Difference between revisions of "SIP to H323"

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  <-- SIP read from 129.125.71.173:5060:
 
  <-- SIP read from 129.125.71.173:5060:
 
  SIP/2.0 200 Ok
 
  SIP/2.0 200 Ok
−
Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK221b90c0
+
Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK221b90c0
 
  From: <sip:601@129.125.21.241>;tag=as4618ff70
 
  From: <sip:601@129.125.21.241>;tag=as4618ff70
 
  To: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204
 
  To: "Fabio - xlite@wingtip113" <sip:502@129.125.21.241>;tag=1975923204

Revision as of 14:16, 12 March 2007

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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