<--- SIP read from UDP:11.22.33.44:5060 ---> INVITE sip:+4412345@11.22.33.44;transport=UDP SIP/2.0 Record-Route: Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.0 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-cf7d037085b5fe20-1---d8754z- Max-Forwards: 69 Contact: To: From: "+4466666";tag=7675a242 Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri Allow-Events: presence, kpml Content-Length: 300 v=0 o=Zoiper_user 0 0 IN IP4 192.168.5.1 s=Zoiper_session c=IN IP4 44.55.66.77 t=0 0 m=audio 8000 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=direction:active a=oldmediaip:192.168.5.1 <-------------> --- (28 headers 14 lines) --- Sending to 11.22.33.44:5060 (no NAT) Using INVITE request as basis request - YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. Found peer 'in.from.a' for 'in.from.a' from 11.22.33.44:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x108 (alaw|g729)/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 44.55.66.77:8000 Looking for +4412345 in in.from.in.from.a (domain 11.22.33.44) list_route: hop: <--- Transmitting (NAT) to 11.22.33.44:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.0;received=11.22.33.44;rport=5060 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-cf7d037085b5fe20-1---d8754z- Record-Route: From: "+4466666";tag=7675a242 To: Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 2 INVITE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [+4412345@in.from.in.from.a:1] NoOp("SIP/in.from.a-00000086", "in.from.in.from.a") in new stack -- Executing [+4412345@in.from.in.from.a:2] Dial("SIP/in.from.a-00000086", "SIP/+4412345@out.to.b,90,ieL(10800000)") in new stack -- Setting call duration limit to 10800.000 seconds. == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 17550 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 33.55.77.99:5060: INVITE sip:+4412345@33.55.77.99 SIP/2.0 Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK7964879d Max-Forwards: 70 From: "+4466666" ;tag=as5160c622 To: Contact: Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 102 INVITE User-Agent: Asteriskserver Date: Mon, 24 Jun 2013 21:43:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: Content-Type: application/sdp Content-Length: 251 v=0 o=root 1707261853 1707261853 IN IP4 88.99.11.22 s=Asteriskserver SDP c=IN IP4 88.99.11.22 t=0 0 m=audio 17550 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 --- -- Called SIP/+4412345@out.to.b <--- SIP read from UDP:33.55.77.99:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK7964879d From: "+4466666" ;tag=as5160c622 To: Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 102 INVITE Server: Asteriskserver Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:33.55.77.99:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK7964879d From: "+4466666" ;tag=as5160c622 To: ;tag=CALL.138.211 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 102 INVITE Content-Type: application/sdp Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE Contact: Server: Asteriskserver Content-Length: 208 v=0 o=- 3581099020 3581099020 IN IP4 22.44.66.88 s=- c=IN IP4 22.44.66.88 t=0 0 m=audio 22122 RTP/AVP 8 101 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 10 lines) --- list_route: hop: 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 22.44.66.88:22122 -- SIP/out.to.b-00000087 is making progress passing it to SIP/in.from.a-00000086 Audio is at 21172 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 11.22.33.44:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.0;received=11.22.33.44;rport=5060 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-cf7d037085b5fe20-1---d8754z- Record-Route: From: "+4466666";tag=7675a242 To: ;tag=as02cf8d7c Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 2 INVITE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 251 v=0 o=root 1894142497 1894142497 IN IP4 88.99.11.22 s=Asteriskserver SDP c=IN IP4 88.99.11.22 t=0 0 m=audio 21172 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 <------------> <--- SIP read from UDP:33.55.77.99:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK7964879d From: "+4466666" ;tag=as5160c622 To: ;tag=CALL.138.211 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 102 INVITE Content-Type: application/sdp Supported: replaces, 100rel, timer Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE Session-Expires: 1800;refresher=uas Require: timer Contact: Server: Asteriskserver Content-Length: 208 v=0 o=- 3581099020 3581099020 IN IP4 22.44.66.88 s=- c=IN IP4 22.44.66.88 t=0 0 m=audio 22122 RTP/AVP 8 101 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 10 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 33.55.77.99:5060 Transmitting (no NAT) to 33.55.77.99:5060: ACK sip:33.55.77.99:5060 SIP/2.0 Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK2a7104f2 Max-Forwards: 70 From: "+4466666" ;tag=as5160c622 To: ;tag=CALL.138.211 Contact: Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 102 ACK User-Agent: Asteriskserver Content-Length: 0 --- -- SIP/out.to.b-00000087 answered SIP/in.from.a-00000086 Audio is at 21172 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 11.22.33.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.0;received=11.22.33.44;rport=5060 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-cf7d037085b5fe20-1---d8754z- Record-Route: From: "+4466666";tag=7675a242 To: ;tag=as02cf8d7c Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 2 INVITE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 251 v=0 o=root 1894142497 1894142498 IN IP4 88.99.11.22 s=Asteriskserver SDP c=IN IP4 88.99.11.22 t=0 0 m=audio 21172 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 <------------> -- Locally bridging SIP/in.from.a-00000086 and SIP/out.to.b-00000087 <--- SIP read from UDP:11.22.33.44:5060 ---> ACK sip:+4412345@88.99.11.22:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.2 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-872e94aaac46ca38-1---d8754z- Max-Forwards: 69 Contact: To: ;tag=as02cf8d7c From: "+4466666";tag=7675a242 Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 2 ACK Proxy-Authorization: Digest username="testphone",realm="11.22.33.44",nonce="51c8bda900011d9d670190766d75d7a751c7d9f0f4fd3393",uri="sip:+4412345@11.22.33.44;transport=UDP",response="eb6e5f32d4004e604a80be8c17349e5e",algorithm=MD5 User-Agent: Zoiper for Windows 2.39 r16838 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- SIP read from UDP:11.22.33.44:5060 ---> INVITE sip:+4412345@88.99.11.22:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKc637.56117dc7.0 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-bcd7fdc83d9ec2c1-1---d8754z- Max-Forwards: 69 Contact: To: ;tag=as02cf8d7c From: "+4466666";tag=7675a242 Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 3 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="testphone",realm="11.22.33.44",nonce="51c8bda900011d9d670190766d75d7a751c7d9f0f4fd3393",uri="sip:+4412345@88.99.11.22:5060",response="c40a13eea34f49fad7cd778b44df4180",algorithm=MD5 Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper for Windows 2.39 r16838 Allow-Events: presence, kpml Content-Length: 403 v=0 o=Zoiper_user 0 1 IN IP4 192.168.5.1 s=Zoiper_session c=IN IP4 44.55.66.77 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 a=direction:active a=oldmediaip:192.168.5.1 <-------------> --- (17 headers 17 lines) --- Sending to 11.22.33.44:5060 (NAT) == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 Got T.38 offer in SDP in dialog YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. 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 (NAT) to 11.22.33.44:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKc637.56117dc7.0;received=11.22.33.44;rport=5060 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-bcd7fdc83d9ec2c1-1---d8754z- Record-Route: From: "+4466666";tag=7675a242 To: ;tag=as02cf8d7c Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 3 INVITE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 set_destination: Parsing for address/port to send to set_destination: set destination to 33.55.77.99:5060 Reliably Transmitting (no NAT) to 33.55.77.99:5060: INVITE sip:33.55.77.99:5060 SIP/2.0 Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK2ea08cac Max-Forwards: 70 From: "+4466666" ;tag=as5160c622 To: ;tag=CALL.138.211 Contact: Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 103 INVITE User-Agent: Asteriskserver Session-Expires: 1800;refresher=uas Min-SE: 90 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: 253 v=0 o=root 1707261853 1707261854 IN IP4 88.99.11.22 s=Asteriskserver SDP c=IN IP4 88.99.11.22 t=0 0 m=image 42859 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:204 a=T38FaxUdpEC:t38UDPFEC --- <--- SIP read from UDP:33.55.77.99:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK2ea08cac From: "+4466666" ;tag=as5160c622 To: ;tag=CALL.138.211 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 103 INVITE Server: Asteriskserver Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:33.55.77.99:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK2ea08cac From: "+4466666" ;tag=as5160c622 To: ;tag=CALL.138.211 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 103 INVITE Content-Type: application/sdp Supported: replaces, 100rel, timer Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE Session-Expires: 1800;refresher=uas Require: timer Contact: Server: Asteriskserver Content-Length: 278 v=0 o=- 3581099020 3581099021 IN IP4 22.44.66.88 s=- c=IN IP4 22.44.66.88 t=0 0 m=image 22122 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:316 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (14 headers 12 lines) --- Got T.38 offer in SDP in dialog 433e148975141e082e378e774d3f532c@88.99.11.22:5060 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. set_destination: Parsing for address/port to send to set_destination: set destination to 33.55.77.99:5060 Transmitting (no NAT) to 33.55.77.99:5060: ACK sip:33.55.77.99:5060 SIP/2.0 Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK56d574cc Max-Forwards: 70 From: "+4466666" ;tag=as5160c622 To: ;tag=CALL.138.211 Contact: Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 103 ACK User-Agent: Asteriskserver Content-Length: 0 --- <--- Reliably Transmitting (NAT) to 11.22.33.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKc637.56117dc7.0;received=11.22.33.44;rport=5060 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-bcd7fdc83d9ec2c1-1---d8754z- Record-Route: From: "+4466666";tag=7675a242 To: ;tag=as02cf8d7c Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 3 INVITE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1894142497 1894142499 IN IP4 88.99.11.22 s=Asteriskserver SDP c=IN IP4 88.99.11.22 t=0 0 m=image 57681 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:410 a=T38FaxUdpEC:t38UDPRedundancy <------------> <--- SIP read from UDP:11.22.33.44:5060 ---> ACK sip:+4412345@88.99.11.22:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKc637.56117dc7.2 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-90dbf8b021152d84-1---d8754z- Max-Forwards: 69 Contact: To: ;tag=as02cf8d7c From: "+4466666";tag=7675a242 Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 3 ACK Proxy-Authorization: Digest username="testphone",realm="11.22.33.44",nonce="51c8bda900011d9d670190766d75d7a751c7d9f0f4fd3393",uri="sip:+4412345@88.99.11.22:5060",response="c40a13eea34f49fad7cd778b44df4180",algorithm=MD5 User-Agent: Zoiper for Windows 2.39 r16838 Content-Length: 0 <-------------> <--- SIP read from UDP:33.55.77.99:5060 ---> INVITE sip:+4466666@88.99.11.22:5060 SIP/2.0 Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.1f394343.0 To: "+4466666" ;tag=as5160c622 From: ;tag=CALL.138.211 CSeq: 2 INVITE Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 Max-Forwards: 70 Content-Length: 278 User-Agent: Asteriskserver Content-Type: application/sdp Supported: replaces, 100rel, timer Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE Session-Expires: 1800;refresher=uac Min-SE: 90 Contact: v=0 o=- 3581099020 3581099022 IN IP4 22.44.66.88 s=- c=IN IP4 22.44.66.88 t=0 0 m=image 22122 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:316 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (15 headers 12 lines) --- Sending to 33.55.77.99:5060 (no NAT) Got T.38 offer in SDP in dialog 433e148975141e082e378e774d3f532c@88.99.11.22:5060 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 33.55.77.99:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.1f394343.0;received=33.55.77.99 From: ;tag=CALL.138.211 To: "+4466666" ;tag=as5160c622 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 2 INVITE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Length: 0 <------------> <--- Reliably Transmitting (no NAT) to 33.55.77.99:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.1f394343.0;received=33.55.77.99 From: ;tag=CALL.138.211 To: "+4466666" ;tag=as5160c622 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 2 INVITE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uac X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <------------> <--- SIP read from UDP:33.55.77.99:5060 ---> ACK sip:+4466666@88.99.11.22:5060 SIP/2.0 Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.1f394343.0 From: ;tag=CALL.138.211 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 To: "+4466666" ;tag=as5160c622 CSeq: 2 ACK Max-Forwards: 70 User-Agent: Asteriskserver Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:33.55.77.99:5060 ---> ACK sip:+4466666@88.99.11.22:5060 SIP/2.0 Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.2f394343.0 To: "+4466666" ;tag=as5160c622 From: ;tag=CALL.138.211 CSeq: 2 ACK Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 Max-Forwards: 70 Content-Length: 0 User-Agent: Asteriskserver Contact: <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:33.55.77.99:5060 ---> INVITE sip:+4466666@88.99.11.22:5060 SIP/2.0 Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK9074.8221f213.0 To: "+4466666" ;tag=as5160c622 From: ;tag=CALL.138.211 CSeq: 3 INVITE Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 Max-Forwards: 70 Content-Length: 164 User-Agent: Asteriskserver Content-Type: application/sdp Supported: replaces, 100rel, timer Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE Session-Expires: 1800;refresher=uac Min-SE: 90 Contact: v=0 o=- 3581099020 3581099023 IN IP4 22.44.66.88 s=- c=IN IP4 22.44.66.88 t=0 0 m=audio 22122 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sendrecv a=ptime:20 <-------------> --- (15 headers 9 lines) --- Sending to 33.55.77.99:5060 (no NAT) Found RTP audio format 8 Found audio description format PCMA for ID 8 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 - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 22.44.66.88:22122 <--- Transmitting (no NAT) to 33.55.77.99:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK9074.8221f213.0;received=33.55.77.99 From: ;tag=CALL.138.211 To: "+4466666" ;tag=as5160c622 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 3 INVITE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Length: 0 <------------> Audio is at 17550 Adding codec 0x8 (alaw) to SDP <--- Reliably Transmitting (no NAT) to 33.55.77.99:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK9074.8221f213.0;received=33.55.77.99 From: ;tag=CALL.138.211 To: "+4466666" ;tag=as5160c622 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 3 INVITE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Type: application/sdp Require: timer Content-Length: 195 v=0 o=root 1707261853 1707261855 IN IP4 88.99.11.22 s=Asteriskserver SDP c=IN IP4 88.99.11.22 t=0 0 m=audio 17550 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:33.55.77.99:5060 ---> ACK sip:+4466666@88.99.11.22:5060 SIP/2.0 Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK9074.9221f213.0 To: "+4466666" ;tag=as5160c622 From: ;tag=CALL.138.211 CSeq: 3 ACK Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 Max-Forwards: 70 Content-Length: 0 User-Agent: Asteriskserver Contact: <-------------> <--- SIP read from UDP:11.22.33.44:5060 ---> BYE sip:+4412345@88.99.11.22:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK9637.218d4292.0 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-ceb627e82ff8607e-1---d8754z- Max-Forwards: 69 Contact: To: ;tag=as02cf8d7c From: "+4466666";tag=7675a242 Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 4 BYE Proxy-Authorization: Digest username="testphone",realm="11.22.33.44",nonce="51c8bda900011d9d670190766d75d7a751c7d9f0f4fd3393",uri="sip:+4412345@88.99.11.22:5060",response="6fdd6777c7d07c453b47cb57e88ce3ff",algorithm=MD5 User-Agent: Zoiper for Windows 2.39 r16838 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 11.22.33.44:5060 (NAT) Scheduling destruction of SIP dialog 'YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 11.22.33.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK9637.218d4292.0;received=11.22.33.44;rport=5060 Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-ceb627e82ff8607e-1---d8754z- Record-Route: From: "+4466666";tag=7675a242 To: ;tag=as02cf8d7c Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU. CSeq: 4 BYE Server: Asteriskserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@in.from.in.from.a:1] NoOp("SIP/in.from.a-00000086", "bye") in new stack == Spawn extension (in.from.in.from.a, h, 1) exited non-zero on 'SIP/out.to.b-00000087' Scheduling destruction of SIP dialog '433e148975141e082e378e774d3f532c@88.99.11.22:5060' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 33.55.77.99:5060 Reliably Transmitting (no NAT) to 33.55.77.99:5060: BYE sip:33.55.77.99:5060 SIP/2.0 Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK55961fa5 Max-Forwards: 70 From: "+4466666" ;tag=as5160c622 To: ;tag=CALL.138.211 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 104 BYE User-Agent: Asteriskserver X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (in.from.in.from.a, +4412345, 2) exited non-zero on 'SIP/in.from.a-00000086' <--- SIP read from UDP:33.55.77.99:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK55961fa5 From: "+4466666" ;tag=as5160c622 To: ;tag=CALL.138.211 Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060 CSeq: 104 BYE Supported: replaces, 100rel, timer Contact: Server: Asteriskserver Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '433e148975141e082e378e774d3f532c@88.99.11.22:5060' Method: ACK