[Freeswitch-users] custom sip header not working in dialplan

windy xiaofengcanyuexp at 163.com
Tue May 21 13:41:43 MSD 2013


 
Hi, Freeswitch Team

There is problem for to modify the outbound sip header.

As the freeswitch wiki says, it can modify/add custom sip header via below method:

<action application="set" data="sip_h_X-Charge=1"/>
<action application="set" data="sip_rh_X-Reason=Destination Number Not in Footprint"/>
<action application="set" data="sip_bye_h_X-Accounting=Some Accounting Data"/>


Now, I add the red lines in dialplan/default.xml. But only the custom header prefixed with "sip_h_" can be shown in the SIP INVITE message.
Other custom headers prefixed with "sip_bye_h" and "sip_ph"... not found in the related messages(BYE, 183).

Could you kindly give comments about this case?

<context name="from-pstn-to-sip">
        <extension name="to-sip">
                <!-- handle the case where there might not be destination number at all -->
                <condition field="destination_number" expression="^(.{1,})$" break="never">
                        <action application="set" data="destnumber=$1"/>
                        <anti-action application="set" data="destnumber=unknown"/>
                </condition>
                
                <!-- Dial to the gateway user (it may ring multiple registrations, first answer wins) -->
                <condition field="destination_number" expression="^(.*)$">
                        <action application="set" data="hangup_after_bridge=yes" />
                        <action application="set" data="tone_detect_hits=1" />
                        <action application="export" data="fax_enable_t38_request=true" />
                        <action application="export" data="fax_enable_t38=true" />
                        <action application="tone_detect" data="faxdisable 1100 r +5000 disable_ec 1"/>
                        <action application="export" data="execute_on_answer=tone_detect fax_disable_ec 2100 r +5000 t38_gateway 'self nocng'" />
                        <action application="set" data="sip_contact_user_replacement=${destnumber}"/>
                        <action application="set" data="hangup_after_bridge=yes"/>

                        <!-- Uncomment this line to switch to SIP INFO when remote SDP does not support RFC-2833 -->
                        <action application="export" data="sip_info_when_no_2833=false" />

                        <action application="set" data="sip_h_X-Charge=1"/>
                        <action application="set" data="sip_rh_X-Reason=Destination Number Not in Footprint"/>
                        <action application="set" data="sip_bye_h_X-Accounting=Some Accounting Data"/>

                        <!-- Uncomment the line below to call directly to a SIP endpoint --> 
                        <!-- <action application="bridge" data="sofia/internal/${destnumber}@${sip_remote_ip}:${sip_remote_port}"/> -->

                        <!-- <action application="bridge" data="${sofia_contact($${gwuser}@$${domain})}"/> -->
                        <action application="bridge" data="sofia/external/sip:${destination_number}@172.20.2.17:5062"/>
                        <action application="hangup" data="${originate_disposition}"/>
                </condition>
        </extension>

Thanks
Windy
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