Sipura Syslogging SIP INVITE incoming

Incoming Completed Call

Here’s an example of an incoming call from 1-866-510-6559 to the SPA.

Assumptions

SIP Port: 5062 (SPA SIP Port)
SIP Proxy Server: sip.voip-info.org
SPA IP address: 192.168.0.2
User ID: spa1
Display Name: Good Body

Message Flow Explanation

An incoming call INVITE conveyed by the SIP Proxy reaches the SPA
SPA response back to the SIP Proxy that it’s trying to complete the call
SPA informs the SIP Proxy it is ringing the phone
SPA informs the SIP Proxy the phone has been picked up
SIP Proxy acknowledges receiving the message that the phone has been picked up
SPA issues a BYE to the SIP Proxy informing the Proxy it wishes to end the inbound call
SIP Proxy confirms the SPA’s request to end the inbound call.

Syslog Entries

Jun 20 17:15:08 192.168.0.2 INVITE sip:[email protected]:5062 SIP/2.0^M Via: SIP/2.0/UDP sip.voip-info.org:5060;branch=z9hG4bK140ff4fb^M From: “18665106559” <sip:[email protected]>;tag=as06b3b8cd^M To: <sip:[email protected]:5062>^M Contact: <sip:[email protected]>^M Call-ID: [email protected]^M CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Date: Sat,20 Jun 2005 09:15:08 GMT^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER^M Content-Type: application/sdp^M Content-Length: 268^M ^M v=0^M o=root 30292 30292 IN IP4 sip.voip-info.org^M s=session^M c=IN IP4 sip.voip-info.org^M t=0 0^M m=audio 16422 RTP/AVP 8 0 4 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:4 G723/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off – – – -^M

Jun 20 17:15:08 192.168.0.2 SIP/2.0 100 Trying^M To: <sip:[email protected]:5062>^M From: “18665106559” <sip:[email protected]>;tag=as06b3b8cd^M Call-ID: [email protected]^M CSeq: 102 INVITE^M Via: SIP/2.0/UDP sip.voip-info.org:5060;branch=z9hG4bK140ff4fb^M Server: Sipura/SPA3000-2.0.13(GWg)^M Content-Length: 0^M ^M

Jun 20 17:15:08 192.168.0.2 SIP/2.0 180 Ringing^M To: <sip:[email protected]:5062>;tag=9e3c2820efacd2d9i0^M From: “18665106559” <sip:[email protected]>;tag=as06b3b8cd^M Call-ID: [email protected]^M CSeq: 102 INVITE^M Via: SIP/2.0/UDP sip.voip-info.org:5060;branch=z9hG4bK140ff4fb^M Server: Sipura/SPA3000-2.0.13(GWg)^M Content-Length: 0^M ^M

Jun 20 17:15:16 192.168.0.2 SIP/2.0 200 OK^M To: <sip:[email protected]:5062>;tag=9e3c2820efacd2d9i0^M From: “18665106559” <sip:[email protected]>;tag=as06b3b8cd^M Call-ID: [email protected]^M CSeq: 102 INVITE^M Via: SIP/2.0/UDP sip.voip-info.org:5060;branch=z9hG4bK140ff4fb^M Contact: Good Body <sip:[email protected]:5062>^M Server: Sipura/SPA3000-2.0.13(GWg)^M Content-Length: 239^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M Content-Type: application/sdp^M ^M v=0^M o=- 174586 174586 IN IP4 192.168.0.2^M s=-^M c=IN IP4 192.168.0.2^M t=0 0^M m=audio 16390 RTP/AVP 8 100 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:100 NSE/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=ptime:30^M a=sendrecv^M

Jun 20 17:15:16 192.168.0.2 ACK sip:[email protected]:5062 SIP/2.0^M Via: SIP/2.0/UDP sip.voip-info.org:5060;branch=z9hG4bK3c6738ce^M From: “18665106559” <sip:[email protected]>;tag=as06b3b8cd^M To: <sip:[email protected]:5062>;tag=9e3c2820efacd2d9i0^M Contact: <sip:[email protected]>^M Call-ID: [email protected]^M CSeq: 102 ACK^M User-Agent: Asterisk PBX^M Content-Length: 0^M ^M

Jun 20 17:17:04 192.168.0.2 BYE sip:[email protected]:5062 SIP/2.0^M Via: SIP/2.0/UDP sip.voip-info.org:5060;branch=z9hG4bK6f8ecf47^M From: “18665106559” <sip:[email protected]>;tag=as06b3b8cd^M To: <sip:[email protected]:5062>;tag=9e3c2820efacd2d9i0^M Contact: <sip:[email protected]>^M Call-ID: [email protected]^M CSeq: 103 BYE^M User-Agent: Asterisk PBX^M Content-Length: 0^M ^M

Jun 20 17:17:04 192.168.0.2 SIP/2.0 200 OK^M To: <sip:[email protected]:5062>;tag=9e3c2820efacd2d9i0^M From: “18665106559” <sip:[email protected]>;tag=as06b3b8cd^M Call-ID: [email protected]^M CSeq: 103 BYE^M Via: SIP/2.0/UDP sip.voip-info.org:5060;branch=z9hG4bK6f8ecf47^M Server: Sipura/SPA3000-2.0.13(GWg)^M Content-Length: 0^M ^M

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