=~=~=~=~=~=~=~=~=~=~=~= PuTTY logasterisk1*CLI> asterisk1*CLI> REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.65.55.7:5060: REGISTER sip:193.65.55.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK02c5e760 From: ;tag=as36472583 To: Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 138 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="liorh", realm="SIP_Auth_Realm", algorithm=MD5, uri="sip:193.65.55.4", nonce="e76ba7b3a1d3a3", response="5e5c71cdaa3d3a944b1d702029c77564", opaque="" Expires: 30 Contact: Event: registration Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: SIP/2.0 200 OK To: ;tag=015e055f From: ;tag=as36472583 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK02c5e760 Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 138 REGISTER Contact: ;expires=25 Content-Length: 0 --- (8 headers 0 lines)--- Scheduling destruction of call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' in 32000 ms asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: INVITE sip:358753250901@192.168.200.53 SIP/2.0 To: From: "tzvika";tag=73c4627d Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-83bc6b0ef377853c919c-1-cHBkMDliMjAyNjQ4YTk1NDdmNGFmZg..-d87543- Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383321 INVITE Contact: Max-Forwards: 9 Content-Type: application/SDP Supported: timer Content-Length: 273 Min-SE: 180 Session-Expires: 180;refresher=uas v=0 o=- 21172806 1968635629 IN IP4 193.65.55.7 s=eyeBeam c=IN IP4 193.65.55.7 t=0 0 m=audio 12052 RTP/AVP 18 8 0 3 100 6 5 101 a=alt:1 1 : 13606585 847A136A 192.168.40.53 5060 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (13 headers 11 lines)--- Using INVITE request as basis request - 2549a7b1def23dd1913f798aeb35ab1c Sending to 193.65.55.7 : 5060 (non-NAT) Found peer 'sip.suomenpuhelin3' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 5 Found RTP audio format 101 Peer audio RTP is at port 193.65.55.7:12052 Found description format speex Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 358753250901 in from-pstn list_route: hop: Transmitting (no NAT) to 193.65.55.7:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-83bc6b0ef377853c919c-1-cHBkMDliMjAyNjQ4YTk1NDdmNGFmZg..-d87543- From: "tzvika";tag=73c4627d To: Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383321 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing SetVar("SIP/358753250903-28b1", "FROM_DID=358753250901") in new stack -- Executing Goto("SIP/358753250903-28b1", "ext-local|901|1") in new stack -- Goto (ext-local,901,1) -- Executing Macro("SIP/358753250903-28b1", "exten-vm|901@default|901") in new stack -- Executing SetVar("SIP/358753250903-28b1", "FROMCONTEXT=exten-vm") in new stack -- Executing Macro("SIP/358753250903-28b1", "record-enable|901|IN") in new stack -- Executing GotoIf("SIP/358753250903-28b1", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing GotoIf("SIP/358753250903-28b1", "0?5:8") in new stack -- Goto (macro-record-enable,s,8) -- Executing GotoIf("SIP/358753250903-28b1", "0?9:12") in new stack -- Goto (macro-record-enable,s,12) -- Executing DBget("SIP/358753250903-28b1", "RecEnable=RECORD-IN/901") in new stack -- DBget: varname=RecEnable, family=RECORD-IN, key=901 -- DBget: Value not found in database. -- Executing SetVar("SIP/358753250903-28b1", "CALLFILENAME=20051110-101510-1131635710.8") in new stack -- Executing GotoIf("SIP/358753250903-28b1", "0?15:99") in new stack -- Goto (macro-record-enable,s,99) -- Executing NoOp("SIP/358753250903-28b1", "NO RECORDING NEEDED") in new stack -- Executing Macro("SIP/358753250903-28b1", "dial|15|trTwW|901") in new stack -- Executing GotoIf("SIP/358753250903-28b1", "0?4:2") in new stack -- Goto (macro-dial,s,2) -- Executing GotoIf("SIP/358753250903-28b1", "0?4:3") in new stack -- Goto (macro-dial,s,3) -- Executing SetCIDName("SIP/358753250903-28b1", "tzvika") in new stack -- Executing AGI("SIP/358753250903-28b1", "dialparties.agi") in new stack asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi asterisk1*CLI> -- dialparties.agi: priority = 4 asterisk1*CLI> -- dialparties.agi: callingani2 = 0 asterisk1*CLI> -- dialparties.agi: accountcode = asterisk1*CLI> -- dialparties.agi: channel = SIP/358753250903-28b1 asterisk1*CLI> -- dialparties.agi: callerid = 0753250904 asterisk1*CLI> -- dialparties.agi: callington = 0 asterisk1*CLI> -- dialparties.agi: context = macro-dial asterisk1*CLI> -- dialparties.agi: dnid = 358753250901 asterisk1*CLI> -- dialparties.agi: request = dialparties.agi asterisk1*CLI> -- dialparties.agi: extension = s asterisk1*CLI> -- dialparties.agi: calleridname = tzvika asterisk1*CLI> -- dialparties.agi: language = en asterisk1*CLI> -- dialparties.agi: uniqueid = 1131635710.8 asterisk1*CLI> -- dialparties.agi: callingpres = 0 asterisk1*CLI> -- dialparties.agi: rdnis = unknown asterisk1*CLI> -- dialparties.agi: type = SIP asterisk1*CLI> -- dialparties.agi: callingtns = 0 asterisk1*CLI> -- dialparties.agi: enhanced = 0.0 asterisk1*CLI> dialparties.agi: Caller ID name and number are '0753250904' asterisk1*CLI> -- dialparties.agi: Added extension 901 to extension map asterisk1*CLI> -- dialparties.agi: Extension 901 cf is disabled asterisk1*CLI> -- dialparties.agi: Extension 901 do not disturb is disabled asterisk1*CLI> == Parsing '/etc/asterisk/manager.conf': Found asterisk1*CLI> == Parsing '/etc/asterisk/manager_custom.conf': Found asterisk1*CLI> == Manager 'admin' logged on from 127.0.0.1 asterisk1*CLI> == Manager 'admin' logged off from 127.0.0.1 asterisk1*CLI> dialparties.agi: Extension 901 has call waiting disabled asterisk1*CLI> -- dialparties.agi: DbSet CALLTRACE/901 to 0753250904 asterisk1*CLI> dialparties.agi: Dial string is SIP/901|15|trTwW asterisk1*CLI> -- AGI Script dialparties.agi completed, returning 0 asterisk1*CLI> -- Executing Dial("SIP/358753250903-28b1", "SIP/901|15|trTwW") in new stack asterisk1*CLI> We're at 192.168.200.53 port 17140 asterisk1*CLI> Answering/Requesting with root capability 0x4 (ulaw) asterisk1*CLI> Answering with preferred capability 0x8 (alaw) asterisk1*CLI> Answering with non-codec capability 0x1 (telephone-event) asterisk1*CLI> 12 headers, 11 lines asterisk1*CLI> Reliably Transmitting (no NAT) to 192.168.200.18:5060: INVITE sip:901@192.168.200.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK74379512 From: "tzvika" ;tag=as78557130 To: Contact: Call-ID: 303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 10 Nov 2005 15:15:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 244 v=0 o=root 10889 10889 IN IP4 192.168.200.53 s=session c=IN IP4 192.168.200.53 t=0 0 m=audio 17140 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 - - - - --- asterisk1*CLI> -- Called 901 asterisk1*CLI> Transmitting (no NAT) to 193.65.55.7:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-83bc6b0ef377853c919c-1-cHBkMDliMjAyNjQ4YTk1NDdmNGFmZg..-d87543- From: "tzvika";tag=73c4627d To: ;tag=as31ad5397 Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383321 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> Retransmitting #1 (no NAT) to 192.168.200.18:5060: INVITE sip:901@192.168.200.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK74379512 From: "tzvika" ;tag=as78557130 To: Contact: Call-ID: 303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 10 Nov 2005 15:15:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 244 v=0 o=root 10889 10889 IN IP4 192.168.200.53 s=session c=IN IP4 192.168.200.53 t=0 0 m=audio 17140 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 - - - - --- asterisk1*CLI> <-- SIP read from 192.168.200.18:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK74379512 From: "tzvika";tag=as78557130 To: Call-ID: 303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53 CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 192.168.200.18:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK74379512 From: "tzvika";tag=as78557130 To: ;tag=414864649 Call-ID: 303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53 CSeq: 102 INVITE Contact: Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI> -- SIP/901-8441 is ringing asterisk1*CLI> <-- SIP read from 192.168.200.18:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK74379512 From: "tzvika";tag=as78557130 To: ;tag=414864649 Call-ID: 303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53 CSeq: 102 INVITE Contact: Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI> -- SIP/901-8441 is ringing asterisk1*CLI> <-- SIP read from 192.168.200.18:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK74379512 From: "tzvika";tag=as78557130 To: ;tag=414864649 Call-ID: 303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53 CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER Content-Type: application/sdp Content-Length: 180 v=0 o=901 123456 654321 IN IP4 192.168.200.18 s=session c=IN IP4 192.168.200.18 t=0 0 m=audio 7080 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 --- (10 headers 8 lines)--- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.200.18:7080 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.18, port 5060 Transmitting (no NAT) to 192.168.200.18:5060: ACK sip:901@192.168.200.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK7f9110b9 From: "tzvika" ;tag=as78557130 To: ;tag=414864649 Contact: Call-ID: 303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- asterisk1*CLI> -- SIP/901-8441 answered SIP/358753250903-28b1 asterisk1*CLI> We're at 192.168.200.53 port 15046 asterisk1*CLI> Answering with preferred capability 0x4 (ulaw) asterisk1*CLI> Answering with preferred capability 0x8 (alaw) asterisk1*CLI> Answering with non-codec capability 0x1 (telephone-event) asterisk1*CLI> Reliably Transmitting (no NAT) to 193.65.55.7:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-83bc6b0ef377853c919c-1-cHBkMDliMjAyNjQ4YTk1NDdmNGFmZg..-d87543- From: "tzvika";tag=73c4627d To: ;tag=as31ad5397 Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383321 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=root 10889 10889 IN IP4 192.168.200.53 s=session c=IN IP4 192.168.200.53 t=0 0 m=audio 15046 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 - - - - --- asterisk1*CLI> -- Attempting native bridge of SIP/358753250903-28b1 and SIP/901-8441 asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: ACK sip:358753250901@192.168.200.53 SIP/2.0 To: ;tag=as31ad5397 From: "tzvika";tag=73c4627d Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-1fc0360727f9ee18-1-cHBkOTk3ODM1YWVhODBhYTM2NDVjZA..-d87543- Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383321 ACK Contact: Max-Forwards: 69 Content-Length: 0 asterisk1*CLI> --- (9 headers 0 lines)--- asterisk1*CLI> Destroying call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' asterisk1*CLI> REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.65.55.7:5060: REGISTER sip:193.65.55.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK48ed583d From: ;tag=as6ecfc5b5 To: Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 139 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="liorh", realm="SIP_Auth_Realm", algorithm=MD5, uri="sip:193.65.55.4", nonce="e76ba7b3a1d3a3", response="5e5c71cdaa3d3a944b1d702029c77564", opaque="" Expires: 30 Contact: Event: registration Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: SIP/2.0 200 OK To: ;tag=b6171650 From: ;tag=as6ecfc5b5 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK48ed583d Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 139 REGISTER Contact: ;expires=25 Content-Length: 0 --- (8 headers 0 lines)--- Scheduling destruction of call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' in 32000 ms asterisk1*CLI> Destroying call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' asterisk1*CLI> REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.65.55.7:5060: REGISTER sip:193.65.55.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK71bfa5f0 From: ;tag=as278b9957 To: Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 140 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="liorh", realm="SIP_Auth_Realm", algorithm=MD5, uri="sip:193.65.55.4", nonce="e76ba7b3a1d3a3", response="5e5c71cdaa3d3a944b1d702029c77564", opaque="" Expires: 30 Contact: Event: registration Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: SIP/2.0 200 OK To: ;tag=7abcad16 From: ;tag=as278b9957 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK71bfa5f0 Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 140 REGISTER Contact: ;expires=25 Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI> Scheduling destruction of call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' in 32000 ms asterisk1*CLI> <-- SIP read from 192.168.200.18:5060: REGISTER sip:192.168.200.53 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.18:5060;branch=z9hG4bK1284416392 From: 0753250901;tag=3397574751 To: 0753250901 Call-ID: 507126886@192.168.200.18 CSeq: 3434 REGISTER Contact: Max-Forwards: 70 Expires: 300 User-Agent: Clipcomm CP-100E v1.2.12 (050718) Content-Length: 0 --- (11 headers 0 lines)--- Using latest request as basis request Sending to 192.168.200.18 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.200.18:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.200.18:5060;branch=z9hG4bK1284416392 From: 0753250901;tag=3397574751 To: 0753250901 Call-ID: 507126886@192.168.200.18 CSeq: 3434 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.200.18:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.200.18:5060;branch=z9hG4bK1284416392 From: 0753250901;tag=3397574751 To: 0753250901;tag=as7c670315 Call-ID: 507126886@192.168.200.18 CSeq: 3434 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="2f705cf0" Content-Length: 0 --- Scheduling destruction of call '507126886@192.168.200.18' in 15000 ms asterisk1*CLI> <-- SIP read from 192.168.200.18:5060: REGISTER sip:192.168.200.53 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.18:5060;branch=z9hG4bK506583995 From: 0753250901;tag=3397574751 To: 0753250901 Call-ID: 507126886@192.168.200.18 CSeq: 3435 REGISTER Contact: Authorization: Digest username="901", realm="asterisk", nonce="2f705cf0", uri="sip:192.168.200.53", response="a4e7d90ba58a4cdd4ffcbac1d9deaa5f", algorithm=MD5 Max-Forwards: 70 Expires: 300 User-Agent: Clipcomm CP-100E v1.2.12 (050718) Content-Length: 0 --- (12 headers 0 lines)--- Using latest request as basis request Sending to 192.168.200.18 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.200.18:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.200.18:5060;branch=z9hG4bK506583995 From: 0753250901;tag=3397574751 To: 0753250901 Call-ID: 507126886@192.168.200.18 CSeq: 3435 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> Transmitting (no NAT) to 192.168.200.18:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.18:5060;branch=z9hG4bK506583995 From: 0753250901;tag=3397574751 To: 0753250901;tag=as7c670315 Call-ID: 507126886@192.168.200.18 CSeq: 3435 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 300 Contact: ;expires=300 Date: Thu, 10 Nov 2005 15:15:54 GMT Content-Length: 0 --- Scheduling destruction of call '507126886@192.168.200.18' in 15000 ms asterisk1*CLI> 11 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.200.18:5060: NOTIFY sip:901@192.168.200.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK35537076 From: "asterisk" ;tag=as3190e91d To: Contact: Call-ID: 70f87cc820a86d7454929fed65aa5722@192.168.200.53 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 80 Message-Account: sip:asterisk@ Messages-Waiting: no Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '70f87cc820a86d7454929fed65aa5722@192.168.200.53' in 15000 ms asterisk1*CLI> <-- SIP read from 192.168.200.18:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK35537076 From: "asterisk";tag=as3190e91d To: ;tag=2336468754 Call-ID: 70f87cc820a86d7454929fed65aa5722@192.168.200.53 CSeq: 102 NOTIFY Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '70f87cc820a86d7454929fed65aa5722@192.168.200.53' asterisk1*CLI> Destroying call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' asterisk1*CLI> REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.65.55.7:5060: REGISTER sip:193.65.55.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK2f287257 From: ;tag=as59423a1c To: Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 141 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="liorh", realm="SIP_Auth_Realm", algorithm=MD5, uri="sip:193.65.55.4", nonce="e76ba7b3a1d3a3", response="5e5c71cdaa3d3a944b1d702029c77564", opaque="" Expires: 30 Contact: Event: registration Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: SIP/2.0 200 OK To: ;tag=f0a5520b From: ;tag=as59423a1c Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK2f287257 Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 141 REGISTER Contact: ;expires=25 Content-Length: 0 --- (8 headers 0 lines)--- Scheduling destruction of call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' in 32000 ms asterisk1*CLI> Destroying call '507126886@192.168.200.18' asterisk1*CLI> Destroying call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' asterisk1*CLI> REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.65.55.7:5060: REGISTER sip:193.65.55.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK16ed9b5e From: ;tag=as577b2724 To: Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 142 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="liorh", realm="SIP_Auth_Realm", algorithm=MD5, uri="sip:193.65.55.4", nonce="e76ba7b3a1d3a3", response="5e5c71cdaa3d3a944b1d702029c77564", opaque="" Expires: 30 Contact: Event: registration Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: SIP/2.0 200 OK To: ;tag=c90b691e From: ;tag=as577b2724 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK16ed9b5e Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 142 REGISTER Contact: ;expires=25 Content-Length: 0 --- (8 headers 0 lines)--- Scheduling destruction of call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' in 32000 ms asterisk1*CLI> Destroying call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: INVITE sip:358753250901@192.168.200.53 SIP/2.0 To: ;tag=as31ad5397 From: "tzvika";tag=73c4627d Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-87efc108bcaf7607c77e-1-cHBkOTk3ODM1YWVhODBhYTM2NDVjZA..-d87543- Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383322 INVITE Contact: Max-Forwards: 70 Content-Type: application/SDP Content-Length: 286 v=0 o=- 21172806 1968635629 IN IP4 193.65.55.7 s=eyeBeam c=IN IP4 193.65.55.7 t=0 0 m=audio 12052 RTP/AVP 0 8 101 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=alt:1 1 : 13606585 847A136A 192.168.40.53 5060 a=sendrecv --- (10 headers 12 lines)--- Using INVITE request as basis request - 2549a7b1def23dd1913f798aeb35ab1c Sending to 193.65.55.7 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 193.65.55.7:12052 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 192.168.200.53 port 15046 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to 193.65.55.7:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-87efc108bcaf7607c77e-1-cHBkOTk3ODM1YWVhODBhYTM2NDVjZA..-d87543- From: "tzvika";tag=73c4627d To: ;tag=as31ad5397 Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383322 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=root 10889 10890 IN IP4 192.168.200.53 s=session c=IN IP4 192.168.200.53 t=0 0 m=audio 15046 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 - - - - --- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: ACK sip:358753250901@192.168.200.53 SIP/2.0 To: ;tag=as31ad5397 From: "tzvika";tag=73c4627d Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-3fac5a77e9796c6f-1-cHBkOTk3ODM1YWVhODBhYTM2NDVjZA..-d87543- Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383322 ACK Contact: Max-Forwards: 70 Content-Length: 0 asterisk1*CLI> --- (9 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: INVITE sip:358753250901@192.168.200.53 SIP/2.0 To: ;tag=as31ad5397 From: "tzvika";tag=73c4627d Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-78bab13407ccfd53201a-1-cHBkOTk3ODM1YWVhODBhYTM2NDVjZA..-d87543- Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383323 INVITE Contact: Max-Forwards: 69 Content-Type: application/SDP Content-Length: 273 v=0 o=- 21172806 1968635630 IN IP4 193.65.55.7 s=eyeBeam c=IN IP4 193.65.55.7 t=0 0 m=audio 12052 RTP/AVP 18 8 0 3 100 6 5 101 a=alt:1 1 : 13606585 847A136A 192.168.40.53 5060 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (10 headers 11 lines)--- Using INVITE request as basis request - 2549a7b1def23dd1913f798aeb35ab1c Sending to 193.65.55.7 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 5 Found RTP audio format 101 Peer audio RTP is at port 193.65.55.7:12052 Found description format speex Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 192.168.200.53 port 15046 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to 193.65.55.7:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-78bab13407ccfd53201a-1-cHBkOTk3ODM1YWVhODBhYTM2NDVjZA..-d87543- From: "tzvika";tag=73c4627d To: ;tag=as31ad5397 Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383323 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=root 10889 10891 IN IP4 192.168.200.53 s=session c=IN IP4 192.168.200.53 t=0 0 m=audio 15046 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 - - - - --- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: ACK sip:358753250901@192.168.200.53 SIP/2.0 To: ;tag=as31ad5397 From: "tzvika";tag=73c4627d Via: SIP/2.0/UDP 193.65.55.7:5060;branch=z9hG4bK-d87543-74555129242dcb1c-1-cHBkOTk3ODM1YWVhODBhYTM2NDVjZA..-d87543- Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 326383323 ACK Contact: Max-Forwards: 69 Content-Length: 0 asterisk1*CLI> --- (9 headers 0 lines)--- asterisk1*CLI> REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.65.55.7:5060: REGISTER sip:193.65.55.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK1e8e453a From: ;tag=as600000d2 To: Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 143 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="liorh", realm="SIP_Auth_Realm", algorithm=MD5, uri="sip:193.65.55.4", nonce="e76ba7b3a1d3a3", response="5e5c71cdaa3d3a944b1d702029c77564", opaque="" Expires: 30 Contact: Event: registration Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: SIP/2.0 200 OK To: ;tag=799cf22f From: ;tag=as600000d2 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK1e8e453a Call-ID: 7677880310721c280a1da7b671ebc6d5@193.65.55.4 CSeq: 143 REGISTER Contact: ;expires=25 Content-Length: 0 --- (8 headers 0 lines)--- Scheduling destruction of call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' in 32000 ms asterisk1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.18, port 5060 Reliably Transmitting (no NAT) to 192.168.200.18:5060: BYE sip:901@192.168.200.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK36e39c92 From: "tzvika" ;tag=as78557130 To: ;tag=414864649 Contact: Call-ID: 303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/358753250903-28b1' in macro 'dial' == Spawn extension (macro-exten-vm, s, 3) exited non-zero on 'SIP/358753250903-28b1' in macro 'exten-vm' == Spawn extension (ext-local, 901, 1) exited non-zero on 'SIP/358753250903-28b1' asterisk1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 193.65.55.7, port 5060 Reliably Transmitting (no NAT) to 193.65.55.7:5060: BYE sip:AcF9LCp7HG7cc3bG5YgktpQvlPfeGCi@193.65.55.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK1d72df7b From: ;tag=as31ad5397 To: "tzvika";tag=73c4627d Contact: Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 193.65.55.7:5060: SIP/2.0 200 OK To: "tzvika";tag=73c4627d From: ;tag=as31ad5397 Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK1d72df7b Call-ID: 2549a7b1def23dd1913f798aeb35ab1c CSeq: 102 BYE Contact: Content-Length: 0 --- (8 headers 0 lines)--- Response message BYE arrived Destroying call '2549a7b1def23dd1913f798aeb35ab1c' asterisk1*CLI> <-- SIP read from 192.168.200.18:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK36e39c92 From: "tzvika";tag=as78557130 To: ;tag=414864649 Call-ID: 303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53 CSeq: 103 BYE Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '303f4c7452b98fda7e9bd7ad265445c1@192.168.200.53' asterisk1*CLI> s Destroying call '7677880310721c280a1da7b671ebc6d5@193.65.55.4' asterisk1*CLI> sip no debug asterisk1*CLI> SIP Debugging Disabled asterisk1*CLI>