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voip technocrat

2004-Sep-30 23:04 UTC

[Asterisk-Users] problem in using sip-communicator with asterisk

hello firends,

iam using sip-communicator with asterisk

its registering and ringing but 

 i could able to listen the voice with only one side

i.e the destination could able to hear what  source is


saying and source number could not here any thing 

the dailer iam running in the public ips 

even when i test with the fxs the source i.e dialer 

couldnot able to listen any thing when session starts 

fxs endpoint could able to listen what dialer party is


saying 

iam attaching the sip-communicator.xml here so if 

possible please tell me where am i going wrong?

<?xml version="1.0" encoding="UTF-8"?>
<configuration>
<log4j>
    <rootLogger
value="net.java.sip.communicator.common.Console.TraceLevel,
RFLogger"/>
    <appender>
        <RFLogger
value="org.apache.log4j.RollingFileAppender">
            <layout
value="org.apache.log4j.PatternLayout">
                <ConversionPattern value="%r [%t] %p
%c{2} %x - %m%n"/>
            </layout>
            <MaxBackupIndex value="1"/>
            <File
value="log/sip-communicator.app.log"/>
            <MaxFileSize value="256KB"/>
        </RFLogger>
    </appender>
</log4j>
<net>
  <java>
    <sip>
      <communicator>
          <FIRST_LAUNCH value="true"/>
          <ENABLE_SIMPLE value="false"/>
          <media>
<!---         <PREFERRED_AUDIO_ENCODING system="false"
value=""/> -->
              <PREFERRED_AUDIO_ENCODING value="0"/>
              <PREFERRED_VIDEO_ENCODING value="26"/>
            <MEDIA_SOURCE value=""/>
            <MEDIA_BUFFER_LENGTH value="100"/>
            <IP_ADDRESS value=""/>
            <AUDIO_PORT value=""/>
            <VIDEO_PORT value=""/>
        </media>
        <sip>
            <PUBLIC_ADDRESS value=""/>
            <TRANSPORT value=""/>
            <REGISTRAR_ADDRESS value="*.*.*.19"/>
            <USER_NAME value=""/>
            <STACK_PATH value="gov.nist"/>
            <PREFERRED_LOCAL_PORT value=""/>
            <DISPLAY_NAME value=""/>
            <REGISTRAR_TRANSPORT value="UDP"/>
            <REGISTRATIONS_EXPIRATION value="3600"/>
            <REGISTRAR_PORT value="5060"/>
            <FAIL_CALLS_ON_DEST_USER_MISMATCH
value="false"/>

            <DEFAULT_DOMAIN_NAME value="*.*.*.19"/>
            <DEFAULT_AUTHENTICATION_REALM
value="*.*.*.19"/>
            <WAIT_UNREGISTGRATION_FOR value="1100"/>
            <SAME_USER_EVERYWHERE value="true"/>
            <simple>
                <CONTACT_LIST_FILE
value="contact-list.xml"/>
                <SUBSCRIPTION_EXP_TIME value="600"/>
                <MIN_EXP_TIME value="120"/>
                <LAST_SELECTED_OPEN_STATUS
value="online"/>
            </simple>
        </sip>

<!--
   
net.java.sip.communicator.sipphone.IS_RUNNING_SIPPHONE=false
   
net.java.sip.communicator.sipphone.MY_SIPPHONE_URL=http://my.sipphone.com
-->
        <sipphone>
            <IS_RUNNING_SIPPHONE value="false"/>
            <MY_SIPPHONE_URL
value="http://my.sipphone.com"/>
        </sipphone>
<!--
net.java.sip.communicator.gui.AUTH_WIN_TITLE=SIP
Authentication!
net.java.sip.communicator.gui.AUTHENTICATION_PROMPT=Please
enter login name and password for the specified realm:
net.java.sip.communicator.gui.USER_NAME_LABEL=SIPphone
Number:
net.java.sip.communicator.sipphone.USER_NAME_EXAMPLE=Example:
1-747-555-1212
net.java.sip.communicator.gui.PASSWORD_LABEL=Password:
-->
        <gui>
            <AUTH_WIN_TITLE value="SIP
Authentication!"/>
            <AUTHENTICATION_PROMPT value="Please enter
login name and password for the specified realm:"/>
            <USER_NAME_LABEL value="User Name:"/>
            <USER_NAME_EXAMPLE value="Example:
1-747-555-1212"/>
            <PASSWORD_LABEL value="Password:"/>
            <GUI_MODE value="PhoneUiMode"/>
            <!--GUI_MODE value="ImUiMode"/-->
            <imp>
                <CONTACT_LIST_X value=""/>
                <CONTACT_LIST_Y value=""/>
                <CONTACT_LIST_WIDTH value=""/>
                <CONTACT_LIST_HEIGHT value=""/>
            </imp>
        </gui>
        <common>
            <PREFERRED_NETWORK_INTERFACE value=""/>
            <PREFERRED_NETWORK_ADDRESS value=""/>
        </common>


<!--
   
net.java.sip.communicator.STUN_SERVER_ADDRESS=stun01.sipphone.com
    net.java.sip.communicator.STUN_SERVER_PORT=3478
   
net.java.sip.communicator.VOICE_MAIL_ADDRESS=17475551212
-->
        <STUN_SERVER_ADDRESS
value="stun01.sipphone.com"/>
        <STUN_SERVER_PORT value="3478"/>
        <VOICE_MAIL_ADDRESS value="17475551212"/>
</communicator>
    </sip>
  </java>
</net>
    <gov>
    <nist>
        <javax>
            <sip>
                <SERVER_LOG
value="log/sip-communicator.stack.log"/>
                <TRACE_LEVEL value="16"/>
            </sip>
        </javax>
    </nist>
</gov>
<javax>
    <sip>
        <IP_ADDRESS value=""/>
        <STACK_NAME value="sip-communicator"/>
        <ROUTER_PATH
value="net.java.sip.communicator.sip.SipCommRouter"/>
        <OUTBOUND_PROXY value="*.*.*.19:5060/udp"/>
        <RETRANSMISSON_FILTER value=""/>
        <EXTENSION_METHODS value=""/>
        <RETRANSMISSION_FILTER value="true"/>
    </sip>
</javax>
<java>
    <net>
        <preferIPv4Stack system="true" value="true"/>
        <preferIPv6Addresses system="true"
value="false"/>
    </net>
</java>
</configuration>

with regards
serdiehard

________________________________________________________________________
Yahoo! India Matrimony: Find your life partner online
Go to: http://yahoo.shaadi.com/india-matrimony

Miroslav Nachev

2004-Oct-01 01:13 UTC

head link

[Asterisk-Users] problem in using sip-communicator with asterisk

Hi,

   Try to set "canreinvite=yes".
   

   Best Regards,
   Miroslav Nachev
   
hello firends,

iam using sip-communicator with asterisk

its registering and ringing but 

 i could able to listen the voice with only one side

i.e the destination could able to hear what  source is


saying and source number could not here any thing 

the dailer iam running in the public ips 

even when i test with the fxs the source i.e dialer 

couldnot able to listen any thing when session starts 

fxs endpoint could able to listen what dialer party is


saying 

iam attaching the sip-communicator.xml here so if 

possible please tell me where am i going wrong?

<?xml version="1.0" encoding="UTF-8"?>
<configuration>
<log4j>
    <rootLogger
value="net.java.sip.communicator.common.Console.TraceLevel,
RFLogger"/>
    <appender>
        <RFLogger
value="org.apache.log4j.RollingFileAppender">
            <layout
value="org.apache.log4j.PatternLayout">
                <ConversionPattern value="%r [%t] %p
%c{2} %x - %m%n"/>
            </layout>
            <MaxBackupIndex value="1"/>
            <File
value="log/sip-communicator.app.log"/>
            <MaxFileSize value="256KB"/>
        </RFLogger>
    </appender>
</log4j>
<net>
  <java>
    <sip>
      <communicator>
          <FIRST_LAUNCH value="true"/>
          <ENABLE_SIMPLE value="false"/>
          <media>
<!---         <PREFERRED_AUDIO_ENCODING system="false"
value=""/> -->
              <PREFERRED_AUDIO_ENCODING value="0"/>
              <PREFERRED_VIDEO_ENCODING value="26"/>
            <MEDIA_SOURCE value=""/>
            <MEDIA_BUFFER_LENGTH value="100"/>
            <IP_ADDRESS value=""/>
            <AUDIO_PORT value=""/>
            <VIDEO_PORT value=""/>
        </media>
        <sip>
            <PUBLIC_ADDRESS value=""/>
            <TRANSPORT value=""/>
            <REGISTRAR_ADDRESS value="*.*.*.19"/>
            <USER_NAME value=""/>
            <STACK_PATH value="gov.nist"/>
            <PREFERRED_LOCAL_PORT value=""/>
            <DISPLAY_NAME value=""/>
            <REGISTRAR_TRANSPORT value="UDP"/>
            <REGISTRATIONS_EXPIRATION value="3600"/>
            <REGISTRAR_PORT value="5060"/>
            <FAIL_CALLS_ON_DEST_USER_MISMATCH
value="false"/>

            <DEFAULT_DOMAIN_NAME value="*.*.*.19"/>
            <DEFAULT_AUTHENTICATION_REALM
value="*.*.*.19"/>
            <WAIT_UNREGISTGRATION_FOR value="1100"/>
            <SAME_USER_EVERYWHERE value="true"/>
            <simple>
                <CONTACT_LIST_FILE
value="contact-list.xml"/>
                <SUBSCRIPTION_EXP_TIME value="600"/>
                <MIN_EXP_TIME value="120"/>
                <LAST_SELECTED_OPEN_STATUS
value="online"/>
            </simple>
        </sip>

<!--
   
net.java.sip.communicator.sipphone.IS_RUNNING_SIPPHONE=false
   
net.java.sip.communicator.sipphone.MY_SIPPHONE_URL=http://my.sipphone.com
-->
        <sipphone>
            <IS_RUNNING_SIPPHONE value="false"/>
            <MY_SIPPHONE_URL
value="http://my.sipphone.com"/>
        </sipphone>
<!--
net.java.sip.communicator.gui.AUTH_WIN_TITLE=SIP
Authentication!
net.java.sip.communicator.gui.AUTHENTICATION_PROMPT=Please
enter login name and password for the specified realm:
net.java.sip.communicator.gui.USER_NAME_LABEL=SIPphone
Number:
net.java.sip.communicator.sipphone.USER_NAME_EXAMPLE=Example:
1-747-555-1212
net.java.sip.communicator.gui.PASSWORD_LABEL=Password:
-->
        <gui>
            <AUTH_WIN_TITLE value="SIP
Authentication!"/>
            <AUTHENTICATION_PROMPT value="Please enter
login name and password for the specified realm:"/>
            <USER_NAME_LABEL value="User Name:"/>
            <USER_NAME_EXAMPLE value="Example:
1-747-555-1212"/>
            <PASSWORD_LABEL value="Password:"/>
            <GUI_MODE value="PhoneUiMode"/>
            <!--GUI_MODE value="ImUiMode"/-->
            <imp>
                <CONTACT_LIST_X value=""/>
                <CONTACT_LIST_Y value=""/>
                <CONTACT_LIST_WIDTH value=""/>
                <CONTACT_LIST_HEIGHT value=""/>
            </imp>
        </gui>
        <common>
            <PREFERRED_NETWORK_INTERFACE value=""/>
            <PREFERRED_NETWORK_ADDRESS value=""/>
        </common>


<!--
   
net.java.sip.communicator.STUN_SERVER_ADDRESS=stun01.sipphone.com
    net.java.sip.communicator.STUN_SERVER_PORT=3478
   
net.java.sip.communicator.VOICE_MAIL_ADDRESS=17475551212
-->
        <STUN_SERVER_ADDRESS
value="stun01.sipphone.com"/>
        <STUN_SERVER_PORT value="3478"/>
        <VOICE_MAIL_ADDRESS value="17475551212"/>
</communicator>
    </sip>
  </java>
</net>
    <gov>
    <nist>
        <javax>
            <sip>
                <SERVER_LOG
value="log/sip-communicator.stack.log"/>
                <TRACE_LEVEL value="16"/>
            </sip>
        </javax>
    </nist>
</gov>
<javax>
    <sip>
        <IP_ADDRESS value=""/>
        <STACK_NAME value="sip-communicator"/>
        <ROUTER_PATH
value="net.java.sip.communicator.sip.SipCommRouter"/>
        <OUTBOUND_PROXY value="*.*.*.19:5060/udp"/>
        <RETRANSMISSON_FILTER value=""/>
        <EXTENSION_METHODS value=""/>
        <RETRANSMISSION_FILTER value="true"/>
    </sip>
</javax>
<java>
    <net>
        <preferIPv4Stack system="true" value="true"/>
        <preferIPv6Addresses system="true"
value="false"/>
    </net>
</java>
</configuration>

with regards
serdiehard

________________________________________________________________________
Yahoo! India Matrimony: Find your life partner online
Go to: http://yahoo.shaadi.com/india-matrimony
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asterisk users - Sep 2004 - problem in using sip-communicator with asterisk

[Asterisk-Users] problem in using sip-communicator with asterisk

[Asterisk-Users] problem in using sip-communicator with asterisk