Toolpack monitoring sip simulator variables

The Toolpack_monitoring_sip_simulator is formatting the generated SIP messages based on a text-based configuration, which describes the SIP message content.

Well-known variables are used to define portion of the SIP messages that change from one call to the other.

Example SIP message format

 INVITE sip:#{called}@#{callee_sip_ip}:#{callee_sip_port} SIP/2.0
 Call-ID:#{call_id}
 X-Monitored-Sender:#{sender_label}:#{sender}
 X-Isdn-Info:interface-id:#{isdn_interface_id};channel-number:#{isdn_channel_number}
 X-Example-Var:#{example_var}
 To:<sip:#{sip_to}>
 From:#{calling_display}<sip:#{sip_from}>;tag=#{caller_tag}
 Remote-Party-ID:#{calling_private_display}<sip:#{calling_private}@#{caller_sip_ip}:#{caller_sip_port}>;party=calling;screen=#{calling_screening};privacy=#{calling_presentation};noa=#{calling_noa}
 Privacy:#{calling_privacy}
 Contact:sip:#{calling}@#{caller_sip_ip}:#{caller_sip_port}
 Via:SIP/2.0/UDP #{caller_sip_ip}:#{caller_sip_port};branch=#{branch};rport
 User-to-User:#{uui};encoding=text;purpose=isdn-interwork;content=isdn-uui
 Date:#{date}
 Timestamp:#{timestamp}
 CSeq:#{cseq}
 Max-Forwards:70
 #{caller_content}

To see other SIP messages default formatting, check in the Toolpack_monitoring_sip_simulator configuration page in the Web Portal.

Variables

All variables are referenced in the SIP messages definition by wrapping them inside #{}

For example: #{call_id}

There are "well-known" variables, which value is dynamically replaced by the SIP simulator according to the context.

There are "user-defined" variables that can be inserted in the SIP messages.

Well-known variables

• sip_to  : The simulated SIP "from" address: called@ip.port
• called  : The called number, as decoded from the monitored call
• called_noa  : The called number NPI (Numbering Plan), as decoded from the monitored call
• called_npi  : The called number NPI (Numbering Plan), as decoded from the monitored call
• sip_from  : The simulated SIP "to" address: calling@ip.port, or anonymous@anonymous.invalid
• calling  : The calling number (or "anonymous" if privacy is enabled), as decoded from the monitored call
• calling_noa  : The calling number NOA (Nature Of Address), as decoded from the monitored call
• calling_npi  : The calling number NPI (Numbering Plan), as decoded from the monitored call
• calling_display  : The calling number display string, as decoded from the monitored call
• calling_private  : The private calling number, as decoded from the monitored call
• calling_private_display  : The private calling number display string (or simply calling number if no privacy is used), as decoded from the monitored call
• calling_privacy  : The "Privacy" header value ("id" or "none"), as decoded from the monitored call's "calling presentation"
• calling_presentation  : The calling number presentation, as decoded from the monitored call, converted SIP as "full", "uri", "name", or "off")
• calling_screening  : The calling number screening, as decoded from the monitored call, converted for SIP as "yes" or "no"
• calling_category  : The calling number category, as decoded from the monitored call, converted for SIP as "operator", "ordinary", "priority", "payphone"...
• call_id  : A generated SIP Call id, unique per call
• uui  : The UUI (User-to-User Information), as decoded from the monitored call, convert for SIP in the escaped-text format
• timestamp  : A precise timestamp (Synchronized on TDM, in milliseconds since Epoch) indicating when the monitored message that corresponds to this SIP packet was received
• date  : Date/time, in a standard SIP format, that corresponds to the timestamp indicating when the monitored message that corresponds to this SIP packet was received. Example: "Thu, 21 Apr 2010 14:12:51 GMT"
• sender_label  : The "label" of sender of the monitored message that corresponds to this SIP packet (the "label" is provided in the Analyzer's configuration)
• sender  : The sender of the monitored message that corresponds to this SIP packet (trunk:timeslot from which the message was received)
• isdn_interface_id  : The ISDN interface id of the corresponding monitored ISDN call (if actually monitoring an ISDN call)
• isdn_channel_number  : The ISDN channel number of the corresponding monitored ISDN call (if actually monitoring an ISDN call)
• cseq  : A generated SIP sequence number, incremented appropriately
• branch  : A generated "branch" parameter for the "Via" header
• q850_cause  : The Q.850 termination cause of the call, as decoded from the monitored call
• sip_cause  : The SIP termination cause of the call (derived from Q850 termination cause when appropriate)
• caller_sip_ip  : The simulated SIP IP address assigned to Caller (either IP assigned to first or second side by configuration, depending on the call direction)
• caller_sip_port  : The simulated SIP port assigned to Caller side (either IP assigned to first or second side by configuration, depending on the call direction)
• caller_media_ip  : The local RTP IP address used by the caller side (as defined by the "IP Interface" that owns the "port range" that is assigned to the "VOIP-media" NAP that is assigned to the SIP Simulator's corresponding side)
• caller_media_port  : The RTP port assigned to this port (as chosen among avaialble ports in the "port range" that is assigned to the "VOIP-media" NAP that is assigned to the SIP Simulator's corresponding side)
• caller_tag  : A generated SIP "tag" for the Caller side
• caller_content  : The generated caller side "Content-Length", "Content-Type" headers, and the SIP body: SDP and/or Raw Q.921 payload when appropriate
• callee_sip_ip  : The simulated SIP IP address assigned to Callee (either IP assigned to first or second side by configuration, depending on the call direction)
• callee_sip_port  : The simulated SIP port assigned to Callee side (either IP assigned to first or second side by configuration, depending on the call direction)
• callee_media_ip  : The local RTP IP address used by the callee side (as defined by the "IP Interface" that owns the "port range" that is assigned to the "VOIP-media" NAP that is assigned to the SIP Simulator's corresponding side)
• callee_media_port  : The RTP port assigned to this port (as chosen among avaialble ports in the "port range" that is assigned to the "VOIP-media" NAP that is assigned to the SIP Simulator's corresponding side)
• callee_tag  : A generated SIP "tag" for the Callee side
• callee_content  : The generated callee side "Content-Length", "Content-Type" headers, and the SIP body: SDP and/or Raw Q.921 payload when appropriate
• multipart_boundary  : A generated SIP multipart body boundary that is used when the SIP body contains multiple parts (when both SDP and Raw Q.921 payload are present)