEEX:99:8000:20:24600]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [opus:125:48000:20:0]/[G7221:115:32000:20:48000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [opus:125:48000:20:0]/[G7221:107:16000:20:32000]
2014-06-25 14:22:24.715793 [DEBUG] switch_core_state_machine.c:338 (sofia/external/11...@192.168.0.15) State NEW
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [opus:125:48000:20:0]/[PCMU:0:8000:20:64000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [opus:125:48000:20:0]/[PCMA:8:8000:20:64000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [G722:9:8000:20:64000]/[SPEEX:99:16000:20:42200]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [G722:9:8000:20:64000]/[SPEEX:99:8000:20:24600]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [G722:9:8000:20:64000]/[G7221:115:32000:20:48000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [G722:9:8000:20:64000]/[G7221:107:16000:20:32000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [G722:9:8000:20:64000]/[PCMU:0:8000:20:64000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [G722:9:8000:20:64000]/[PCMA:8:8000:20:64000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:16000:20:42200]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:8000:20:24600]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4442 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:2750 Set Codec sofia/external/11...@192.168.0.15 PCMU/8000 20 ms 160 samples 64000 bits
2014-06-25 14:22:24.715793 [DEBUG] sofia_glue.c:4546 Set 2833 dtmf send/recv payload to 101
2014-06-25 14:22:24.715793 [DEBUG] sofia.c:4802 (sofia/external/11...@192.168.0.15) State Change CS_NEW -> CS_INIT
2014-06-25 14:22:24.715793 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/11...@192.168.0.15 [BREAK]
2014-06-25 14:22:24.716794 [DEBUG] switch_core_state_machine.c:320 (sofia/external/11...@192.168.0.15) Running State Change CS_INIT
2014-06-25 14:22:24.716794 [DEBUG] switch_core_state_machine.c:356 (sofia/external/11...@192.168.0.15) State INIT
2014-06-25 14:22:24.716794 [DEBUG] mod_sofia.c:84 sofia/external/11...@192.168.0.15 SOFIA INIT
2014-06-25 14:22:24.716794 [DEBUG] mod_sofia.c:124 (sofia/external/11...@192.168.0.15) State Change CS_INIT -> CS_ROUTING
2014-06-25 14:22:24.716794 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/11...@192.168.0.15 [BREAK]
2014-06-25 14:22:24.716794 [DEBUG] switch_core_state_machine.c:356 (sofia/external/11...@192.168.0.15) State INIT going to sleep
2014-06-25 14:22:24.716794 [DEBUG] switch_core_state_machine.c:320 (sofia/external/11...@192.168.0.15) Running State Change CS_ROUTING
2014-06-25 14:22:24.716794 [DEBUG] switch_channel.c:1657 (sofia/external/11...@192.168.0.15) Callstate Change DOWN -> RINGING
2014-06-25 14:22:24.716794 [DEBUG] switch_core_state_machine.c:359 (sofia/external/11...@192.168.0.15) State ROUTING
2014-06-25 14:22:24.716794 [DEBUG] mod_sofia.c:147 sofia/external/11...@192.168.0.15 SOFIA ROUTING
2014-06-25 14:22:24.716794 [DEBUG] switch_core_state_machine.c:77 sofia/external/11...@192.168.0.15 Standard ROUTING
2014-06-25 14:22:24.716794 [INFO] mod_dialplan_xml.c:331 Processing 1100 <1100>->1200 in context public
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->unloop] continue=false
Dialplan: sofia/external/11...@192.168.0.15 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/11...@192.168.0.15 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->outside_call] continue=true
Dialplan: sofia/external/11...@192.168.0.15 Absolute Condition [outside_call]
Dialplan: sofia/external/11...@192.168.0.15 Action set(outside_call=true)
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->call_debug] continue=true
Dialplan: sofia/external/11...@192.168.0.15 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/11...@192.168.0.15 Regex (FAIL) [public_extensions] destination_number(1200) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->bbb_conferences] continue=false
Dialplan: sofia/external/11...@192.168.0.15 Regex (FAIL) [bbb_conferences] destination_number(1200) =~ /^(\d{5})$/ break=on-false
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->public_did] continue=false
Dialplan: sofia/external/11...@192.168.0.15 Regex (FAIL) [public_did] destination_number(1200) =~ /^(5551212)$/ break=on-false
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:119 (sofia/external/11...@192.168.0.15) State Change CS_ROUTING -> CS_EXECUTE
2014-06-25 14:22:24.717792 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/11...@192.168.0.15 [BREAK]
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:359 (sofia/external/11...@192.168.0.15) State ROUTING going to sleep
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:320 (sofia/external/11...@192.168.0.15) Running State Change CS_EXECUTE
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:366 (sofia/external/11...@192.168.0.15) State EXECUTE
2014-06-25 14:22:24.717792 [DEBUG] mod_sofia.c:240 sofia/external/11...@192.168.0.15 SOFIA EXECUTE
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:157 sofia/external/11...@192.168.0.15 Standard EXECUTE
EXECUTE sofia/external/11...@192.168.0.15 set(outside_call=true)
2014-06-25 14:22:24.717792 [DEBUG] mod_dptools.c:1050 sofia/external/11...@192.168.0.15 SET [outside_call]=[true]
2014-06-25 14:22:24.717792 [NOTICE] switch_core_state_machine.c:189 sofia/external/11...@192.168.0.15 has executed the last dialplan instruction, hanging up.
2014-06-25 14:22:24.717792 [DEBUG] switch_channel.c:2535 (sofia/external/11...@192.168.0.15) Callstate Change RINGING -> HANGUP
2014-06-25 14:22:24.717792 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/external/11...@192.168.0.15 [CS_EXECUTE] [NORMAL_CLEARING]
2014-06-25 14:22:24.717792 [DEBUG] switch_channel.c:2551 Send signal sofia/external/11...@192.168.0.15 [KILL]
2014-06-25 14:22:24.717792 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/11...@192.168.0.15 [BREAK]
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:366 (sofia/external/11...@192.168.0.15) State EXECUTE going to sleep
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:320 (sofia/external/11...@192.168.0.15) Running State Change CS_HANGUP
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:557 (sofia/external/11...@192.168.0.15) State HANGUP
2014-06-25 14:22:24.717792 [DEBUG] mod_sofia.c:457 Channel sofia/external/11...@192.168.0.15 hanging up, cause: NORMAL_CLEARING
2014-06-25 14:22:24.717792 [DEBUG] mod_sofia.c:519 Responding to INVITE with: 480
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:46 sofia/external/11...@192.168.0.15 Standard HANGUP, cause: NORMAL_CLEARING
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:557 (sofia/external/11...@192.168.0.15) State HANGUP going to sleep
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:351 (sofia/external/11...@192.168.0.15) State Change CS_HANGUP -> CS_REPORTING
2014-06-25 14:22:24.717792 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/11...@192.168.0.15 [BREAK]
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:320 (sofia/external/11...@192.168.0.15) Running State Change CS_REPORTING
2014-06-25 14:22:24.717792 [DEBUG] switch_core_state_machine.c:617 (sofia/external/11...@192.168.0.15) State REPORTING
2014-06-25 14:22:24.718794 [DEBUG] switch_core_state_machine.c:53 sofia/external/11...@192.168.0.15 Standard REPORTING, cause: NORMAL_CLEARING
2014-06-25 14:22:24.718794 [DEBUG] switch_core_state_machine.c:617 (sofia/external/11...@192.168.0.15) State REPORTING going to sleep
2014-06-25 14:22:24.718794 [DEBUG] switch_core_state_machine.c:345 (sofia/external/11...@192.168.0.15) State Change CS_REPORTING -> CS_DESTROY
2014-06-25 14:22:24.718794 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/11...@192.168.0.15 [BREAK]
2014-06-25 14:22:24.718794 [DEBUG] switch_core_session.c:1288 Session 4 (sofia/external/11...@192.168.0.15) Locked, Waiting on external entities
2014-06-25 14:22:24.718794 [NOTICE] switch_core_session.c:1306 Session 4 (sofia/external/11...@192.168.0.15) Ended
2014-06-25 14:22:24.718794 [NOTICE] switch_core_session.c:1308 Close Channel sofia/external/11...@192.168.0.15 [CS_DESTROY]
2014-06-25 14:22:24.718794 [DEBUG] switch_core_state_machine.c:449 (sofia/external/11...@192.168.0.15) Callstate Change HANGUP -> DOWN
2014-06-25 14:22:24.718794 [DEBUG] switch_core_state_machine.c:452 (sofia/external/11...@192.168.0.15) Running State Change CS_DESTROY
2014-06-25 14:22:24.718794 [DEBUG] switch_core_state_machine.c:462 (sofia/external/11...@192.168.0.15) State DESTROY
2014-06-25 14:22:24.718794 [DEBUG] mod_sofia.c:362 sofia/external/11...@192.168.0.15 SOFIA DESTROY
2014-06-25 14:22:24.718794 [DEBUG] switch_core_state_machine.c:60 sofia/external/11...@192.168.0.15 Standard DESTROY
2014-06-25 14:22:24.718794 [DEBUG] switch_core_state_machine.c:462 (sofia/external/11...@192.168.0.15) State DESTROY going to sleep

Richard Alam

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Jun 25, 2014, 9:40:34 AM6/25/14
to BigBlueButton-dev
On Wed, Jun 25, 2014 at 5:30 AM, Prashanth Francis <prash...@telenovanetworks.com> wrote:
I created two extensions and registered on two softphones(x-lite) installed on 2 pcs in my network and tried calling,but it wasnt going through(firewalls on both pcs are turned off). This is the debug information from fs_cli

2014-06-25 14:22:24.714795 [NOTICE] switch_channel.c:808 New Channel sofia/external/11...@192.168.0.15 [0708870a-fc46-11e3-b398-d9722d5f6f3e]
2014-06-25 14:22:24.715793 [DEBUG] sofia.c:4624 Channel sofia/external/11...@192.168.0.15 entering state [received][100]


<snip>
 
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->unloop] continue=false
Dialplan: sofia/external/11...@192.168.0.15 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/11...@192.168.0.15 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->outside_call] continue=true
Dialplan: sofia/external/11...@192.168.0.15 Absolute Condition [outside_call]
Dialplan: sofia/external/11...@192.168.0.15 Action set(outside_call=true)
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->call_debug] continue=true
Dialplan: sofia/external/11...@192.168.0.15 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/11...@192.168.0.15 Regex (FAIL) [public_extensions] destination_number(1200) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/11...@192.168.0.15 parsing [public->bbb_conferences] continue=false
Dialplan: sofia/external/11...@192.168.0.15 Regex (FAIL) [bbb_conferences] destination_number(1200) =~ /^(\d{5})$/ break=on-false

DIal a 5-digit number as the conference dialplan expects 5-digits  (destination_number(1200) =~ /^(\d{5})$/)


Richard

 


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Prashanth Francis

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Jun 26, 2014, 5:24:40 AM6/26/14
to bigblueb...@googlegroups.com

After creating two 5-digit extensions, when I try to make a call between them, it shows that the call is established, but the other person does not recieve the call.(Assuming freeswitch functionaliy is similar to that of asterisk). Sorry for my lack of knowledge about bbb and freeswitch, I've still not got a clear idea about how to join a conference(from a sip phone or softphone like zoiper) after creating a conference from the API demos. Please help.

Fred Dixon

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Jun 26, 2014, 6:38:35 AM6/26/14
to BigBlueButton-dev
Hi Prashanth,

Sorry for my lack of knowledge about bbb and freeswitch ...

Have you gone through the FreeSWITCH documentation on setting up for a SIP call?  There is a wealth of documentation at the FreeSWITCH wiki that may help.

Are you able to join to FreeSWITCH directly using a soft phone and understand how the FreeSWITCH configuration works?  See also


Regards,... Fred


On Thu, Jun 26, 2014 at 5:24 AM, Prashanth Francis <prash...@telenovanetworks.com> wrote:

After creating two 5-digit extensions, when I try to make a call between them, it shows that the call is established, but the other person does not recieve the call.(Assuming freeswitch functionaliy is similar to that of asterisk). Sorry for my lack of knowledge about bbb and freeswitch, I've still not got a clear idea about how to join a conference(from a sip phone or softphone like zoiper) after creating a conference from the API demos. Please help.

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Prashanth Francis

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Jun 26, 2014, 8:35:01 AM6/26/14
to bigblueb...@googlegroups.com

Thanks Fred for your reply, so how do you suggest I join a conference from a sip phone? Where do I set the conference id and other parameters required to join a conference on the browser/flash player from a softphone?

Fred Dixon

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Jun 26, 2014, 8:55:59 AM6/26/14
to BigBlueButton-dev
Hi Prashanth,

Thanks Fred for your reply, so how do you suggest I join a conference from a sip phone?

Have you been able to call into FreeSWITCH without going through a conference created in BigBlueButton?  Part of understanding FreeSWITCH is the dialplan configuration.  

DIal a 5-digit number as the conference dialplan expects 5-digits  (destination_number(1200) =~ /^(\d{5})$/)


Regards,... Fred



On Thu, Jun 26, 2014 at 8:35 AM, Prashanth Francis <prash...@telenovanetworks.com> wrote:

Thanks Fred for your reply, so how do you suggest I join a conference from a sip phone? Where do I set the conference id and other parameters required to join a conference on the browser/flash player from a softphone?

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Prashanth Francis

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Jun 27, 2014, 1:03:27 AM6/27/14
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Well as I had mentioned before, I created two 5-digit numbers in freeswitch /opt/freeswitch/conf/directory/default/ named 12000 and 11000, registered on two softphones and tried calling, the call is shown established in the softphone, but the other number doesnt receive the call.

Prashanth Francis

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Jun 27, 2014, 8:24:26 AM6/27/14
to bigblueb...@googlegroups.com

Hi Fred, I just found out that the extensions that I created are getting registered on the sofia external profile rather than the internal profile. Could that be the reason why I'm not able to call between users? When I call a 5-digit extension for a conference I can hear a voice saying "you are the only person in this conference" Please take a look at these outputs from the freeswitch command line.


freeswitch@internal> sofia status profile internal reg

Registrations:
=================================================================================================
Total items returned: 0
=================================================================================================

freeswitch@internal> sofia status profile external reg

Registrations:
=================================================================================================
Call-ID:        852864...@76.74.239.202
User:           bbb...@76.74.239.202
Contact:        "bbbuser" <sip:bbb...@76.74.239.202:5070>
Agent:          mjsip stack 1.6
Status:         Registered(UDP)(unknown) EXP(2011-07-08 23:24:13) EXPSECS(-93723572)
Host:           p2400481.pubip.serverbeach.com
IP:             76.74.239.202
Port:           5070
Auth-User:      bbbuser
Auth-Realm:     76.74.239.202
MWI-Account:    bbb...@76.74.239.202

Call-ID:        ZjAxYzdiMjQ1ZWE4NDczMGQ0ZDUwOGFkZDY5ZTIzMjg.
User:           11...@192.168.0.15
Contact:        "user" <sip:11...@192.168.0.15:49575;rinstance=aeae3e97b1649705;transport=TCP>
Agent:          Z 3.2.21357 r21103
Status:         Registered(TCP)(unknown) EXP(2014-06-27 17:49:16) EXPSECS(331)
Host:           blue-desktop
IP:             192.168.0.15
Port:           59648
Auth-User:      11000
Auth-Realm:     192.168.0.15
MWI-Account:    11...@192.168.0.15

Call-ID:        080099...@192.168.0.15
User:           bbb...@192.168.0.15
Contact:        "bbbuser" <sip:bbb...@192.168.0.15:5070>
Agent:          mjsip stack 1.6
Status:         Registered(UDP)(unknown) EXP(2014-06-27 18:19:15) EXPSECS(2130)
Host:           blue-desktop
IP:             192.168.0.15
Port:           5070
Auth-User:      bbbuser
Auth-Realm:     192.168.0.15
MWI-Account:    bbb...@192.168.0.15

Call-ID:        YzBhMmE0ZTQ2MDMyMzgwMDM0Y2M3NDk3ZDBkNTNhODg
User:           10...@192.168.0.15
Contact:        "1002" <sip:10...@192.168.0.192:40542;rinstance=f65ab5f4a0a6a2b6>
Agent:          X-Lite release 4.5.5  stamp 71236
Status:         Registered(UDP)(unknown) EXP(2014-06-27 18:31:46) EXPSECS(2881)
Host:           blue-desktop
IP:             192.168.0.192
Port:           40542
Auth-User:      1002
Auth-Realm:     192.168.0.15
MWI-Account:    10...@192.168.0.15

Call-ID:        M2Y2M2Q2ZTQ4YWFiYzBhN2QzYzdjN2E0ZTlmOTNjNjI
User:           10...@192.168.0.15
Contact:        "1001" <sip:10...@192.168.0.19:27504;rinstance=dec7e9c86628f77b>
Agent:          X-Lite release 4.5.5  stamp 71236
Status:         Registered(UDP)(unknown) EXP(2014-06-27 18:02:34) EXPSECS(1129)
Host:           blue-desktop
IP:             192.168.0.19
Port:           27504
Auth-User:      1001
Auth-Realm:     192.168.0.15
MWI-Account:    10...@192.168.0.15

Call-ID:        MDliYmMxYzVkYmM4NjUzZGJjOGViMmE4ZWZmYzlhZjc.
User:           10...@192.168.0.15
Contact:        "user" <sip:10...@192.168.0.15:49575;rinstance=4073791aa1a8f17b;transport=TCP>
Agent:          Z 3.2.21357 r21103
Status:         Registered(TCP)(unknown) EXP(2014-06-27 18:44:14) EXPSECS(3629)
Host:           blue-desktop
IP:             192.168.0.15
Port:           59648
Auth-User:      1000
Auth-Realm:     192.168.0.15
MWI-Account:    10...@192.168.0.15

Total items returned: 6
=================================================================================================

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