asterisk*CLI> Verbosity is at least 100 Core debug is at least 100 asterisk*CLI>  -- Starting simple switch on 'DAHDI/6-1' asterisk*CLI>  -- Executing [my-eutelia-number@from-internal:1] Set("DAHDI/6-1", "DYNAMIC_FEATURES=automixmon") in new stack -- Executing [my-eutelia-number@from-internal:2] Dial("DAHDI/6-1", "WOOMERA/1/my-eutelia-number,,TX") in new stack **[WOOMERA]** +++REQ WOOMERA/1/my-eutelia-number-e116 **[WOOMERA]** +++GETOPT WOOMERA/1/my-eutelia-number-e116 -- Called 1/my-eutelia-number asterisk*CLI> **[WOOMERA]** Receive Message: {default} [localhost/42420] -------------------------------------------------------------------------------- EVENT HELLO Sangoma Media Gateway Supported-Protocols: TDM Version: v1.72 Remote-Address: 127.0.0.1 Remote-Port: 46886 Raw-Format: ALAW xUDP-Seq: Disabled xUDP-Seq-Err: 0 xNative-Bridge: Enabled **[WOOMERA]** Send Message: {default} [localhost/42420] -------------------------------------------------------------------------------- CALL 1/my-eutelia-number Raw-Audio: 127.0.0.1:20008 Local-Name: ! Local-Number: Presentation:0 Screening:0 Bearer-Cap:SPEECH uil1p:G_711_ALAW RDNIS: xCalledTon:255 xCalledNpi:255 xCallingTon:255 xCallingNpi:255 xRdnisTon:255 xRdnisNpi:255 xCustom: asterisk*CLI> **[WOOMERA]** Receive Message: {default} [localhost/42420] -------------------------------------------------------------------------------- 100 Trying asterisk*CLI> Reliably Transmitting (no NAT) to 83.211.227.21:5060: OPTIONS sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK6e5da491 Max-Forwards: 70 From: "asterisk" ;tag=as77125a39 To: Contact: Call-ID: 0a3446eb0564ade836f99ab32b3ca46f@my-asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Mon, 11 Apr 2011 22:15:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.1.158:5060: OPTIONS sip:zJZlTpTaQyFc4k0voKjL@192.168.1.158 SIP/2.0 Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK003d95df Max-Forwards: 70 From: "asterisk" ;tag=as797984e6 To: Contact: Call-ID: 386d075f3cfe44f3710ef902339bbaf1@my-asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Mon, 11 Apr 2011 22:15:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK6e5da491 From: "asterisk" ;tag=as77125a39 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.72f5 Call-ID: 0a3446eb0564ade836f99ab32b3ca46f@my-asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '0a3446eb0564ade836f99ab32b3ca46f@my-asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.1.158:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK003d95df To: ;tag=q4t6gf3rgphc76ip4l9h From: "asterisk" ;tag=as797984e6 Call-ID: 386d075f3cfe44f3710ef902339bbaf1@my-asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: application/sdp Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '386d075f3cfe44f3710ef902339bbaf1@my-asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI> **[WOOMERA]** Receive Message: {default} [localhost/42420] -------------------------------------------------------------------------------- EVENT PROCEED s1c1 Channel-Name: g1/1 Unique-Call-Id: s1c1-1967513926-2044897763 **[WOOMERA]** Queue Event: {default} [PROCEED] asterisk*CLI> **[WOOMERA]** Receive Message: {default} [localhost/42420] -------------------------------------------------------------------------------- 201 Accepted Unique-Call-Id: s1c1-1967513926-2044897763 **[WOOMERA]** Receive Message: {default} [localhost/42420] -------------------------------------------------------------------------------- EVENT MEDIA s1c1 AUDIO Unique-Call-Id: s1c1-1967513926-2044897763 Raw-Audio: 127.0.0.1:10008 Call-ID: s1c1 Raw-Format: PCM-16 DTMF: OutofBand **[WOOMERA]** HW DTMF supported s1c1-1967513926-2044897763 -- WOOMERA/g1/1 is ringing asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.3.3:2048: OPTIONS sip:104@192.168.3.3:2048;line=k5l8dvj0 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK6a727a1d Max-Forwards: 70 From: "asterisk" ;tag=as6f5e9697 To: Contact: Call-ID: 084274360dc15a5059e3650217dc4bb8@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Mon, 11 Apr 2011 22:15:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.3.3:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK6a727a1d From: "asterisk" ;tag=as6f5e9697 To: Call-ID: 084274360dc15a5059e3650217dc4bb8@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.31 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '084274360dc15a5059e3650217dc4bb8@192.168.3.1:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> INVITE sip:my-eutelia-number@my-asterisk-public-ip:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKca3c.c77e3733.0 Via: SIP/2.0/UDP 195.62.226.6:5060;rport=64016;received=195.62.226.6;x-route-tag="tgrp:Slot6";branch=z9hG4bK1D32AE9286 From: ;tag=61ECD74-B7 To: Call-ID: FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 User-Agent: Cisco-SIPGateway/IOS-12.x CSeq: 101 INVITE Max-Forwards: 9 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 419 P-hint: 2 Niente 2 v=0 o=CiscoSystemsSIP-GW-UserAgent 8188 3640 IN IP4 195.62.226.6 s=SIP Call c=IN IP4 83.211.223.197 t=0 0 m=audio 59246 RTP/AVP 18 8 0 4 3 125 101 c=IN IP4 83.211.223.197 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=5.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:125 X-CCD/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (16 headers 17 lines) --- == Using UDPTL CoS mark 5 Sending to 83.211.227.21:5060 (no NAT) Using INVITE request as basis request - FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 Found peer 'eutelia' for '195.62.226.6' from 83.211.227.21:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 125 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format GSM for ID 3 Found audio description format X-CCD for ID 125 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) Peer audio RTP is at port 83.211.223.197:59246 Looking for my-eutelia-number in eutelia-in (domain my-asterisk-public-ip:5060) list_route: hop: <--- Transmitting (no NAT) to 83.211.227.21:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKca3c.c77e3733.0;received=83.211.227.21 Via: SIP/2.0/UDP 195.62.226.6:5060;rport=64016;received=195.62.226.6;x-route-tag="tgrp:Slot6";branch=z9hG4bK1D32AE9286 Record-Route: From: ;tag=61ECD74-B7 To: Call-ID: FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 CSeq: 101 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [my-eutelia-number@eutelia-in:1] Set("SIP/eutelia-0000000e", "FAXFILE=/var/spool/asterisk/fax/prova4.tif") in new stack -- Executing [my-eutelia-number@eutelia-in:2] ReceiveFAX("SIP/eutelia-0000000e", "/var/spool/asterisk/fax/prova4.tif") in new stack -- Channel 'SIP/eutelia-0000000e' receiving FAX '/var/spool/asterisk/fax/prova4.tif' Audio is at 5060 Adding codec 0x8 (alaw) to SDP <--- Reliably Transmitting (no NAT) to 83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKca3c.c77e3733.0;received=83.211.227.21 Via: SIP/2.0/UDP 195.62.226.6:5060;rport=64016;received=195.62.226.6;x-route-tag="tgrp:Slot6";branch=z9hG4bK1D32AE9286 Record-Route: From: ;tag=61ECD74-B7 To: ;tag=as1c9a57dc Call-ID: FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 CSeq: 101 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 195 v=0 o=root 1957904998 1957904998 IN IP4 my-asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 my-asterisk-public-ip t=0 0 m=audio 19588 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv <------------> asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> ACK sip:my-eutelia-number@my-asterisk-public-ip:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKca3c.c77e3733.2 Via: SIP/2.0/UDP 195.62.226.6:5060;rport=64016;received=195.62.226.6;x-route-tag="tgrp:Slot6";branch=z9hG4bK1D32AEA2C6 From: ;tag=61ECD74-B7 To: ;tag=as1c9a57dc Call-ID: FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 Max-Forwards: 9 CSeq: 101 ACK C asterisk*CLI> ontent-Length: 0 P-hint: rr-enforced <-------------> --- (11 headers 0 lines) --- asterisk*CLI> **[WOOMERA]** Receive Message: {default} [localhost/42420] -------------------------------------------------------------------------------- EVENT CONNECT s1c1 Unique-Call-Id: s1c1-1967513926-2044897763 -- WOOMERA/g1/1 answered DAHDI/6-1 **[WOOMERA]** +++SETOPT WOOMERA/g1/1 [Apr 12 00:15:05] NOTICE[7062]: chan_woomera.c:5010 tech_indicate: Don't know how to indicate condition 20 asterisk*CLI> **[WOOMERA]** +++SETOPT WOOMERA/g1/1 asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2da9ec218ffcdac83068f10b26d51e16;rport From: "152" ;tag=1780595225 To: "152" Call-ID: 2654950264@192_168_1_151 CSeq: 2387 REGISTER Contact: "152" Max-Forwards: 70 User-Agent: C450 IP010720000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk*CLI> Sending to 192.168.1.151:5060 (no NAT) asterisk*CLI>  <--- Transmitting (no NAT) to 192.168.1.151:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2da9ec218ffcdac83068f10b26d51e16;received=192.168.1.151;rport=5060 From: "152" ;tag=1780595225 To: "152" ;tag=as727b01bb Call-ID: 2654950264@192_168_1_151 CSeq: 2387 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer W asterisk*CLI> WW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2336e3a0" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2654950264@192_168_1_151' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bKaa7d951784b01b21fe70a5af8953bd4;rport From: "152" ;tag=1780595225 To: "152" Call-ID: 2654950264@192_168_1_151 CSeq: 2388 REGISTER Contact: "152" Authorization: Digest username="152", realm="asterisk", algorithm=MD5, uri="sip:192.168.3.1", nonce="2336e3a0", response="b6329b4cb705cefa8a2814584ff17197" Max-Forwards: 70 User-Agent: C450 IP010720000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (13 headers 0 lines) --- asterisk*CLI> Sending to 192.168.1.151:5060 (no NAT) asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.1.151:5060: OPTIONS sip:152@192.168.1.151:5060 SIP/2.0 Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK70203a8b Max-Forwards: 70 From: "asterisk" ;tag=as58baa736 To: Contact: Call-ID: 6d4581c7569a21e25f40f2446691b800@my-asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Mon, 11 Apr 2011 22:15:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.1.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bKaa7d951784b01b21fe70a5af8953bd4;received=192.168.1.151;rport=5060 From: "152" ;tag=1780595225 To: "152" ;tag=as727b01bb Call-ID: 2654950264@192_168_1_151 CSeq: 2388 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 180 Contact: ;expires=180 Date: Mon, 11 Apr 2011 22:15:07 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2654950264@192_168_1_151' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK70203a8b From: "asterisk" ;tag=as58baa736 To: ;tag=180371390 Call-ID: 6d4581c7569a21e25f40f2446691b800@my-asterisk-public-ip:5060 CSeq: 102 OPTIONS Contact: "152" Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 < asterisk*CLI> -------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '6d4581c7569a21e25f40f2446691b800@my-asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.4:5060;branch=z9hG4bKa45835fb218cf2bcc82f909dc2d3df53;rport From: "102" ;tag=592953543 To: "102" Call-ID: 2999504275@192_168_3_4 CSeq: 342 REGISTER Contact: "102" Max-Forwards: 70 User-Agent: C450 IP010720000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.3.4:5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.4:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.4:5060;branch=z9hG4bKa45835fb218cf2bcc82f909dc2d3df53;received=192.168.3.4;rport=5060 From: "102" ;tag=592953543 To: "102" ;tag=as47c7df17 Call-ID: 2999504275@192_168_3_4 CSeq: 342 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f4ad33c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2999504275@192_168_3_4' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.4:5060;branch=z9hG4bKa64ce4f53159307146f570ea298032;rport From: "102" ;tag=592953543 To: "102" Call-ID: 2999504275@192_168_3_4 CSeq: 343 REGISTER Contact: "102" Authorization: Digest username="102", realm="asterisk", algorithm=MD5, uri="sip:192.168.3.1", nonce="4f4ad33c", response="cbe697cb710226bc76f2c5c7df6012bf" Max-Forwards: 70 asterisk*CLI> User-Agent: C450 IP010720000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.3.4:5060 (no NAT) Reliably Transmitting (no NAT) to 192.168.3.4:5060: OPTIONS sip:102@192.168.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK610c3a15 Max-Forwards: 70 From: "asterisk" ;tag=as49d08e10 To: Contact: Call-ID: 4ba3511745770e3c528a3a87681dca29@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Mon, 11 Apr 2011 22:15:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.3.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.4:5060;branch=z9hG4bKa64ce4f53159307146f570ea298032;received=192.168.3.4;rport=5060 From: "102" ;tag=592953543 To: "102" ;tag=as47c7df17 Call-ID: 2999504275@192_168_3_4 CSeq: 343 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 180 Contact: ;expires=180 Date: Mon, 11 Apr 2011 22:15:08 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2999504275@192_168_3_4' in 32000 ms (Method: REGISTER) asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 Reliably Transmitting (no NAT) to 83.211.227.21:5060: INVITE sip:195.62.226.6:64016 SIP/2.0 Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK7a10a8a5 Route: Max-Forwards: 70 From: ;tag=as1c9a57dc To: ;tag=61ECD74-B7 Contact: Call-ID: FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 279 v=0 o=root 1957904998 1957904999 IN IP4 my-asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 my-asterisk-public-ip t=0 0 m=image 4515 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK7a10a8a5 From: ;tag=as1c9a57dc To: ;tag=61ECD74-B7 Call-ID: FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 CSeq: 102 INVITE Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK7a10a8a5 From: ;tag=as1c9a57dc To: ;tag=61ECD74-B7 Date: Mon, 11 Apr 2011 22:15:08 GMT Call-ID: FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 266 v=0 o=CiscoSystemsSIP-GW-UserAgent 8188 3641 IN IP4 195.62.226.6 s=SIP Call c=IN IP4 83.211.223.197 t=0 0 m=image 57194 udptl t38 c=IN IP4 83.211.223.197 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:72 <-------------> --- (15 headers 11 lines) --- asterisk*CLI> Got T.38 offer in SDP in dialog FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. asterisk*CLI> set_destination: Parsing for address/port to send to asterisk*CLI> set_destination: set destination to 83.211.227.21:5060 Transmitting (no NAT) to 83.211.227.21:5060: ACK sip:195.62.226.6:5060 SIP/2.0 Via: SIP/2.0/UDP my-asterisk-public-ip:5060;branch=z9hG4bK1eef8613 Route: Max-Forwards: 70 From: ;tag=as1c9a57dc To: ;tag=61ECD74-B7 Contact: Call-ID: FC8C6E01-63BF11E0-A67ADEC0-F5E8E90C@195.62.226.6 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK610c3a15 From: "asterisk" ;tag=as49d08e10 To: ;tag=1163301585 Call-ID: 4ba3511745770e3c528a3a87681dca29@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: "102" Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '4ba3511745770e3c528a3a87681dca29@192.168.3.1:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bKcba515b600 From: "wind8" ;tag=45d2ae59 To: "wind8" Call-ID: 4c90573d184f8d100eb621e8576c0de3@192.168.3.2 Contact: CSeq: 13800 REGISTER Max-Forwards: 70 Expires: 60 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Authorization: Digest username="wind8",realm="asterisk",nonce="271a820a",response="618a99c13203d1918ede221aaabd4af2",uri="sip:192.168.3.1",algorithm=MD5 User-Agent: Mv-37x (904290) Content-Length: 0 <-------------> --- (13 headers 0 lines) --- asterisk*CLI> Sending to 192.168.3.2:5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bKcba515b600;received=192.168.3.2;rport=5060 From: "wind8" ;tag=45d2ae59 To: "wind8" ;tag=as14ec844b Call-ID: 4c90573d184f8d100eb621e8576c0de3@192.168.3.2 CSeq: 13800 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3590ce54" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4c90573d184f8d100eb621e8576c0de3@192.168.3.2' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bK87af931855 From: "wind8" ;tag=45d2ae59 To: "wind8" Call-ID: 4c90573d184f8d100eb621e8576c0de3@192.168.3.2 Contact: CSeq: 13801 REGISTER Max-Forwards: 70 Expires: 60 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Authorization: Digest username="wind8",realm="asterisk",nonce="3590ce54",response="5b8544345ea8bf00c1a26f417385709a",uri="sip:192.168.3.1",algorithm=MD5 User-Agent: Mv-37x (904290) Content-Length: 0 <-------------> --- (13 headers 0 lines) --- asterisk*CLI> Sending to 192.168.3.2:5060 (no NAT) asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.3.2:5060: OPTIONS sip:wind8@192.168.3.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK52d34589 Max-Forwards: 70 From: "asterisk" ;tag=as4f1f9368 To: Contact: Call-ID: 2000e61f740ac3a839885b075bbcf6ea@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Mon, 11 Apr 2011 22:15:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK87af931855;received=192.168.3.2;rport=5060 From: "wind8" ;tag=45d2ae59 To: "wind8" ;tag=as14ec844b Call-ID: 4c90573d184f8d100eb621e8576c0de3@192.168.3.2 CSeq: 13801 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Mon, 11 Apr 2011 22:15:17 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4c90573d184f8d100eb621e8576c0de3@192.168.3.2' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK52d34589 From: "asterisk" ;tag=as4f1f9368 To: ;tag=1255d5cc Call-ID: 2000e61f740ac3a839885b075bbcf6ea@192.168.3.1:5060 CSeq: 102 OPTIONS Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '2000e61f740ac3a839885b075bbcf6ea@192.168.3.1:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> asterisk*CLI> Disconnected from Asterisk server