SIP to H323

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== Asterisk ==

<-- SIP read from 129.125.71.173:5060: INVITE sip:601@129.125.21.241 SIP/2.0 Via: SIP/2.0/UDP 129.125.71.173:5060;rport;branch=z9hG4bK28425BFFA543520B0B9D0FE0CA28AD10 From: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 To: <sip:601@129.125.21.241> Contact: <sip:502@129.125.71.173:5060> Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 24313 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 312

v=0 o=502 1178656389 1178656399 IN IP4 129.125.71.173 s=X-Lite c=IN IP4 129.125.71.173 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv

--- (11 headers 14 lines)--- Using INVITE request as basis request - 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 Sending to 129.125.71.173 : 5060 (NAT) Reliably Transmitting (no NAT) to 129.125.71.173:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 129.125.71.173:5060;rport;branch=z9hG4bK28425BFFA543520B0B9D0FE0CA28AD10;received=129.125.71.173 From: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 To: <sip:601@129.125.21.241>;tag=as3321ad7e Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 24313 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:601@129.125.21.241> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="18d198f5" Content-Length: 0


--- Scheduling destruction of call '39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173' in 15000 ms Found user '502'

<-- SIP read from 129.125.71.173:5060: ACK sip:601@129.125.21.241 SIP/2.0 Via: SIP/2.0/UDP 129.125.71.173:5060;rport;branch=z9hG4bK28425BFFA543520B0B9D0FE0CA28AD10 From: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 To: <sip:601@129.125.21.241>;tag=as3321ad7e Contact: <sip:502@129.125.71.173:5060> Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 24313 ACK Max-Forwards: 70 Content-Length: 0


--- (9 headers 0 lines)---

<-- SIP read from 129.125.71.173:5060: INVITE sip:601@129.125.21.241 SIP/2.0 Via: SIP/2.0/UDP 129.125.71.173:5060;rport;branch=z9hG4bK5ECEFA77EC7446BB73C0364FD5D9D9FB From: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 To: <sip:601@129.125.21.241> Contact: <sip:502@129.125.71.173:5060> Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 24314 INVITE Proxy-Authorization: Digest username="502",realm="asterisk",nonce="18d198f5",response="af4294cd96f7dba63e0fbf3ea9c0e161",uri="sip:601@129.125.21.241",algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 312

v=0 o=502 1178656389 1178656399 IN IP4 129.125.71.173 s=X-Lite c=IN IP4 129.125.71.173 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv

--- (12 headers 14 lines)--- Using INVITE request as basis request - 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 Sending to 129.125.71.173 : 5060 (NAT) Found user '502' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 129.125.71.173:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 601 in from-internal (domain 129.125.21.241) list_route: hop: <sip:502@129.125.71.173:5060> Transmitting (no NAT) to 129.125.71.173:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 129.125.71.173:5060;rport;branch=z9hG4bK5ECEFA77EC7446BB73C0364FD5D9D9FB;received=129.125.71.173 From: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 To: <sip:601@129.125.21.241> Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 24314 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:601@129.125.21.241> Content-Length: 0


--- 

   -- Executing Macro("SIP/502-083b1650", "exten-vm|novm|601") in new stack
   -- Executing Macro("SIP/502-083b1650", "user-callerid") in new stack
   -- Executing GotoIf("SIP/502-083b1650", "0?report") in new stack
   -- Executing GotoIf("SIP/502-083b1650", "0?start") in new stack
   -- Executing Set("SIP/502-083b1650", "REALCALLERIDNUM=502") in new stack
   -- Executing NoOp("SIP/502-083b1650", "REALCALLERIDNUM is 502") in new stack
   -- Executing Set("SIP/502-083b1650", "AMPUSER=502") in new stack
   -- Executing Set("SIP/502-083b1650", "AMPUSERCIDNAME=Fabio SIP freePBX") in new stack
   -- Executing GotoIf("SIP/502-083b1650", "0?report") in new stack
   -- Executing Set("SIP/502-083b1650", "CALLERID(all)=Fabio SIP freePBX <502>") in new stack
   -- Executing NoOp("SIP/502-083b1650", "Using CallerID "Fabio SIP freePBX" <502>") in new stack
   -- Executing Set("SIP/502-083b1650", "FROMCONTEXT=exten-vm") in new stack
   -- Executing Set("SIP/502-083b1650", "VMBOX=novm") in new stack
   -- Executing Set("SIP/502-083b1650", "EXTTOCALL=601") in new stack
   -- Executing Set("SIP/502-083b1650", "CFUEXT=") in new stack
   -- Executing Set("SIP/502-083b1650", "RT=") in new stack
   -- Executing Macro("SIP/502-083b1650", "record-enable|601|IN") in new stack
   -- Executing GotoIf("SIP/502-083b1650", "0 > 0?2:4") in new stack
   -- Goto (macro-record-enable,s,4)
   -- Executing AGI("SIP/502-083b1650", "recordingcheck|20070312-140616|1173704776.2") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0 

 recordingcheck|20070312-140616|1173704776.2: Inbound recording not enabled
   -- AGI Script recordingcheck completed, returning 0
   -- Executing NoOp("SIP/502-083b1650", "No recording needed") in new stack
   -- Executing GotoIf("SIP/502-083b1650", "0?dolocaldial|1") in new stack
   -- Executing Macro("SIP/502-083b1650", "dial||tr|601") in new stack
   -- Executing AGI("SIP/502-083b1650", "dialparties.agi") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi

PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0 

 dialparties.agi: Starting New Dialparties.agi
   --  dialparties.agi: priority is 1
 dialparties.agi: Caller ID name is 'Fabio SIP freePBX' number is '502'
 dialparties.agi: Methodology of ring is  'none'
   --  dialparties.agi: Added extension 601 to extension map
   --  dialparties.agi: Extension 601 cf is disabled
   --  dialparties.agi: Extension 601 do not disturb is disabled
      >  dialparties.agi: extnum: 601
      >  dialparties.agi: exthascw: 0
      >  dialparties.agi: exthascfb: 0
      >  dialparties.agi: extcfb:
      >  dialparties.agi: exthascfu: 0
      >  dialparties.agi: extcfu:
 == Parsing '/etc/asterisk/manager.conf': Found
 == Parsing '/etc/asterisk/manager_custom.conf': Found
 == Manager 'admin' logged on from 127.0.0.1
 == Manager 'admin' logged off from 127.0.0.1
      >  dialparties.agi: ExtensionState: 0
   --  dialparties.agi: Checking CW and CFB status for extension 601
   --  dialparties.agi: DbSet CALLTRACE/601 to 502
   -- AGI Script dialparties.agi completed, returning 0
   -- Executing Dial("SIP/502-083b1650", "OOH323/601||tr") in new stack

--- ooh323_request - data 601 format 0x4 (ulaw) --- find_peer +++ find_peer +++ ooh323_request --- ooh323_call- 601 +++ ooh323_call 

   -- Called 601

Transmitting (no NAT) to 129.125.71.173:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 129.125.71.173:5060;rport;branch=z9hG4bK5ECEFA77EC7446BB73C0364FD5D9D9FB;received=129.125.71.173 From: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 To: <sip:601@129.125.21.241>;tag=as4618ff70 Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 24314 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:601@129.125.21.241> Content-Length: 0


--- --- onNewCallCreated ooh323c_o_2 --- find_call +++ find_call setting callid number 502 

Outgoing call 601(ooh323c_o_2) - Codec prefs - (gsm|ulaw)
       Adding capabilities to call(outgoing, ooh323c_o_2)
       Adding gsm capability to call(outgoing, ooh323c_o_2)
       Adding g711 ulaw capability to call(outgoing, ooh323c_o_2)

--- configure_local_rtp +++ configure_local_rtp +++ onNewCallCreated ooh323c_o_2 --- onAlerting ooh323c_o_2 --- find_call +++ find_call +++ onAlerting ooh323c_o_2 

   -- OOH323/601-489a is ringing

--- onCallEstablished ooh323c_o_2 --- find_call +++ find_call +++ onCallEstablished ooh323c_o_2 

   -- OOH323/601-489a answered SIP/502-083b1650

We're at 129.125.21.241 port 15850 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 129.125.71.173:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 129.125.71.173:5060;rport;branch=z9hG4bK5ECEFA77EC7446BB73C0364FD5D9D9FB;received=129.125.71.173 From: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 To: <sip:601@129.125.21.241>;tag=as4618ff70 Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 24314 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:601@129.125.21.241> Content-Type: application/sdp Content-Length: 242

v=0 o=root 9003 9003 IN IP4 129.125.21.241 s=session c=IN IP4 129.125.21.241 t=0 0 m=audio 15850 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -

---

<-- SIP read from 129.125.71.173:5060: ACK sip:601@129.125.21.241 SIP/2.0 Via: SIP/2.0/UDP 129.125.71.173:5060;rport;branch=z9hG4bK7A7CAC8E1253CB3802136E51E8472AE3 From: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 To: <sip:601@129.125.21.241>;tag=as4618ff70 Contact: <sip:502@129.125.71.173:5060> Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 24314 ACK Max-Forwards: 70 Content-Length: 0


--- (9 headers 0 lines)---

<-- SIP read from 129.125.71.173:5060:


--- (0 headers 0 lines) Nat keepalive --- --- setup_rtp_connection --- find_call +++ find_call +++ setup_rtp_connection --- close_rtp_connection --- find_call +++ find_call +++ close_rtp_connection --- onCallCleared ooh323c_o_2 --- find_call +++ find_call --- ooh323_hangup 

   hanging 601

+++ ooh323_hangup 

 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/502-083b1650' in macro 'dial'
 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/502-083b1650' in macro 'exten-vm'
 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/502-083b1650'

Scheduling destruction of call '39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173' in 32000 ms set_destination: Parsing <sip:502@129.125.71.173:5060> for address/port to send to set_destination: set destination to 129.125.71.173, port 5060 Reliably Transmitting (no NAT) to 129.125.71.173:5060: BYE sip:502@129.125.71.173:5060 SIP/2.0 Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK221b90c0 From: <sip:601@129.125.21.241>;tag=as4618ff70 To: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 Contact: <sip:601@129.125.21.241> Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0


---

<-- SIP read from 129.125.71.173:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK221b90c0 From: <sip:601@129.125.21.241>;tag=as4618ff70 To: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204 Contact: <sip:502@129.125.71.173:5060> Call-ID: 39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173 CSeq: 102 BYE Server: X-Lite release 1105d Content-Length: 0


--- (9 headers 0 lines)--- Destroying call '39573BC3-AB5C-189A-DB2D-EFD3CDA07489@129.125.71.173' --- ooh323_destroy 

Destroying 601

+++ ooh323_destroy 



== Gatekeeper == 
== Xlite ==
== Gnomemeeting ==
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