-- Attempting call on Local/4483706@client/n for s@faxtx:1 (Retry 1) -- Executing [4483706@client:1] NoOp("Local/4483706@client-a676;2", ""CLID = 3364079"") in new stack -- Executing [4483706@client:2] Dial("Local/4483706@client-a676;2", "SIP/kamailio/4483706") in new stack == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 213.170.75.90 port 13408 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.170.75.90:5050: INVITE sip:4483706@213.170.75.90:5050 SIP/2.0 Via: SIP/2.0/UDP 213.170.75.90:5060;branch=z9hG4bK433c1781;rport Max-Forwards: 70 From: "3364079" ;tag=as45609cd1 To: Contact: Call-ID: 3644ee1e1d2c8eea5d9b820141700560@213.170.75.90 CSeq: 102 INVITE User-Agent: GW Date: Wed, 03 Mar 2010 10:38:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 244249172 244249172 IN IP4 213.170.75.90 s=Asterisk PBX 1.6.2.4 c=IN IP4 213.170.75.90 t=0 0 m=audio 13408 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- ubuntutest*CLI> <--- SIP read from UDP:213.170.75.90:5050 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 213.170.75.90:5060;branch=z9hG4bK433c1781;rport=5060 From: "3364079" ;tag=as45609cd1 To: Call-ID: 3644ee1e1d2c8eea5d9b820141700560@213.170.75.90 CSeq: 102 INVITE Server: OpenSIPS (1.6.1-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- ubuntutest*CLI> -- Called kamailio/4483706 ubuntutest*CLI> <--- SIP read from UDP:213.170.75.90:5050 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 213.170.75.90:5060;received=213.170.75.90;branch=z9hG4bK433c1781;rport=5060 From: "3364079" ;tag=as45609cd1 To: ;tag=EBA15504-14F5 Date: Thu, 07 Dec 2000 20:17:33 GMT Call-ID: 3644ee1e1d2c8eea5d9b820141700560@213.170.75.90 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Contact: R ubuntutest*CLI> ecord-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 249 v=0 o=CiscoSystemsSIP-GW-UserAgent 788 2153 IN IP4 213.170.74.62 s=SIP Call c=IN IP4 213.170.74.62 t=0 0 m=audio 16838 RTP/AVP 8 101 c=IN IP4 213.170.74.62 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.170.74.62:16838 -- SIP/kamailio-00000000 is making progress passing it to Local/4483706@client-a676;2 ubuntutest*CLI> <--- SIP read from UDP:213.170.75.90:5050 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.170.75.90:5060;received=213.170.75.90;branch=z9hG4bK433c1781;rport=5060 From: "3364079" ;tag=as45609cd1 To: ;tag=EBA15504-14F5 Date: Thu, 07 Dec 2000 20:17:33 GMT Call-ID: 3644ee1e1d2c8eea5d9b820141700560@213.170.75.90 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO A ubuntutest*CLI> llow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 249 v=0 o=CiscoSystemsSIP-GW-UserAgent 788 2153 IN IP4 213.170.74.62 s=SIP Call c=IN IP4 213.170.74.62 t=0 0 m=audio 16838 RTP/AVP 8 101 c=IN IP4 213.170.74.62 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.170.75.90, port 5050 Transmitting (no NAT) to 213.170.75.90:5050: ACK sip:4483706@213.170.74.62:5060 SIP/2.0 Via: SIP/2.0/UDP 213.170.75.90:5060;branch=z9hG4bK4cb24680;rport Route: Max-Forwards: 70 From: "3364079" ;tag=as45609cd1 To: ;tag=EBA15504-14F5 Contact: Call-ID: 3644ee1e1d2c8eea5d9b820141700560@213.170.75.90 CSeq: 102 ACK User-Agent: GW Content-Length: 0 --- -- SIP/kamailio-00000000 answered Local/4483706@client-a676;2 ubuntutest*CLI> > Channel Local/4483706@client-a676;1 was answered. -- Executing [s@faxtx:1] Wait("Local/4483706@client-a676;1", "3") in new stack ubuntutest*CLI> -- Executing [s@faxtx:2] SendFAX("Local/4483706@client-a676;1", "/home/den/faxdemand.tif") in new stack ubuntutest*CLI> <--- SIP read from UDP:213.170.75.90:5050 ---> INVITE sip:3364079@213.170.75.90:5060 SIP/2.0 Via: SIP/2.0/UDP 213.170.75.90:5050;branch=z9hG4bKda26.7d026463.0 Via: SIP/2.0/UDP 213.170.74.62:5060 From: ;tag=EBA15504-14F5 To: "3364079" ;tag=as45609cd1 Date: Thu, 07 Dec 2000 20:17:45 GMT Call-ID: 3644ee1e1d2c8eea5d9b820141700560@213.170.75.90 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 3474936558-3417575892-2771689371-3892023434 User-Agent: Cisco-SIPGateway/IOS-12.x Accept-Language: ru Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: ;party=calling;screen=no;privacy=full Timestamp: 976220265 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 398 v=0 o=CiscoSystemsSIP-GW-UserAgent 788 2154 IN IP4 213.170.74.62 s=SIP Call c=IN IP4 213.170.74.62 t=0 0 m=image 16838 udptl t38 c=IN IP4 213.170.74.62 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (22 headers 16 lines) --- Sending to 213.170.75.90 : 5050 (no NAT) Got T.38 offer in SDP in dialog 3644ee1e1d2c8eea5d9b820141700560@213.170.75.90 Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. <--- Transmitting (no NAT) to 213.170.75.90:5050 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.170.75.90:5050;branch=z9hG4bKda26.7d026463.0;received=213.170.75.90 Via: SIP/2.0/UDP 213.170.74.62:5060 From: ;tag=EBA15504-14F5 To: "3364079" ;tag=as45609cd1 Call-ID: 3644ee1e1d2c8eea5d9b820141700560@213.170.75.90 CSeq: 101 INVITE Server: GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> ubuntutest*CLI> <--- Reliably Transmitting (no NAT) to 213.170.75.90:5050 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.170.75.90:5050;branch=z9hG4bKda26.7d026463.0;received=213.170.75.90 Via: SIP/2.0/UDP 213.170.74.62:5060 From: ;tag=EBA15504-14F5 To: "3364079" ;tag=as45609cd1 Call-ID: 3644ee1e1d2c8eea5d9b820141700560@213.170.75.90 CSeq: 101 INVITE Server: GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 272 v=0 o=root 244249172 244249173 IN IP4 213.170.75.90 s=Asterisk PBX 1.6.2.4 c=IN IP4 213.170.75.90 t=0 0 m=image 4940 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <------------> ubuntutest*CLI> Disconnected from Asterisk server root@ubuntutest:~#