pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.0.4.3:5060;branch=z9hG4bK50383360;rport=5060 From: "asterisk" ;tag=as42cadd58 To: Call-ID: 044beff041fecf6d7eca64aa7b59f6c2@10.0.4.3 CSeq: 102 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '044beff041fecf6d7eca64aa7b59f6c2@10.0.4.3' Method: NOTIFY pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> INVITE sip:16175043652@10.0.4.3;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-x65qzoocqrd1;rport From: "BTC" ;tag=4sp9m5wwkm To: Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.8 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 475 v=0 o=root 1281895614 1281895614 IN IP4 10.0.249.254 s=call c=IN IP4 10.0.249.254 t=0 0 m=audio 51846 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0XTYHWud6CX7Y8jkdNMtZlQTMXg3WO7Nkowm/60j a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (18 headers 19 lines) --- Sending to 10.0.249.254 : 2054 (NAT) Using INVITE request as basis request - 3c643582dbba-rgbkvqf3mple@snom360-00041323719F Found peer '200' <--- Reliably Transmitting (no NAT) to 10.0.249.254:2054 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-x65qzoocqrd1;received=10.0.249.254;rport=2054 From: "BTC" ;tag=4sp9m5wwkm To: ;tag=as62813a25 Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02e143d1" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c643582dbba-rgbkvqf3mple@snom360-00041323719F' in 32000 ms (Method: INVITE) pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> ACK sip:16175043652@10.0.4.3;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-x65qzoocqrd1;rport From: "BTC" ;tag=4sp9m5wwkm To: ;tag=as62813a25 Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> INVITE sip:16175043652@10.0.4.3;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-zvpso6od9cuf;rport From: "BTC" ;tag=4sp9m5wwkm To: Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.8 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="02e143d1",uri="sip:16175043652@10.0.4.3;user=phone",response="74b4594598cbba4f4f4147f17bd5dee7",algorithm=md5 Content-Type: application/sdp Content-Length: 475 v=0 o=root 1281895614 1281895614 IN IP4 10.0.249.254 s=call c=IN IP4 10.0.249.254 t=0 0 m=audio 51846 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0XTYHWud6CX7Y8jkdNMtZlQTMXg3WO7Nkowm/60j a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (19 headers 19 lines) --- Sending to 10.0.249.254 : 2054 (NAT) Using INVITE request as basis request - 3c643582dbba-rgbkvqf3mple@snom360-00041323719F Found peer '200' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 10.0.249.254:51846 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g722 for ID 9 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g729 for ID 18 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.249.254:51846 Looking for 16175043652 in numberplan-custom-1 (domain 10.0.4.3) list_route: hop: <--- Transmitting (no NAT) to 10.0.249.254:2054 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-zvpso6od9cuf;received=10.0.249.254;rport=2054 From: "BTC" ;tag=4sp9m5wwkm To: Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [16175043652@numberplan-custom-1:1] Macro("SIP/200-00745370", "trunkdial|SIP/trunk_1/16175043652") in new stack -- Executing [s@macro-trunkdial:1] Dial("SIP/200-00745370", "SIP/trunk_1/16175043652") in new stack -- Called trunk_1/16175043652 -- Call on SIP/trunk_1-00766f50 left from hold -- SIP/trunk_1-00766f50 is making progress passing it to SIP/200-00745370 Audio is at 10.0.4.3 port 18272 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.0.249.254:2054 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-zvpso6od9cuf;received=10.0.249.254;rport=2054 From: "BTC" ;tag=4sp9m5wwkm To: ;tag=as7ea30464 Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 230 v=0 o=root 3216 3216 IN IP4 10.0.4.3 s=session c=IN IP4 10.0.4.3 t=0 0 m=audio 18272 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Call on SIP/trunk_1-00766f50 left from hold -- SIP/trunk_1-00766f50 answered SIP/200-00745370 Audio is at 10.0.4.3 port 18272 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.0.249.254:2054 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-zvpso6od9cuf;received=10.0.249.254;rport=2054 From: "BTC" ;tag=4sp9m5wwkm To: ;tag=as7ea30464 Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 230 v=0 o=root 3216 3217 IN IP4 10.0.4.3 s=session c=IN IP4 10.0.4.3 t=0 0 m=audio 18272 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/200-00745370 and SIP/trunk_1-00766f50 pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> ACK sip:16175043652@10.0.4.3 SIP/2.0 Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-00q2sszyhpi6;rport From: "BTC" ;tag=4sp9m5wwkm To: ;tag=as7ea30464 Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> BYE sip:16175043652@10.0.4.3 SIP/2.0 Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7rlxtyasgr1y;rport From: "BTC" ;tag=4sp9m5wwkm To: ;tag=as7ea30464 Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/6.5.8 RTP-RxStat: Total_Rx_Pkts=299,Rx_Pkts=299,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=547,Tx_Pkts=547,Remote_Tx_Pkts=297 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.0.249.254 : 2054 (NAT) <--- Transmitting (NAT) to 10.0.249.254:2054 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7rlxtyasgr1y;received=10.0.249.254;rport=2054 From: "BTC" ;tag=4sp9m5wwkm To: ;tag=as7ea30464 Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F CSeq: 3 BYE ser-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (macro-trunkdial, s, 1) exited non-zero on 'SIP/200-00745370' in macro 'trunkdial' == Spawn extension (macro-trunkdial, s, 1) exited non-zero on 'SIP/200-00745370' Really destroying SIP dialog '3c643582dbba-rgbkvqf3mple@snom360-00041323719F' Method: BYE pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> INVITE sip:16175043652@10.0.4.3;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7gfpqr1tmgde;rport From: "BTC" ;tag=9xwftzfxej To: Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.8 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 473 v=0 o=root 449870880 449870880 IN IP4 10.0.249.254 s=call c=IN IP4 10.0.249.254 t=0 0 m=audio 49762 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aF3cZLtxa7hvj2dP6JZXgnz0/HJ9NfXNu8En92/K a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (18 headers 19 lines) --- Sending to 10.0.249.254 : 2054 (NAT) Using INVITE request as basis request - 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F Found peer '200' <--- Reliably Transmitting (no NAT) to 10.0.249.254:2054 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7gfpqr1tmgde;received=10.0.249.254;rport=2054 From: "BTC" ;tag=9xwftzfxej To: ;tag=as24d8bfd6 Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a74886a" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c6435a130d4-uyv1juesmkdd@snom360-00041323719F' in 32000 ms (Method: INVITE) pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> ACK sip:16175043652@10.0.4.3;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7gfpqr1tmgde;rport From: "BTC" ;tag=9xwftzfxej To: ;tag=as24d8bfd6 Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> INVITE sip:16175043652@10.0.4.3;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-w9kwf6sh9i9m;rport From: "BTC" ;tag=9xwftzfxej To: Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.5.8 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="0a74886a",uri="sip:16175043652@10.0.4.3;user=phone",response="585b1b9653c3378602b1545b87311b9d",algorithm=md5 Content-Type: application/sdp Content-Length: 473 v=0 o=root 449870880 449870880 IN IP4 10.0.249.254 s=call c=IN IP4 10.0.249.254 t=0 0 m=audio 49762 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aF3cZLtxa7hvj2dP6JZXgnz0/HJ9NfXNu8En92/K a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (19 headers 19 lines) --- Sending to 10.0.249.254 : 2054 (NAT) Using INVITE request as basis request - 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F Found peer '200' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 10.0.249.254:49762 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g722 for ID 9 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g729 for ID 18 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.249.254:49762 Looking for 16175043652 in numberplan-custom-1 (domain 10.0.4.3) list_route: hop: <--- Transmitting (no NAT) to 10.0.249.254:2054 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-w9kwf6sh9i9m;received=10.0.249.254;rport=2054 From: "BTC" ;tag=9xwftzfxej To: Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [16175043652@numberplan-custom-1:1] Macro("SIP/200-00753790", "trunkdial|SIP/trunk_1/16175043652") in new stack -- Executing [s@macro-trunkdial:1] Dial("SIP/200-00753790", "SIP/trunk_1/16175043652") in new stack -- Called trunk_1/16175043652 -- SIP/trunk_1-00743740 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-trunkdial:2] Goto("SIP/200-00753790", "s-CONGESTION|1") in new stack -- Goto (macro-trunkdial,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-trunkdial:1] NoOp("SIP/200-00753790", "") in new stack == Auto fallthrough, channel 'SIP/200-00753790' status is 'CONGESTION' pbx*CLI> <--- Transmitting (no NAT) to 10.0.249.254:2054 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-w9kwf6sh9i9m;received=10.0.249.254;rport=2054 From: "BTC" ;tag=9xwftzfxej To: ;tag=as3159d4b5 Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 <------------> pbx*CLI> <--- SIP read from 10.0.249.254:2054 ---> ACK sip:16175043652@10.0.4.3;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-w9kwf6sh9i9m;rport From: "BTC" ;tag=9xwftzfxej To: ;tag=as3159d4b5 Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '3c6435a130d4-uyv1juesmkdd@snom360-00041323719F' Method: ACK [Jun 13 16:44:26] NOTICE[3267]: chan_sip.c:7147 sip_reregister: -- Re-registration for prichar@inbound1.vitelity.net [Jun 13 16:44:26] NOTICE[3267]: chan_sip.c:12117 handle_response_register: Outbound Registration: Expiry for inbound1.vitelity.net is 60 sec (Scheduling reregistration in 45 s)