Using INVITE request as basis request - 577c4eb4-ee8554f6-8a319bff@192.168.0.7 Sending to 192.168.0.7: 5060; transport UDP (non-NAT) Found peer '102' Reliably Transmitting (no NAT) to 192.168.0.7:5060:UDP with fd 15 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK869709106BE141CB;received=192.168.0.7 From: "Front Desk" ;tag=839BB768-7EBCC249 To: ;tag=as17d35f9e Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7 CSeq: 3 INVITE User-Agent: ATA Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40f6f98c" Content-Length: 0 --- UDP write: fd 15 Scheduling destruction of call '577c4eb4-ee8554f6-8a319bff@192.168.0.7' in 15000 ms asterisk1*CLI> <-- Sip read from 192.168.0.7:5060:UDP ACK sip:101@192.168.0.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK869709106BE141CB From: "Front Desk" ;tag=839BB768-7EBCC249 To: ;tag=as17d35f9e CSeq: 3 ACK Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- asterisk1*CLI> <-- Sip read from 192.168.0.7:5060:UDP INVITE sip:101@192.168.0.6;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735 From: "Front Desk" ;tag=839BB768-7EBCC249 To: CSeq: 4 INVITE Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.3.0019 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="102", realm="asterisk", nonce="40f6f98c", uri="sip:101@192.168.0.6;user=phone", response="8adc9ca5c6942ae325328deb151c13d9", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 1158970534 1158970534 IN IP4 192.168.0.7 s=Polycom IP Phone c=IN IP4 192.168.0.7 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines)--- Using INVITE request as basis request - 577c4eb4-ee8554f6-8a319bff@192.168.0.7 Sending to 192.168.0.7: 5060; transport UDP (non-NAT) Found peer '102' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.7:2228 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 101 in from-internal (domain 192.168.0.6;user=phone) list_route: hop: Transmitting (no NAT) to 192.168.0.7:5060:UDP with fd 15 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735;received=192.168.0.7 From: "Front Desk" ;tag=839BB768-7EBCC249 To: Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7 CSeq: 4 INVITE User-Agent: ATA Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- UDP write: fd 15 -- Executing Macro("SIP/102-08848eb0", "exten-vm|101|101") in new stack -- Executing Macro("SIP/102-08848eb0", "user-callerid") in new stack -- Executing GotoIf("SIP/102-08848eb0", "0?report") in new stack -- Executing GotoIf("SIP/102-08848eb0", "0?start") in new stack -- Executing Set("SIP/102-08848eb0", "REALCALLERIDNUM=102") in new stack -- Executing NoOp("SIP/102-08848eb0", "REALCALLERIDNUM is 102") in new stack -- Executing Set("SIP/102-08848eb0", "AMPUSER=102") in new stack -- Executing Set("SIP/102-08848eb0", "AMPUSERCIDNAME=Front Desk") in new stack -- Executing GotoIf("SIP/102-08848eb0", "0?report") in new stack -- Executing Set("SIP/102-08848eb0", "CALLERID(all)=Front Desk <102>") in new stack -- Executing NoOp("SIP/102-08848eb0", "Using CallerID "Front Desk" <102>") in new stack -- Executing Set("SIP/102-08848eb0", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/102-08848eb0", "VMBOX=101") in new stack -- Executing Set("SIP/102-08848eb0", "EXTTOCALL=101") in new stack -- Executing Set("SIP/102-08848eb0", "CFUEXT=") in new stack -- Executing Set("SIP/102-08848eb0", "RT=15") in new stack -- Executing Macro("SIP/102-08848eb0", "record-enable|101|IN") in new stack -- Executing GotoIf("SIP/102-08848eb0", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/102-08848eb0", "recordingcheck|20060922-191540|1158970540.5") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060922-191540|1158970540.5: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/102-08848eb0", "No recording needed") in new stack -- Executing GotoIf("SIP/102-08848eb0", "0?dolocaldial|1") in new stack -- Executing Macro("SIP/102-08848eb0", "dial|15|tr|101") in new stack -- Executing AGI("SIP/102-08848eb0", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi -- dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'Front Desk' number is '102' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 101 to extension map -- dialparties.agi: Extension 101 cf is disabled -- dialparties.agi: Extension 101 do not disturb is disabled > dialparties.agi: extnum: 101 > 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 101 -- dialparties.agi: DbSet CALLTRACE/101 to 102 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("SIP/102-08848eb0", "SIP/101|15|tr") in new stack UDP write: fd 15 -- Called 101 Transmitting (no NAT) to 192.168.0.7:5060:UDP with fd 15 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735;received=192.168.0.7 From: "Front Desk" ;tag=839BB768-7EBCC249 To: ;tag=as76448c74 Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7 CSeq: 4 INVITE User-Agent: ATA Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- UDP write: fd 15 -- SIP/101-088546e8 is ringing UDP write: fd 15 -- SIP/101-088546e8 answered SIP/102-08848eb0 UDP write: fd 15 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-08848eb0' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-08848eb0' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-08848eb0' Reliably Transmitting (no NAT) to 192.168.0.7:5060:UDP with fd 15 SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735;received=192.168.0.7 From: "Front Desk" ;tag=839BB768-7EBCC249 To: ;tag=as76448c74 Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7 CSeq: 4 INVITE User-Agent: ATA Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- UDP write: fd 15 asterisk1*CLI> <-- Sip read from 192.168.0.7:5060:UDP ACK sip:101@192.168.0.6;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735 From: "Front Desk" ;tag=839BB768-7EBCC249 To: ;tag=as76448c74 CSeq: 4 ACK Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '577c4eb4-ee8554f6-8a319bff@192.168.0.7'