Connected to Asterisk SVN-trunk-r341024M currently running on lx3 (pid = 15632) Verbosity is at least 1 Audio is at 12384 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100011 (g726) to SDP Adding codec 100009 (speex) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.7.65:2048: INVITE sip:wimpys@192.168.7.65:2048;line=tnlq97gr SIP/2.0 v: SIP/2.0/UDP 192.168.7.23:5060;branch=z9hG4bK71a47efb Max-Forwards: 70 f: ;tag=as2d8928f1 t: m: i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 102 INVITE User-Agent: Asterisk Date: Tue, 18 Oct 2011 21:30:32 GMT x: 600 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer Remote-Party-ID: "+4946123240" ;party=calling;privacy=off;screen=yes c: application/sdp l: 384 v=0 o=nobody 2139947814 2139947814 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/AVP 8 9 0 3 111 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv --- <--- SIP read from UDP:192.168.7.65:2048 ---> SIP/2.0 180 Ringing v: SIP/2.0/UDP 192.168.7.23:5060;branch=z9hG4bK71a47efb f: ;tag=as2d8928f1 t: "WIMPy" ;tag=70x4olau35 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 102 INVITE m: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info l: 0 <-------------> --- (10 headers 0 lines) --- list_route: hop: <--- SIP read from UDP:192.168.7.65:2048 ---> SIP/2.0 180 Ringing v: SIP/2.0/UDP 192.168.7.23:5060;branch=z9hG4bK71a47efb f: ;tag=as2d8928f1 t: "WIMPy" ;tag=70x4olau35 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 102 INVITE m: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info l: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> SIP/2.0 180 Ringing v: SIP/2.0/UDP 192.168.7.23:5060;branch=z9hG4bK71a47efb f: ;tag=as2d8928f1 t: "WIMPy" ;tag=70x4olau35 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 102 INVITE m: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info l: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.7.23:5060;branch=z9hG4bK71a47efb f: ;tag=as2d8928f1 t: "WIMPy" ;tag=70x4olau35 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 102 INVITE m: ;reg-id=1 Require: timer Session-Expires: 600;refresher=uac User-Agent: snom360/8.4.27 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 c: application/sdp l: 284 v=0 o=root 361817363 361817364 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/AVP 8 9 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=direction:both a=sendrecv <-------------> --- (15 headers 14 lines) --- Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g726|speex|g722), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.7.65:65426 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.7.65:2048 Transmitting (no NAT) to 192.168.7.65:2048: ACK sip:wimpys@192.168.7.65:2048;line=tnlq97gr SIP/2.0 v: SIP/2.0/UDP 192.168.7.23:5060;branch=z9hG4bK3b50d6c1 Max-Forwards: 70 f: ;tag=as2d8928f1 t: ;tag=70x4olau35 m: i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 102 ACK User-Agent: Asterisk l: 0 --- <--- SIP read from UDP:192.168.7.65:2048 ---> INVITE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-b0awmydkfwkh;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 1 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 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 Require: timer Session-Expires: 600;refresher=uas Min-SE: 90 c: application/sdp l: 724 v=0 o=root 361817363 361817365 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/SAVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendonly m=audio 65426 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendonly <-------------> --- (20 headers 33 lines) --- Sending to 192.168.7.65:2048 (no NAT) Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 WARNING[15689]: chan_sip.c:9035 process_sdp: Can't provide secure audio requested in SDP offer <--- Reliably Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 488 Not acceptable here v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-b0awmydkfwkh;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 1 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uac X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 l: 0 <------------> Retransmitting #1 (no NAT) to 192.168.7.65:2048: SIP/2.0 488 Not acceptable here v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-b0awmydkfwkh;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 1 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uac X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 l: 0 --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-b0awmydkfwkh;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 1 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-b0awmydkfwkh;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 1 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> INVITE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-uyvb561ic09d;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 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 Require: timer Session-Expires: 600;refresher=uas Min-SE: 90 c: application/sdp l: 808 v=0 o=root 361817363 361817366 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/SAVP 9 8 0 3 18 99 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:yqBtWhzluhxOMG7S7qnlj/s6barDh4Ubm6XTncn+ a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv m=audio 65426 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv <-------------> --- (20 headers 34 lines) --- Sending to 192.168.7.65:2048 (no NAT) Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g726|speex|g722), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g726|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.7.65:65426 <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-uyvb561ic09d;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: l: 0 <------------> Audio is at 12384 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100011 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-uyvb561ic09d;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947815 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r <------------> Retransmitting #1 (no NAT) to 192.168.7.65:2048: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-uyvb561ic09d;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947815 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r --- <--- SIP read from UDP:192.168.7.65:2048 ---> INVITE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-uyvb561ic09d;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 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 Require: timer Session-Expires: 600;refresher=uas Min-SE: 90 c: application/sdp l: 808 v=0 o=root 361817363 361817366 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/SAVP 9 8 0 3 18 99 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:yqBtWhzluhxOMG7S7qnlj/s6barDh4Ubm6XTncn+ a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv m=audio 65426 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv <-------------> --- (20 headers 34 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-uyvb561ic09d;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: l: 0 <------------> Audio is at 12384 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100011 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-uyvb561ic09d;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947816 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r <------------> Retransmitting #2 (no NAT) to 192.168.7.65:2048: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-uyvb561ic09d;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947815 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-jpu6in1d7b9t;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-jpu6in1d7b9t;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-a7xpn2qh0lc6;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-jpu6in1d7b9t;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 2 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> INVITE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-renonaaebbj7;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 3 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 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 Require: timer Session-Expires: 600;refresher=uas Min-SE: 90 c: application/sdp l: 724 v=0 o=root 361817363 361817367 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/SAVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendonly m=audio 65426 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendonly <-------------> --- (20 headers 33 lines) --- Sending to 192.168.7.65:2048 (no NAT) Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g726|speex|g722), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g726|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.7.65:65426 <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-renonaaebbj7;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 3 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: l: 0 <------------> Audio is at 12384 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100011 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-renonaaebbj7;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 3 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947817 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=recvonly a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r <------------> Retransmitting #1 (no NAT) to 192.168.7.65:2048: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-renonaaebbj7;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 3 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947817 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=recvonly a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-1v5927rn8y50;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 3 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-1v5927rn8y50;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 3 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> INVITE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-sbtfpivgw3o4;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 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 Require: timer Session-Expires: 600;refresher=uas Min-SE: 90 c: application/sdp l: 808 v=0 o=root 361817363 361817368 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/SAVP 9 8 0 3 18 99 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:yqBtWhzluhxOMG7S7qnlj/s6barDh4Ubm6XTncn+ a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv m=audio 65426 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv <-------------> --- (20 headers 34 lines) --- Sending to 192.168.7.65:2048 (no NAT) Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g726|speex|g722), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g726|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.7.65:65426 <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-sbtfpivgw3o4;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: l: 0 <------------> Audio is at 12384 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100011 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-sbtfpivgw3o4;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947818 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r <------------> Retransmitting #1 (no NAT) to 192.168.7.65:2048: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-sbtfpivgw3o4;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947818 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r --- <--- SIP read from UDP:192.168.7.65:2048 ---> INVITE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-sbtfpivgw3o4;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 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 Require: timer Session-Expires: 600;refresher=uas Min-SE: 90 c: application/sdp l: 808 v=0 o=root 361817363 361817368 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/SAVP 9 8 0 3 18 99 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:yqBtWhzluhxOMG7S7qnlj/s6barDh4Ubm6XTncn+ a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv m=audio 65426 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv <-------------> --- (20 headers 34 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-sbtfpivgw3o4;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: l: 0 <------------> Audio is at 12384 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100011 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-sbtfpivgw3o4;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947819 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r <------------> Retransmitting #2 (no NAT) to 192.168.7.65:2048: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-sbtfpivgw3o4;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947818 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-xj2tul0v0p43;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-xj2tul0v0p43;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-gwapef96bn5k;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-xj2tul0v0p43;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 4 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> INVITE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-k9luijuba5yz;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 5 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 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 Require: timer Session-Expires: 600;refresher=uas Min-SE: 90 c: application/sdp l: 724 v=0 o=root 361817363 361817369 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/SAVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendonly m=audio 65426 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendonly <-------------> --- (20 headers 33 lines) --- Sending to 192.168.7.65:2048 (no NAT) Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g726|speex|g722), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g726|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.7.65:65426 <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-k9luijuba5yz;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 5 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: l: 0 <------------> Audio is at 12384 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100011 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-k9luijuba5yz;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 5 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947820 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=recvonly a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r <------------> Retransmitting #1 (no NAT) to 192.168.7.65:2048: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-k9luijuba5yz;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 5 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947820 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=recvonly a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-pnkttsl2ixe0;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 5 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-pnkttsl2ixe0;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 5 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> <-------------> <--- SIP read from UDP:192.168.7.65:2048 ---> INVITE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-2v3ncwcxphr7;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 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 Require: timer Session-Expires: 600;refresher=uas Min-SE: 90 c: application/sdp l: 808 v=0 o=root 361817363 361817370 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/SAVP 9 8 0 3 18 99 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:yqBtWhzluhxOMG7S7qnlj/s6barDh4Ubm6XTncn+ a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv m=audio 65426 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv <-------------> --- (20 headers 34 lines) --- Sending to 192.168.7.65:2048 (no NAT) Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 99 Found RTP audio format 4 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 99 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g726|speex|g722), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g726|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.7.65:65426 <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-2v3ncwcxphr7;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: l: 0 <------------> Audio is at 12384 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100011 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-2v3ncwcxphr7;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947821 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r <------------> Retransmitting #1 (no NAT) to 192.168.7.65:2048: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-2v3ncwcxphr7;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947821 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r --- <--- SIP read from UDP:192.168.7.65:2048 ---> <-------------> <--- SIP read from UDP:192.168.7.65:2048 ---> INVITE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-2v3ncwcxphr7;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 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 Require: timer Session-Expires: 600;refresher=uas Min-SE: 90 c: application/sdp l: 808 v=0 o=root 361817363 361817370 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 65426 RTP/SAVP 9 8 0 3 18 99 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:yqBtWhzluhxOMG7S7qnlj/s6barDh4Ubm6XTncn+ a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv m=audio 65426 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv <-------------> --- (20 headers 34 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-2v3ncwcxphr7;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: l: 0 <------------> Audio is at 12384 Adding codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100011 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-2v3ncwcxphr7;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947822 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r <------------> Retransmitting #2 (no NAT) to 192.168.7.65:2048: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-2v3ncwcxphr7;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer x: 600;refresher=uas m: c: application/sdp l: 438 v=0 o=nobody 2139947814 2139947821 IN IP4 192.168.7.23 s=Asterisk c=IN IP4 192.168.7.23 t=0 0 m=audio 12384 RTP/SAVP 8 9 0 3 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/6CEAG1oeyCcYRu+bAcOD7XIO4P+QSyIBiTMi16r --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-k5iea1ej8ycl;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-k5iea1ej8ycl;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-biotfmt11hod;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> ACK sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-k5iea1ej8ycl;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 6 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.7.65:2048 ---> BYE sip:+4946123240@192.168.7.23:5060 SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-d0u21mtmbv0m;rport f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 7 BYE Max-Forwards: 16 m: ;reg-id=1 User-Agent: snom360/8.4.27 RTP-RxStat: Total_Rx_Pkts=245,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=225,Tx_Pkts=49,Remote_Tx_Pkts=0 l: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.7.65:2048 (no NAT) Scheduling destruction of SIP dialog '3428bba201e0123e3a4d10b43ae2f71b@yeti.dk' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.7.65:2048 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.7.65:2048;branch=z9hG4bK-d0u21mtmbv0m;received=192.168.7.65;rport=2048 f: "WIMPy" ;tag=70x4olau35 t: ;tag=as2d8928f1 i: 3428bba201e0123e3a4d10b43ae2f71b@yeti.dk CSeq: 7 BYE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer l: 0 <------------> lx3*CLI> Disconnected from Asterisk server