<--- SIP read from UDP://217.64.49.49:5060 ---> INVITE sip:0143620918@sec1.acropolistelecom.net;transport=UDP SIP/2.0 Record-Route: Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK329d.88cf7091.0 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-7ce019c1828ee11c-1---d8754z- Max-Forwards: 70 Contact: To: From: "";tag=765e5e21 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.4829 Content-Length: 329 x-acro-client: Fcois x-acro-chemin: SEC-sec1 x-acro-nbappel: 971 x-acro-called: france_fixe x-acro-dst_ext: t38 x-acro-tarifplan: 1 x-acro-factuprefix: 0143 x-acro-factumin: 0.002 x-acro-credit: 0.00000 x-acro-sda-orig: sip:FCOIST38@sec1.acropolistelecom.net;transport=UDP x-acro-caller: france x-acro-caller: int x-acro-masquage: 1 x-acro-codec: T38 x-acro-t38: oui x-acro-dst_ext: t38 v=0 o=Zoiper_user 0 0 IN IP4 192.168.1.100 s=Zoiper_session c=IN IP4 192.168.1.100 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (30 headers 15 lines) --- == Using SIP RTP CoS mark 5 Sending to 217.64.49.49 : 5060 (no NAT) Using INVITE request as basis request - MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. No user 'FCOIST38' in SIP users list Found peer 'ser_sec1_g711' for 'FCOIST38' from 217.64.49.49:5060 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 110 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.100:8000 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format speex for ID 110 Found audio description format iLBC for ID 98 Got unsupported a:fmtp in SDP offer Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.100:8000 Looking for 0143620918 in ser_sec (domain sec1.acropolistelecom.net) list_route: hop: ares*CLI> <--- Transmitting (NAT) to 217.64.49.49:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK329d.88cf7091.0;received=217.64.49.49 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-7ce019c1828ee11c-1---d8754z- Record-Route: From: "";tag=765e5e21 To: Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.14-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [s@macro-envoi_sei:40] Dial("SIP/ser_sec1_g711-b6601460", "SIP/ser_sei0/0143620918") in new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.7.40 port 10090 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 192.168.7.36:5060: INVITE sip:0143620918@192.168.7.36 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK51e48569;rport Max-Forwards: 70 From: "" ;tag=as2b3834c4 To: Contact: Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.14-rc1 Date: Tue, 25 Aug 2009 15:32:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer x-acro-t38: oui x-acro-sda-orig: sip:FCOIST38@sec1.acropolistelecom.net;transport=UDP x-acro-nbappel: 971 x-acro-dst_ext: t38 x-acro-codec: T38 x-acro-masquage: 1 x-acro-caller: france x-acro-called: france_fixe x-acro-credit: 0.00000 x-acro-client: Fcois x-acro-chemin: SEC-sec1|media-ares Content-Type: application/sdp Content-Length: 290 v=0 o=root 420657761 420657761 IN IP4 192.168.7.40 s=Asterisk PBX 1.6.0.14-rc1 c=IN IP4 192.168.7.40 t=0 0 m=audio 10090 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 - - - - a=ptime:20 a=sendrecv --- -- Called ser_sei0/0143620918 ares*CLI> <--- SIP read from UDP://192.168.7.36:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK51e48569;rport=5060 From: "" ;tag=as2b3834c4 To: Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 INVITE Server: Kamailio (1.4.3-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- ares*CLI> <--- SIP read from UDP://192.168.7.36:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK51e48569;rport=5060 Record-Route: From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 INVITE Contact: Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/ser_sei0-08255178 is ringing <--- Transmitting (NAT) to 217.64.49.49:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK329d.88cf7091.0;received=217.64.49.49 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-7ce019c1828ee11c-1---d8754z- Record-Route: From: "";tag=765e5e21 To: ;tag=as5eac4c71 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.14-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> ares*CLI> <--- SIP read from UDP://192.168.7.36:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK51e48569;rport=5060 Record-Route: From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 INVITE Contact: Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28 Content-Type: application/sdp Content-Length: 221 v=0 o=MxSIP 0 1299 IN IP4 192.168.8.40 s=SIP Call c=IN IP4 192.168.8.40 t=0 0 m=audio 4910 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=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.8.40:4910 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.8.40:4910 -- SIP/ser_sei0-08255178 is making progress passing it to SIP/ser_sec1_g711-b6601460 Audio is at 217.64.49.43 port 11572 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP ares*CLI> <--- Transmitting (NAT) to 217.64.49.49:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK329d.88cf7091.0;received=217.64.49.49 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-7ce019c1828ee11c-1---d8754z- Record-Route: From: "";tag=765e5e21 To: ;tag=as5eac4c71 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.14-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 202921524 202921524 IN IP4 217.64.49.43 s=Asterisk PBX 1.6.0.14-rc1 c=IN IP4 217.64.49.43 t=0 0 m=audio 11572 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 - - - - a=ptime:20 a=sendrecv <------------> ares*CLI> <--- SIP read from UDP://192.168.7.36:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK51e48569;rport=5060 Record-Route: From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 INVITE Contact: Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28 Content-Type: application/sdp Content-Length: 221 v=0 o=MxSIP 0 1299 IN IP4 192.168.8.40 s=SIP Call c=IN IP4 192.168.8.40 t=0 0 m=audio 4910 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=sendrecv <-------------> --- (11 headers 11 lines) --- -- SIP/ser_sei0-08255178 is ringing -- SIP/ser_sei0-08255178 is making progress passing it to SIP/ser_sec1_g711-b6601460 Reliably Transmitting (NAT) to 217.64.49.49:5060: OPTIONS sip:217.64.49.49 SIP/2.0 Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK19756aa6;rport Max-Forwards: 70 From: "asterisk" ;tag=as24ace49a To: Contact: Call-ID: 3c6c79137b795c5273287f42530f9b1d@217.64.49.43 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.14-rc1 Date: Tue, 25 Aug 2009 15:32:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 217.64.49.49:5060: OPTIONS sip:217.64.49.49 SIP/2.0 Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK0aea4f73;rport Max-Forwards: 70 From: "asterisk" ;tag=as08ab2927 To: Contact: Call-ID: 3dcf4126748247a74a5aaf3d339447ab@217.64.49.43 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.14-rc1 Date: Tue, 25 Aug 2009 15:32:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.7.36:5060: OPTIONS sip:192.168.7.36 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK4b69ddc2;rport Max-Forwards: 70 From: "asterisk" ;tag=as79bb2955 To: Contact: Call-ID: 6e5834010cc0f2fd4455516b1c5b7ff1@192.168.7.40 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.14-rc1 Date: Tue, 25 Aug 2009 15:32:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 217.64.49.49:5060: OPTIONS sip:217.64.49.49 SIP/2.0 Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK423bfc46;rport Max-Forwards: 70 From: "asterisk" ;tag=as5fb9bb25 To: Contact: Call-ID: 4af285796a266d075c8a66183acdfddf@217.64.49.43 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.14-rc1 Date: Tue, 25 Aug 2009 15:32:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- ares*CLI> <--- SIP read from UDP://192.168.7.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK4b69ddc2;rport=5060 From: "asterisk" ;tag=as79bb2955 To: ;tag=62f18ef1efcf42af43028c33e1d47f4a.fe8d Call-ID: 6e5834010cc0f2fd4455516b1c5b7ff1@192.168.7.40 CSeq: 102 OPTIONS Server: Kamailio (1.4.3-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '6e5834010cc0f2fd4455516b1c5b7ff1@192.168.7.40' Method: OPTIONS ares*CLI> <--- SIP read from UDP://217.64.49.49:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK19756aa6;rport=5060 From: "asterisk" ;tag=as24ace49a To: ;tag=5838f2f36eb9689a2709b2fd51d40ba0.c956 Call-ID: 3c6c79137b795c5273287f42530f9b1d@217.64.49.43 CSeq: 102 OPTIONS Server: Kamailio (1.4.3-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '3c6c79137b795c5273287f42530f9b1d@217.64.49.43' Method: OPTIONS ares*CLI> <--- SIP read from UDP://217.64.49.49:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK0aea4f73;rport=5060 From: "asterisk" ;tag=as08ab2927 To: ;tag=5838f2f36eb9689a2709b2fd51d40ba0.cbd3 Call-ID: 3dcf4126748247a74a5aaf3d339447ab@217.64.49.43 CSeq: 102 OPTIONS Server: Kamailio (1.4.3-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '3dcf4126748247a74a5aaf3d339447ab@217.64.49.43' Method: OPTIONS <--- SIP read from UDP://217.64.49.49:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK423bfc46;rport=5060 From: "asterisk" ;tag=as5fb9bb25 To: ;tag=5838f2f36eb9689a2709b2fd51d40ba0.c8c1 Call-ID: 4af285796a266d075c8a66183acdfddf@217.64.49.43 CSeq: 102 OPTIONS Server: Kamailio (1.4.3-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '4af285796a266d075c8a66183acdfddf@217.64.49.43' Method: OPTIONS ares*CLI> <--- SIP read from UDP://192.168.7.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK51e48569;rport=5060 Record-Route: From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 INVITE Contact: Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28 upported: replaces Content-Type: application/sdp Content-Length: 221 v=0 o=MxSIP 0 1299 IN IP4 192.168.8.40 s=SIP Call c=IN IP4 192.168.8.40 t=0 0 m=audio 4910 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=sendrecv <-------------> --- (12 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 0.0.0.0, port 5060 Transmitting (no NAT) to 0.0.0.0:5060: ACK sip:0143620918@192.168.8.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK3aad44d3;rport Route: Max-Forwards: 70 From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Contact: Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.14-rc1 Content-Length: 0 --- <--- SIP read from UDP://127.0.0.1:5060 ---> ACK sip:0143620918@192.168.8.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK3aad44d3;rport Route: Max-Forwards: 70 From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Contact: Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.14-rc1 Content-Length: 0 <-------------> -- SIP/ser_sei0-08255178 answered SIP/ser_sec1_g711-b6601460 --- (11 headers 0 lines) --- Audio is at 217.64.49.43 port 11572 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 217.64.49.49:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK329d.88cf7091.0;received=217.64.49.49 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-7ce019c1828ee11c-1---d8754z- Record-Route: From: "";tag=765e5e21 To: ;tag=as5eac4c71 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.14-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 202921524 202921525 IN IP4 217.64.49.43 s=Asterisk PBX 1.6.0.14-rc1 c=IN IP4 217.64.49.43 t=0 0 m=audio 11572 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 - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/ser_sec1_g711-b6601460 and SIP/ser_sei0-08255178 ares*CLI> <--- SIP read from UDP://217.64.49.49:5060 ---> ACK sip:0143620918@217.64.49.43 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK329d.88cf7091.2 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-11ca34d7a04e8890-1---d8754z- Max-Forwards: 70 Route: Contact: To: ;tag=as5eac4c71 From: "";tag=765e5e21 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 1 ACK User-Agent: Zoiper rev.4829 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- ares*CLI> <--- SIP read from UDP://217.64.49.49:5060 ---> INVITE sip:0143620918@217.64.49.43 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK029d.3d2dd003.0 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-fb7a46d597a737d1-1---d8754z- Max-Forwards: 70 Route: Contact: To: ;tag=as5eac4c71 From: "";tag=765e5e21 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.4829 Content-Length: 375 x-acro-client: Fcois x-acro-chemin: SEC-sec1 x-acro-nbappel: 972 x-acro-called: france_fixe x-acro-dst_ext: t38 x-acro-tarifplan: 1 x-acro-factuprefix: 0143 x-acro-factumin: 0.002 x-acro-credit: 0.00000 x-acro-sda-orig: sip:FCOIST38@sec1.acropolistelecom.net;transport=UDP x-acro-caller: france x-acro-caller: int x-acro-masquage: 1 x-acro-codec: T38 x-acro-t38: oui x-acro-dst_ext: t38 v=0 o=Zoiper_user 1228341416 261835895 IN IP4 192.168.1.100 s=Zoiper_session c=IN IP4 192.168.1.100 t=0 0 m=image 8000 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (31 headers 15 lines) --- Sending to 217.64.49.49 : 5060 (NAT) [Aug 25 17:32:30] WARNING[5198]: chan_sip.c:7025 process_sdp: Unsupported SDP media type in offer: image 8000 udptl t38 <--- Reliably Transmitting (NAT) to 217.64.49.49:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK029d.3d2dd003.0;received=217.64.49.49 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-fb7a46d597a737d1-1---d8754z- From: "";tag=765e5e21 To: ;tag=as5eac4c71 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.14-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> ares*CLI> <--- SIP read from UDP://217.64.49.49:5060 ---> ACK sip:0143620918@217.64.49.43 SIP/2.0 Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK029d.3d2dd003.0 From: "";tag=765e5e21 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. To: ;tag=as5eac4c71 CSeq: 2 ACK Max-Forwards: 70 Route: User-Agent: Kamailio (1.4.3-notls (i386/linux)) Content-Length: 0 ares*CLI> <-------------> --- (10 headers 0 lines) --- ares*CLI> <--- SIP read from UDP://217.64.49.49:5060 ---> BYE sip:0143620918@217.64.49.43 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK129d.7934f5c7.0 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-8e0ae9721c9f68d1-1---d8754z- Max-Forwards: 70 Route: Contact: To: ;tag=as5eac4c71 From: "";tag=765e5e21 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 3 BYE User-Agent: Zoiper rev.4829 Content-Length: 0 x-acro-dst_ext: <-------------> --- (14 headers 0 lines) --- Sending to 217.64.49.49 : 5060 (NAT) ares*CLI> <--- Transmitting (NAT) to 217.64.49.49:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.64.49.49;branch=z9hG4bK129d.7934f5c7.0;received=217.64.49.49 Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-8e0ae9721c9f68d1-1---d8754z- Record-Route: From: "";tag=765e5e21 To: ;tag=as5eac4c71 Call-ID: MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk. CSeq: 3 BYE User-Agent: Asterisk PBX 1.6.0.14-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (macro-envoi_sei, h, 5) exited non-zero on 'SIP/ser_sec1_g711-b6601460' Scheduling destruction of SIP dialog '3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 0.0.0.0, port 5060 Reliably Transmitting (no NAT) to 0.0.0.0:5060: BYE sip:0143620918@192.168.8.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK64374140;rport Route: Max-Forwards: 70 From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.0.14-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- ares*CLI> <--- SIP read from UDP://127.0.0.1:5060 ---> BYE sip:0143620918@192.168.8.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK64374140;rport Route: Max-Forwards: 70 From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.0.14-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.7.40 : 5060 (no NAT) Scheduling destruction of SIP dialog '3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40' in 6400 ms (Method: BYE) ares*CLI> <--- Transmitting (no NAT) to 192.168.7.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK64374140;received=127.0.0.1;rport=5060 From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.0.14-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> ares*CLI> <--- SIP read from UDP://192.168.7.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK64374140;received=127.0.0.1;rport=5060 From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.0.14-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- == Spawn extension (macro-envoi_sei, s, 40) exited non-zero on 'SIP/ser_sec1_g711-b6601460' in macro 'envoi_sei' == Spawn extension (ser_sec, 0143620918, 9) exited non-zero on 'SIP/ser_sec1_g711-b6601460' ares*CLI> <--- SIP read from UDP://192.168.7.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK51e48569;rport=5060 Record-Route: From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 INVITE Contact: Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28 upported: replaces Content-Type: application/sdp Content-Length: 221 v=0 o=MxSIP 0 1299 IN IP4 192.168.8.40 s=SIP Call c=IN IP4 192.168.8.40 t=0 0 m=audio 4910 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=sendrecv <-------------> --- (12 headers 11 lines) --- Really destroying SIP dialog 'MWQ4NTRiNzc3ZDUwZWEyNTE0MTc4MTlhZWI2ZTllYTk.' Method: BYE ares*CLI> <--- SIP read from UDP://192.168.7.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK51e48569;rport=5060 Record-Route: From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 INVITE Contact: Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28 upported: replaces Content-Type: application/sdp Content-Length: 221 v=0 o=MxSIP 0 1299 IN IP4 192.168.8.40 s=SIP Call c=IN IP4 192.168.8.40 t=0 0 m=audio 4910 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=sendrecv <-------------> --- (12 headers 11 lines) --- ares*CLI> <--- SIP read from UDP://192.168.7.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK51e48569;rport=5060 Record-Route: From: "" ;tag=as2b3834c4 To: ;tag=1841599123 Call-ID: 3a6e12665b4dc8e70cb1cd2c17141857@192.168.7.40 CSeq: 102 INVITE Contact: Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28 upported: replaces Content-Type: application/sdp Content-Length: 221 v=0 o=MxSIP 0 1299 IN IP4 192.168.8.40 s=SIP Call c=IN IP4 192.168.8.40 t=0 0 m=audio 4910 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=sendrecv <-------------> --- (12 headers 11 lines) --- ares*CLI> Disconnected from Asterisk server