trixbox*CLI> sip debug SIP Debugging re-enabled trixbox*CLI> <-- SIP read from 200.200.200.41:5060: INVITE sip:600@200.200.200.11:5060 SIP/2.0 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11 Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKbe226279c Call-ID: 76712560dda5b8d CSeq: 2009506258 INVITE Max-Forwards: 70 Content-Length: 378 Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,REFER,INFO,PRACK,UPDATE P-Asserted-Identity: Content-Type: application/sdp Contact: sip:163@200.200.200.41:5060 Supported: replaces User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10 v=0 o=MxSIP 0 1 IN IP4 200.200.200.168 s=SIP Call c=IN IP4 200.200.200.168 t=0 0 a=sendrecv m=audio 5004 RTP/AVP 0 8 9 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:4 annexa=no a=fmtp:18 annexb=no a=fmtp:101 0-15 a=ptime:20 --- (14 headers 18 lines) --- Using INVITE request as basis request - 76712560dda5b8d Sending to 200.200.200.41 : 5060 (non-NAT) Found peer 'HiPath3800' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 200.200.200.168:5004 Found description format PCMU Found description format PCMA Found description format G722 Found description format G729 Found description format G723 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 600 in from-sip-external (domain 200.200.200.11) list_route: hop: Transmitting (no NAT) to 200.200.200.41:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKbe226279c;received=200.200.200.41 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11 Call-ID: 76712560dda5b8d CSeq: 2009506258 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 200.200.200.41:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKbe226279c;received=200.200.200.41 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11;tag=as1bc531fd Call-ID: 76712560dda5b8d CSeq: 2009506258 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- We're at 200.200.200.11 port 11088 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 200.200.200.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKbe226279c;received=200.200.200.41 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11;tag=as1bc531fd Call-ID: 76712560dda5b8d CSeq: 2009506258 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2802 2802 IN IP4 200.200.200.11 s=session c=IN IP4 200.200.200.11 t=0 0 m=audio 11088 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 - - - - --- trixbox*CLI> <-- SIP read from 200.200.200.41:5060: ACK sip:600@200.200.200.11 SIP/2.0 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11;tag=as1bc531fd Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKdf087f008 Call-ID: 76712560dda5b8d CSeq: 2009506258 ACK Max-Forwards: 70 Content-Length: 0 User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10 --- (9 headers 0 lines) --- trixbox*CLI> <-- SIP read from 200.200.200.41:5060: INVITE sip:600@200.200.200.11 SIP/2.0 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11;tag=as1bc531fd Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK0d1c2a983 Call-ID: 76712560dda5b8d CSeq: 2009506259 INVITE Max-Forwards: 70 Content-Length: 366 Content-Type: application/sdp Supported: replaces Contact: sip:163@200.200.200.41:5060 User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10 v=0 o=MxSIP 0 2 IN IP4 200.200.200.168 s=SIP Call c=IN IP4 200.200.200.168 t=0 0 m=audio 5004 RTP/AVP 0 8 9 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:4 annexa=no a=fmtp:101 0-15 a=inactive --- (12 headers 17 lines) --- Using INVITE request as basis request - 76712560dda5b8d Sending to 200.200.200.41 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 200.200.200.168:5004 Found description format PCMU Found description format PCMA Found description format G722 Found description format G729 Found description format G723 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 200.200.200.11 port 11088 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 200.200.200.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK0d1c2a983;received=200.200.200.41 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11;tag=as1bc531fd Call-ID: 76712560dda5b8d CSeq: 2009506259 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2802 2803 IN IP4 200.200.200.11 s=session c=IN IP4 200.200.200.11 t=0 0 m=audio 11088 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 - - - - --- trixbox*CLI> <-- SIP read from 200.200.200.41:5060: ACK sip:600@200.200.200.11 SIP/2.0 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11;tag=as1bc531fd Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK95f8b9aa6 Call-ID: 76712560dda5b8d CSeq: 2009506259 ACK Max-Forwards: 70 Content-Length: 0 User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10 --- (9 headers 0 lines) --- trixbox*CLI> <-- SIP read from 200.200.200.41:5060: BYE sip:600@200.200.200.11 SIP/2.0 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11;tag=as1bc531fd Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK378992d6f Call-ID: 76712560dda5b8d CSeq: 2009506260 BYE Max-Forwards: 70 Content-Length: 0 Reason:Q.850; cause=16; text="Normal call clearing" Supported: replaces User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10 --- (11 headers 0 lines) --- Sending to 200.200.200.41 : 5060 (non-NAT) Transmitting (no NAT) to 200.200.200.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK378992d6f;received=200.200.200.41 From: 163 ;tag=b175082a51435db To: sip:600@200.200.200.11;tag=as1bc531fd Call-ID: 76712560dda5b8d CSeq: 2009506260 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Destroying call '76712560dda5b8d' trixbox*CLI>