asterisk*CLI> Verbosity is at least 10 Core debug is at least 1 asterisk*CLI> [Apr 12 18:00:19] NOTICE[8401]: chan_sip.c:12155 sip_reregister: -- Re-registration for just-another-number@voip.eutelia.it > doing dnsmgr_lookup for 'voip.eutelia.it' asterisk*CLI>  > ast_get_srv: SRV lookup for '_sip._udp.voip.eutelia.it' mapped to host voip.eutelia.it, port 5060 asterisk*CLI> [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1 - REGISTER (No RTP) [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:2894 obproxy_get: OBPROXY: Not applying OBproxy to this call [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:12389 transmit_register: Scheduled a registration timeout for voip.eutelia.it id #685 [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:12438 transmit_register: >>> Re-using Auth data for just-another-number@voip.eutelia.it [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method REGISTER - callid 28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1 REGISTER 11 headers, 0 lines Reliably Transmitting (no NAT) to 83.211.227.21:5060: REGISTER sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4daaf53c Max-Forwards: 70 From: ;tag=as6737a08f To: Call-ID: 28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1 CSeq: 104 REGISTER User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Authorization: Digest username="just-another-number", realm="voip.eutelia.it", algorithm=MD5, uri="sip:voip.eutelia.it", nonce="4da476e624b19bf7e2139cde2b84bad539a2e02b", response="afb25eaeeb1953beceb639842d4ff71e", qop=auth, cnonce="33575ae5", nc=00000002 Expires: 120 Contact: Content-Length: 0 --- [Apr 12 18:00:19] NOTICE[8401]: chan_sip.c:12155 sip_reregister: -- Re-registration for my-eutelia-number@voip.eutelia.it > doing dnsmgr_lookup for 'voip.eutelia.it' asterisk*CLI>  > ast_get_srv: SRV lookup for '_sip._udp.voip.eutelia.it' mapped to host voip.eutelia.it, port 5060 asterisk*CLI> [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1 - REGISTER (No RTP) [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:2894 obproxy_get: OBPROXY: Not applying OBproxy to this call [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:12389 transmit_register: Scheduled a registration timeout for voip.eutelia.it id #687 [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:12438 transmit_register: >>> Re-using Auth data for my-eutelia-number@voip.eutelia.it [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method REGISTER - callid 34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1 REGISTER 11 headers, 0 lines Reliably Transmitting (no NAT) to 83.211.227.21:5060: REGISTER sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK60258a3d Max-Forwards: 70 From: ;tag=as1a199ce6 To: Call-ID: 34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1 CSeq: 104 REGISTER User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Authorization: Digest username="my-eutelia-number", realm="voip.eutelia.it", algorithm=MD5, uri="sip:voip.eutelia.it", nonce="4da476e624b19bf7e2139cde2b84bad539a2e02b", response="2cb34d97b6c2fee2968c38f731511fc5", qop=auth, cnonce="68b7bb8b", nc=00000002 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4daaf53c From: ;tag=as6737a08f To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.e1c3 Call-ID: 28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1 CSeq: 104 REGISTER WWW-Authenticate: Digest realm="voip.eutelia.it", nonce="4da4774f72a6a3d14c4e9302cc1c36496eae24e5", qop="auth", stale=true Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Apr 12 18:00:19] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1' of Request 104: Match Found Responding to challenge, registration to domain/host name voip.eutelia.it > doing dnsmgr_lookup for 'voip.eutelia.it' asterisk*CLI>  > ast_get_srv: SRV lookup for '_sip._udp.voip.eutelia.it' mapped to host voip.eutelia.it, port 5060 asterisk*CLI> [Apr 12 18:00:20] DEBUG[8401]: chan_sip.c:2821 initialize_initreq: Initializing already initialized SIP dialog 28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1 (presumably reinvite) REGISTER 11 headers, 0 lines Reliably Transmitting (no NAT) to 83.211.227.21:5060: REGISTER sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK63423951 Max-Forwards: 70 From: ;tag=as2ff19640 To: Call-ID: 28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1 CSeq: 105 REGISTER User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Authorization: Digest username="just-another-number", realm="voip.eutelia.it", algorithm=MD5, uri="sip:voip.eutelia.it", nonce="4da4774f72a6a3d14c4e9302cc1c36496eae24e5", response="7b4903f3bf10bc3b30a57d181905ae6f", qop=auth, cnonce="1f308444", nc=00000001 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK60258a3d From: ;tag=as1a199ce6 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.fc1d Call-ID: 34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1 CSeq: 104 REGISTER WWW-Authenticate: Digest realm="voip.eutelia.it", nonce="4da4774f72a6a3d14c4e9302cc1c36496eae24e5", qop="auth", stale=true Server: SPS EUT RM GW 01 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Apr 12 18:00:20] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1' of Request 104: Match Found Responding to challenge, registration to domain/host name voip.eutelia.it > doing dnsmgr_lookup for 'voip.eutelia.it' asterisk*CLI>  > ast_get_srv: SRV lookup for '_sip._udp.voip.eutelia.it' mapped to host voip.eutelia.it, port 5060 asterisk*CLI> [Apr 12 18:00:20] DEBUG[8401]: chan_sip.c:2821 initialize_initreq: Initializing already initialized SIP dialog 34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1 (presumably reinvite) REGISTER 11 headers, 0 lines Reliably Transmitting (no NAT) to 83.211.227.21:5060: REGISTER sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK1227071a Max-Forwards: 70 From: ;tag=as6722ea16 To: Call-ID: 34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1 CSeq: 105 REGISTER User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Authorization: Digest username="my-eutelia-number", realm="voip.eutelia.it", algorithm=MD5, uri="sip:voip.eutelia.it", nonce="4da4774f72a6a3d14c4e9302cc1c36496eae24e5", response="7bd901ce90a1cf3e0bbd6030975cc4d9", qop=auth, cnonce="17f5d1f5", nc=00000001 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK63423951 From: ;tag=as2ff19640 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.b458 Call-ID: 28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1 CSeq: 105 REGISTER Date: Tue, 12 Apr 2011 16:00:20 GMT Contact: ;q=0.5;expires=120 Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Apr 12 18:00:20] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1' of Request 105: Match Found [Apr 12 18:00:20] DEBUG[8401]: chan_sip.c:19487 handle_response_register: Registration successful [Apr 12 18:00:20] DEBUG[8401]: chan_sip.c:19489 handle_response_register: Cancelling timeout 685 Scheduling destruction of SIP dialog '28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1' in 32000 ms (Method: REGISTER) [Apr 12 18:00:20] NOTICE[8401]: chan_sip.c:19539 handle_response_register: Outbound Registration: Expiry for voip.eutelia.it is 120 sec (Scheduling reregistration in 105 s) asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK1227071a From: ;tag=as6722ea16 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.d08b Call-ID: 34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1 CSeq: 105 REGISTER Date: Tue, 12 Apr 2011 16:00:20 GMT Contact: ;q=0.5;expires=120 Server: SPS EUT RM GW 01 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Apr 12 18:00:20] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1' of Request 105: Match Found [Apr 12 18:00:20] DEBUG[8401]: chan_sip.c:19487 handle_response_register: Registration successful [Apr 12 18:00:20] DEBUG[8401]: chan_sip.c:19489 handle_response_register: Cancelling timeout 687 Scheduling destruction of SIP dialog '34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1' in 32000 ms (Method: REGISTER) [Apr 12 18:00:20] NOTICE[8401]: chan_sip.c:19539 handle_response_register: Outbound Registration: Expiry for voip.eutelia.it is 120 sec (Scheduling reregistration in 105 s) asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> INVITE sip:dialed-number@asterisk-public-ip;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-2a5349bef5cb39ce-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "153";tag=fc8e837e Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.6739 Content-Length: 304 v=0 o=Z 0 0 IN IP4 192.168.1.44 s=Z c=IN IP4 192.168.1.44 t=0 0 m=audio 8000 RTP/AVP 98 0 8 3 110 101 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (12 headers 15 lines) --- asterisk*CLI>  == Using UDPTL CoS mark 5 [Apr 12 18:00:21] DEBUG[8401]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off [Apr 12 18:00:21] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. - INVITE (No RTP) asterisk*CLI> Sending to 192.168.1.44:5060 (no NAT) [Apr 12 18:00:21] DEBUG[8401]: chan_sip.c:21172 handle_request_invite: Initializing initreq for method INVITE - callid ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. Using INVITE request as basis request - ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. Found peer '153' for '153' from 192.168.1.44:5060 [Apr 12 18:00:21] DEBUG[8401]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off <--- Reliably Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-2a5349bef5cb39ce-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=fc8e837e To: ;tag=as078c8ab4 Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59266a67" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg.' in 6400 ms (Method: INVITE) asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> ACK sip:dialed-number@asterisk-public-ip;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-2a5349bef5cb39ce-1---d8754z-;rport Max-Forwards: 70 To: ;tag=as078c8ab4 From: "153";tag=fc8e837e Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Apr 12 18:00:21] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on 'ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg.' of Response 1: Match Found asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> INVITE sip:dialed-number@asterisk-public-ip;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-b1bf197cdb006e5b-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "153";tag=fc8e837e Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="59266a67",uri="sip:dialed-number@asterisk-public-ip;transport=UDP",response="1db8053d7963e7a6307c4b361542883e",algorithm=MD5 Content-Length: 304 v=0 o=Z 0 0 IN IP4 192.168.1.44 s=Z c=IN IP4 192.168.1.44 t=0 0 m=audio 8000 RTP/AVP 98 0 8 3 110 101 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (13 headers 15 lines) --- Sending to 192.168.1.44:5060 (no NAT) [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:21172 handle_request_invite: Initializing initreq for method INVITE - callid ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. Using INVITE request as basis request - ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. Found peer '153' for '153' from 192.168.1.44:5060 [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:344 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x1349808' [Apr 12 18:00:22] DEBUG[8401]: res_rtp_asterisk.c:472 ast_rtp_new: Allocated port 10260 for RTP instance '0x1349808' [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:353 ast_rtp_instance_new: RTP instance '0x1349808' is setup and ready to go [Apr 12 18:00:22] DEBUG[8401]: res_rtp_asterisk.c:2370 ast_rtp_prop_set: Setup RTCP on RTP instance '0x1349808' == Using SIP RTP CoS mark 5 [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:4683 do_setnat: Setting NAT on RTP to Off [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 98 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 98 based on m type on 0x7fa1ef1041c0 Found RTP audio format 0 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7fa1ef1041c0 Found RTP audio format 8 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fa1ef1041c0 Found RTP audio format 3 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0x7fa1ef1041c0 Found RTP audio format 110 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 110 based on m type on 0x7fa1ef1041c0 Found RTP audio format 101 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa1ef1041c0 Found audio description format iLBC for ID 98 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format speex for ID 110 Found audio description format telephone-event for ID 101 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x7fa1ef1041c0 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 3 on 0x7fa1ef1041c0 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x7fa1ef1041c0 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 98 on 0x7fa1ef1041c0 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa1ef1041c0 [Apr 12 18:00:22] DEBUG[8401]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 110 on 0x7fa1ef1041c0 Capabilities: us - 0x8 (alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/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|) [Apr 12 18:00:22] DEBUG[8401]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x1349808' Peer audio RTP is at port 192.168.1.44:8000 [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:21320 handle_request_invite: Checking SIP call limits for device 153 Looking for dialed-number in phones-sip (domain asterisk-public-ip) list_route: hop: <--- Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-b1bf197cdb006e5b-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=fc8e837e To: Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 2 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Apr 12 18:00:22] DEBUG[8487]: pbx.c:4067 pbx_extension_helper: Launching 'Dial' -- Executing [dialed-number@phones-sip:1] Dial("SIP/153-0000000a", "SIP/eutelia/dialed-number,120") in new stack [Apr 12 18:00:22] DEBUG[8487]: chan_sip.c:25071 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw) == Using UDPTL CoS mark 5 [Apr 12 18:00:22] DEBUG[8487]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 00654dbf38c334d876f886fe37cb97d2@127.0.1.1:0 - INVITE (No RTP) [Apr 12 18:00:22] DEBUG[8487]: rtp_engine.c:344 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x13807b8' [Apr 12 18:00:22] DEBUG[8487]: res_rtp_asterisk.c:472 ast_rtp_new: Allocated port 18602 for RTP instance '0x13807b8' [Apr 12 18:00:22] DEBUG[8487]: rtp_engine.c:353 ast_rtp_instance_new: RTP instance '0x13807b8' is setup and ready to go [Apr 12 18:00:22] DEBUG[8487]: res_rtp_asterisk.c:2370 ast_rtp_prop_set: Setup RTCP on RTP instance '0x13807b8' == Using SIP RTP CoS mark 5 [Apr 12 18:00:22] DEBUG[8487]: chan_sip.c:4683 do_setnat: Setting NAT on RTP to Off [Apr 12 18:00:22] DEBUG[8487]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off [Apr 12 18:00:22] DEBUG[8487]: chan_sip.c:2894 obproxy_get: OBPROXY: Not applying OBproxy to this call [Apr 12 18:00:22] DEBUG[8487]: rtp_engine.c:1446 ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 'SIP/eutelia-0000000b' with that of 'SIP/153-0000000a' [Apr 12 18:00:22] DEBUG[8487]: channel.c:5900 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Apr 12 18:00:22] DEBUG[8487]: channel.c:5900 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Apr 12 18:00:22] DEBUG[8487]: channel.c:5900 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Apr 12 18:00:22] DEBUG[8487]: channel.c:5900 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Apr 12 18:00:22] DEBUG[8487]: channel.c:5900 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Apr 12 18:00:22] DEBUG[8487]: channel.c:5900 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Apr 12 18:00:22] DEBUG[8487]: channel.c:5900 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Apr 12 18:00:22] DEBUG[8487]: channel.c:5900 ast_channel_inherit_variables: Not copying variable SIPURI. [Apr 12 18:00:22] DEBUG[8487]: chan_sip.c:5212 sip_call: Outgoing Call for dialed-number [Apr 12 18:00:22] DEBUG[8487]: chan_sip.c:10570 add_sdp: ** Our capability: 0x8 (alaw) Video flag: False Text flag: False [Apr 12 18:00:22] DEBUG[8487]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Apr 12 18:00:22] DEBUG[8487]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method INVITE - callid 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 Reliably Transmitting (no NAT) to 83.211.227.21:5060: INVITE sip:dialed-number@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK63b9669b Max-Forwards: 70 From: "153" ;tag=as1220bb53 To: Contact: Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 16:00:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 1476143484 1476143484 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 18602 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called eutelia/dialed-number asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK63b9669b From: "153" ;tag=as1220bb53 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.2177 Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="voip.eutelia.it", nonce="4da47752b38cc32675ef46e0822052a663fe6204", qop="auth" Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:3742 __sip_ack: Acked pending invite 102 [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060' of Request 102: Match Found Transmitting (no NAT) to 83.211.227.21:5060: ACK sip:dialed-number@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK63b9669b Max-Forwards: 70 From: "153" ;tag=as1220bb53 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.2177 Contact: Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Content-Length: 0 --- [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:10570 add_sdp: ** Our capability: 0x8 (alaw) Video flag: False Text flag: False [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 83.211.227.21:5060: INVITE sip:dialed-number@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK69b6069a Max-Forwards: 70 From: "153" ;tag=as1220bb53 To: Contact: Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Proxy-Authorization: Digest username="my-eutelia-number", realm="voip.eutelia.it", algorithm=MD5, uri="sip:dialed-number@voip.eutelia.it", nonce="4da47752b38cc32675ef46e0822052a663fe6204", response="95ee0ab1187fd7e5b67d21b8a2a73b53", qop=auth, cnonce="53e1389a", nc=00000001 Date: Tue, 12 Apr 2011 16:00:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 1476143484 1476143485 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 18602 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK69b6069a From: "153" ;tag=as1220bb53 To: Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 103 INVITE Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Apr 12 18:00:22] DEBUG[8401]: chan_sip.c:3821 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060' Request 103: Found asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK69b6069a From: "153" ;tag=as1220bb53 To: ;tag=14AD594C-1856 Date: Tue, 12 Apr 2011 16:00:22 GMT Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 249 v=0 o=CiscoSystemsSIP-GW-UserAgent 1679 3418 IN IP4 83.211.2.218 s=SIP Call c=IN IP4 83.211.227.12 t=0 0 m=audio 59568 RTP/AVP 8 101 c=IN IP4 83.211.227.12 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- [Apr 12 18:00:23] DEBUG[8401]: chan_sip.c:3821 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060' Request 103: Found Found RTP audio format 8 [Apr 12 18:00:23] DEBUG[8401]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fa1ef104bf0 Found RTP audio format 101 [Apr 12 18:00:23] DEBUG[8401]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa1ef104bf0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 [Apr 12 18:00:23] DEBUG[8401]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x7fa1ef104bf0 [Apr 12 18:00:23] DEBUG[8401]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa1ef104bf0 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|) [Apr 12 18:00:23] DEBUG[8401]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x13807b8' Peer audio RTP is at port 83.211.227.12:59568 [Apr 12 18:00:23] DEBUG[8401]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL -- SIP/eutelia-0000000b is making progress passing it to SIP/153-0000000a [Apr 12 18:00:23] DEBUG[8487]: rtp_engine.c:1531 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/153-0000000a' with that of 'SIP/eutelia-0000000b' [Apr 12 18:00:23] DEBUG[8487]: chan_sip.c:10924 transmit_response_with_sdp: Setting framing from config on incoming call [Apr 12 18:00:23] DEBUG[8487]: chan_sip.c:10570 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Apr 12 18:00:23] DEBUG[8487]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-b1bf197cdb006e5b-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=fc8e837e To: ;tag=as228b86dd Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 2 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 249 v=0 o=root 346772113 346772113 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 10260 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> asterisk*CLI> [Apr 12 18:00:24] DEBUG[8487]: res_rtp_asterisk.c:1239 ast_rtp_write: Ooh, format changed from unknown to alaw [Apr 12 18:00:24] DEBUG[8487]: res_rtp_asterisk.c:1270 ast_rtp_write: Created smoother: format: alaw ms: 20 len: 160 [Apr 12 18:00:24] DEBUG[8487]: res_rtp_asterisk.c:1140 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x13807b8' asterisk*CLI> [Apr 12 18:00:24] DEBUG[8487]: res_rtp_asterisk.c:1239 ast_rtp_write: Ooh, format changed from unknown to alaw [Apr 12 18:00:24] DEBUG[8487]: res_rtp_asterisk.c:1270 ast_rtp_write: Created smoother: format: alaw ms: 20 len: 160 asterisk*CLI> [Apr 12 18:00:27] DEBUG[8487]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 76 bytes asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> <-------------> asterisk*CLI> [Apr 12 18:00:30] DEBUG[8487]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 76 bytes asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK69b6069a From: "153" ;tag=as1220bb53 To: ;tag=14AD594C-1856 Date: Tue, 12 Apr 2011 16:00:22 GMT Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 249 v=0 o=CiscoSystemsSIP-GW-UserAgent 1679 3418 IN IP4 83.211.2.218 s=SIP Call c=IN IP4 83.211.227.12 t=0 0 m=audio 59568 RTP/AVP 8 101 c=IN IP4 83.211.227.12 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:3742 __sip_ack: Acked pending invite 103 [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060' of Request 103: Match Found list_route: hop: set_destination: Parsing for address/port to send to asterisk*CLI> set_destination: set destination to 83.211.227.21:5060 Transmitting (no NAT) to 83.211.227.21:5060: ACK sip:494dialed-number@83.211.2.218:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK3e93f5e8 Route: Max-Forwards: 70 From: "153" ;tag=as1220bb53 To: ;tag=14AD594C-1856 Contact: Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 103 ACK asterisk*CLI> User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Content-Length: 0 --- -- SIP/eutelia-0000000b answered SIP/153-0000000a [Apr 12 18:00:31] DEBUG[8487]: rtp_engine.c:1531 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/153-0000000a' with that of 'SIP/eutelia-0000000b' [Apr 12 18:00:31] DEBUG[8487]: chan_sip.c:6028 sip_answer: SIP answering channel: SIP/153-0000000a [Apr 12 18:00:31] DEBUG[8487]: chan_sip.c:10924 transmit_response_with_sdp: Setting framing from config on incoming call [Apr 12 18:00:31] DEBUG[8487]: chan_sip.c:10570 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Apr 12 18:00:31] DEBUG[8487]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-b1bf197cdb006e5b-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=fc8e837e To: ;tag=as228b86dd Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 2 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 249 v=0 o=root 346772113 346772114 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 10260 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> -- Locally bridging SIP/153-0000000a and SIP/eutelia-0000000b asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> ACK sip:dialed-number@asterisk-public-ip:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-22f4e8b869add7ff-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as228b86dd From: "153";tag=fc8e837e Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 2 ACK User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="59266a67",uri="sip:dialed-number@asterisk-public-ip;transport=UDP",response="1db8053d7963e7a6307c4b361542883e",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on 'ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg.' of Response 2: Match Found asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> INVITE sip:dialed-number@asterisk-public-ip:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-b271493bfe5268cf-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as228b86dd From: "153";tag=fc8e837e Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 3 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="59266a67",uri="sip:dialed-number@asterisk-public-ip:5060",response="d97a685457be4fec556955a1824da670",algorithm=MD5 Content-Length: 333 v=0 o=Z 0 1 IN IP4 192.168.1.44 s=Z c=IN IP4 192.168.1.44 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 <-------------> --- (13 headers 15 lines) --- Sending to 192.168.1.44:5060 (no NAT) [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:21172 handle_request_invite: Initializing initreq for method INVITE - callid ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. Got T.38 offer in SDP in dialog ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. 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. [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:8570 process_sdp: Peer T.38 UDPTL is at port 192.168.1.44:8000 <--- Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-b271493bfe5268cf-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=fc8e837e To: ;tag=as228b86dd Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 3 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 [Apr 12 18:00:31] DEBUG[8487]: chan_sip.c:10715 add_sdp: T.38 UDPTL is at asterisk-public-ip port 4975 [Apr 12 18:00:31] DEBUG[8487]: chan_sip.c:2821 initialize_initreq: Initializing already initialized SIP dialog 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 (presumably reinvite) Reliably Transmitting (no NAT) to 83.211.227.21:5060: INVITE sip:494dialed-number@83.211.2.218:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK65ad54f2 Route: Max-Forwards: 70 From: "153" ;tag=as1220bb53 To: ;tag=14AD594C-1856 Contact: Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze 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: 279 v=0 o=root 1476143484 1476143486 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=image 4975 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:204 a=T38FaxUdpEC:t38UDPFEC --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK65ad54f2 From: "153" ;tag=as1220bb53 To: ;tag=14AD594C-1856 Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 104 INVITE Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:3821 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060' Request 104: Found asterisk*CLI> [Apr 12 18:00:31] DEBUG[8487]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 72 bytes asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK65ad54f2 From: "153" ;tag=as1220bb53 To: ;tag=14AD594C-1856 Date: Tue, 12 Apr 2011 16:00:31 GMT Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 264 v=0 o=CiscoSystemsSIP-GW-UserAgent 1679 3419 IN IP4 83.211.2.218 s=SIP Call c=IN IP4 83.211.227.12 t=0 0 m=image 55110 udptl t38 c=IN IP4 83.211.227.12 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:72 <-------------> --- (15 headers 11 lines) --- [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:3742 __sip_ack: Acked pending invite 104 [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060' of Request 104: Match Found Got T.38 offer in SDP in dialog 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:9002 process_sdp_a_image: Overriding T38FaxMaxDatagram '72' with '400' 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. [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:8570 process_sdp: Peer T.38 UDPTL is at port 83.211.227.12:55110 set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 Transmitting (no NAT) to 83.211.227.21:5060: ACK sip:494dialed-number@83.211.2.218:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK7a0b3490 Route: Max-Forwards: 70 From: "153" ;tag=as1220bb53 To: ;tag=14AD594C-1856 Contact: asterisk*CLI> Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Content-Length: 0 --- [Apr 12 18:00:31] DEBUG[8487]: chan_sip.c:10715 add_sdp: T.38 UDPTL is at asterisk-public-ip port 4136 <--- Reliably Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-b271493bfe5268cf-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=fc8e837e To: ;tag=as228b86dd Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 3 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 285 v=0 o=root 346772113 346772115 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=image 4136 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <------------> asterisk*CLI> [Apr 12 18:00:31] DEBUG[8487]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 60 bytes asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> ACK sip:dialed-number@asterisk-public-ip:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-963759f6cf4e8152-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as228b86dd From: "153";tag=fc8e837e Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 3 ACK User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="59266a67",uri="sip:dialed-number@asterisk-public-ip:5060",response="d97a685457be4fec556955a1824da670",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Apr 12 18:00:31] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on 'ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg.' of Response 3: Match Found asterisk*CLI> [Apr 12 18:00:33] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 323903b60373ce2765261e5244ecb70b@127.0.1.1:0 - OPTIONS (No RTP) [Apr 12 18:00:33] DEBUG[8401]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method OPTIONS - callid 6399e2f45ce98e87427753da7b38718b@192.168.3.1:5060 Reliably Transmitting (no NAT) to 192.168.3.3:2048: OPTIONS sip:104@192.168.3.3:2048;line=k5l8dvj0 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK677b1f23 Max-Forwards: 70 From: "asterisk" ;tag=as0c4fa72f To: Contact: Call-ID: 6399e2f45ce98e87427753da7b38718b@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 16:00:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.3.3:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK677b1f23 From: "asterisk" ;tag=as0c4fa72f To: Call-ID: 6399e2f45ce98e87427753da7b38718b@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.31 Accept-Language: en 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 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- [Apr 12 18:00:33] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '6399e2f45ce98e87427753da7b38718b@192.168.3.1:5060' of Request 102: Match Found Really destroying SIP dialog '6399e2f45ce98e87427753da7b38718b@192.168.3.1:5060' Method: OPTIONS asterisk*CLI> [Apr 12 18:00:33] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 6ba61a742b8869210a46847659281c24@127.0.1.1:0 - OPTIONS (No RTP) [Apr 12 18:00:33] DEBUG[8401]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method OPTIONS - callid 79bf4d384cffdf185c33c2ca3416d2f8@asterisk-public-ip:5060 Reliably Transmitting (no NAT) to 192.168.1.158:5060: OPTIONS sip:zJZlTpTaQyFc4k0voKjL@192.168.1.158 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK2c6a69b1 Max-Forwards: 70 From: "asterisk" ;tag=as19a00542 To: Contact: Call-ID: 79bf4d384cffdf185c33c2ca3416d2f8@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 16:00:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.1.158:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK2c6a69b1 To: ;tag=gk77mu8fslhc6179c761 From: "asterisk" ;tag=as19a00542 Call-ID: 79bf4d384cffdf185c33c2ca3416d2f8@asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: application/sdp Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Apr 12 18:00:33] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '79bf4d384cffdf185c33c2ca3416d2f8@asterisk-public-ip:5060' of Request 102: Match Found Really destroying SIP dialog '79bf4d384cffdf185c33c2ca3416d2f8@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI> [Apr 12 18:00:34] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 345241e10c63222614f922aa47cfb9f8@127.0.1.1:0 - OPTIONS (No RTP) [Apr 12 18:00:34] DEBUG[8401]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method OPTIONS - callid 7b6cf614349d22800c590f1048367fda@asterisk-public-ip:5060 Reliably Transmitting (no NAT) to 83.211.227.21:5060: OPTIONS sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK36e1e861 Max-Forwards: 70 From: "asterisk" ;tag=as61c5463a To: Contact: Call-ID: 7b6cf614349d22800c590f1048367fda@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 16:00:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK36e1e861 From: "asterisk" ;tag=as61c5463a To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.7355 Call-ID: 7b6cf614349d22800c590f1048367fda@asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Server: SPS EUT RM GW 04 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Apr 12 18:00:34] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '7b6cf614349d22800c590f1048367fda@asterisk-public-ip:5060' of Request 102: Match Found Really destroying SIP dialog '7b6cf614349d22800c590f1048367fda@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI> Really destroying SIP dialog '618dd9e627dcbd130d013ed74cc49bec@192.168.3.2' Method: REGISTER asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> asterisk*CLI> [Apr 12 18:00:42] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 785b91521d88e98d24c9b2b37a86b89f@127.0.1.1:0 - OPTIONS (No RTP) [Apr 12 18:00:42] DEBUG[8401]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method OPTIONS - callid 61eddc9c1f67ade51d2dced27489b759@192.168.3.1:5060 Reliably Transmitting (no NAT) to 192.168.3.4:5060: OPTIONS sip:102@192.168.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK4490039c Max-Forwards: 70 From: "asterisk" ;tag=as4a9763d8 To: Contact: Call-ID: 61eddc9c1f67ade51d2dced27489b759@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 16:00:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK4490039c From: "asterisk" ;tag=as4a9763d8 To: ;tag=809068585 Call-ID: 61eddc9c1f67ade51d2dced27489b759@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: "102" Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Apr 12 18:00:42] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '61eddc9c1f67ade51d2dced27489b759@192.168.3.1:5060' of Request 102: Match Found Really destroying SIP dialog '61eddc9c1f67ade51d2dced27489b759@192.168.3.1:5060' Method: OPTIONS asterisk*CLI> [Apr 12 18:00:46] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 25a73d540d1835813d6a74d937667c9d@127.0.1.1:0 - OPTIONS (No RTP) [Apr 12 18:00:46] DEBUG[8401]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method OPTIONS - callid 28fa2ded4be78158096ed6da009c9c0b@asterisk-public-ip:5060 Reliably Transmitting (no NAT) to 192.168.1.44:5060: OPTIONS sip:153@192.168.1.44:5060;rinstance=f8079bf81406728c;transport=UDP SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4bc88191 Max-Forwards: 70 From: "asterisk" ;tag=as7e45c95a To: Contact: Call-ID: 28fa2ded4be78158096ed6da009c9c0b@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 16:00:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4bc88191 Contact: To: ;tag=32c42330 From: "asterisk";tag=as7e45c95a Call-ID: 28fa2ded4be78158096ed6da009c9c0b@asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper rev.6739 Allow-Events: presence Content-Length: 0 <-------------> --- (13 headers 0 lines) --- [Apr 12 18:00:46] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '28fa2ded4be78158096ed6da009c9c0b@asterisk-public-ip:5060' of Request 102: Match Found Really destroying SIP dialog '28fa2ded4be78158096ed6da009c9c0b@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> <-------------> asterisk*CLI> Really destroying SIP dialog '28beaf80386ac24c3cd2d8ab33bc5dfe@127.0.1.1' Method: REGISTER asterisk*CLI> Really destroying SIP dialog '34f2b45c7ed8509b68cad2ea5db8f5d5@127.0.1.1' Method: REGISTER asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bK22a08694eb From: "wind8" ;tag=385f2600 To: "wind8" Call-ID: 618dd9e627dcbd130d013ed74cc49bec@192.168.3.2 Contact: CSeq: 16053 REGISTER Max-Forwards: 70 Expires: 60 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Authorization: Digest username="wind8",realm="asterisk",nonce="4a3eabf9",response="71c2c05a5383d486fa2100437b8f085e",uri="sip:192.168.3.1",algorithm=MD5 User-Agent: Mv-37x (904290) Content-Length: 0 <-------------> --- (13 headers 0 lines) --- [Apr 12 18:00:59] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 618dd9e627dcbd130d013ed74cc49bec@192.168.3.2 - REGISTER (No RTP) Sending to 192.168.3.2:5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK22a08694eb;received=192.168.3.2;rport=5060 From: "wind8" ;tag=385f2600 To: "wind8" ;tag=as3169b18c Call-ID: 618dd9e627dcbd130d013ed74cc49bec@192.168.3.2 CSeq: 16053 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54cd26d2" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '618dd9e627dcbd130d013ed74cc49bec@192.168.3.2' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bKa0fa8081b8 From: "wind8" ;tag=385f2600 To: "wind8" Call-ID: 618dd9e627dcbd130d013ed74cc49bec@192.168.3.2 Contact: CSeq: 16054 REGISTER Max-Forwards: 70 Expires: 60 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Authorization: Digest username="wind8",realm="asterisk",nonce="54cd26d2",response="743831a5b76397dd2e23d4efd8edec4e",uri="sip:192.168.3.1",algorithm=MD5 User-Agent: Mv-37x (904290) Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.3.2:5060 (no NAT) [Apr 12 18:00:59] DEBUG[8401]: chan_sip.c:13089 parse_register_contact: Store REGISTER's Contact header for call routing. [Apr 12 18:00:59] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 60d810a873188dbd792981ce56c9cbef@127.0.1.1:0 - OPTIONS (No RTP) [Apr 12 18:00:59] DEBUG[8401]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method OPTIONS - callid 781e5e9b2d1eed3d3dec71595814f06d@192.168.3.1:5060 Reliably Transmitting (no NAT) to 192.168.3.2:5060: OPTIONS sip:wind8@192.168.3.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK3de07508 Max-Forwards: 70 From: "asterisk" ;tag=as2d8ee193 asterisk*CLI> To: Contact: Call-ID: 781e5e9b2d1eed3d3dec71595814f06d@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 16:00:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bKa0fa8081b8;received=192.168.3.2;rport=5060 From: "wind8" ;tag=385f2600 To: "wind8" ;tag=as3169b18c Call-ID: 618dd9e627dcbd130d013ed74cc49bec@192.168.3.2 CSeq: 16054 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 12 Apr 2011 16:00:59 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '618dd9e627dcbd130d013ed74cc49bec@192.168.3.2' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK3de07508 From: "asterisk" ;tag=as2d8ee193 To: ;tag=427c0d0b Call-ID: 781e5e9b2d1eed3d3dec71595814f06d@192.168.3.1:5060 CSeq: 102 OPTIONS Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Apr 12 18:00:59] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '781e5e9b2d1eed3d3dec71595814f06d@192.168.3.1:5060' of Request 102: Match Found Really destroying SIP dialog '781e5e9b2d1eed3d3dec71595814f06d@192.168.3.1:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> BYE sip:dialed-number@asterisk-public-ip:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-12d6d68cbebaff85-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as228b86dd From: "153";tag=fc8e837e Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 4 BYE User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="59266a67",uri="sip:dialed-number@asterisk-public-ip:5060",response="9b64f8f8b7bfe58919136e3ba3a0596b",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Apr 12 18:01:03] DEBUG[8401]: chan_sip.c:22410 handle_request_bye: Initializing initreq for method BYE - callid ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. Sending to 192.168.1.44:5060 (no NAT) [Apr 12 18:01:03] DEBUG[8401]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x1349808' Scheduling destruction of SIP dialog 'ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg.' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-12d6d68cbebaff85-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=fc8e837e To: ;tag=as228b86dd Call-ID: ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. CSeq: 4 BYE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Apr 12 18:01:03] DEBUG[8487]: rtp_engine.c:888 local_bridge_loop: rtp-engine-local-bridge: Ooh, got a hangup [Apr 12 18:01:03] DEBUG[8487]: channel.c:7206 ast_channel_bridge: Returning from native bridge, channels: SIP/153-0000000a, SIP/eutelia-0000000b asterisk*CLI> [Apr 12 18:01:03] DEBUG[8487]: cdr_radius.c:207 radius_log: Unable to create RADIUS record. CDR not recorded! [Apr 12 18:01:03] DEBUG[8487]: res_config_sqlite.c:833 cdr_handler: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"153" <153>','153','dialed-number','phones-sip','SIP/153-0000000a','SIP/eutelia-0000000b','Dial','SIP/eutelia/dialed-number,120','2011-04-12 18:00:22','2011-04-12 18:00:31','2011-04-12 18:01:03','41','32','ANSWERED','DOCUMENTATION','1302624022.20') asterisk*CLI> [Apr 12 18:01:03] DEBUG[8487]: channel.c:2733 ast_hangup: Hanging up channel 'SIP/eutelia-0000000b' [Apr 12 18:01:03] DEBUG[8487]: chan_sip.c:5827 sip_hangup: Hangup call SIP/eutelia-0000000b, SIP callid 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 [Apr 12 18:01:03] DEBUG[8487]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x13807b8' Scheduling destruction of SIP dialog '676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060' in 6400 ms (Method: INVITE) asterisk*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 asterisk*CLI> Reliably Transmitting (no NAT) to 83.211.227.21:5060: BYE sip:494dialed-number@83.211.2.218:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK75fcfa4b Route: Max-Forwards: 70 From: "153" ;tag=as1220bb53 To: ;tag=14AD594C-1856 Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 CSeq: 105 BYE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Proxy-Authorization: Digest username="my-eutelia-number", realm="voip.eutelia.it", algorithm=MD5, uri="sip:494dialed-number@83.211.2.218:5060", nonce="4da47752b38cc32675ef46e0822052a663fe6204", response="75ca9ed6d073c971574c31cc6b0f5a64", qop=auth, cnonce="7030e2a6", nc=00000002 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Apr 12 18:01:03] DEBUG[8487]: app_dial.c:2713 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Apr 12 18:01:03] DEBUG[8487]: pbx.c:4752 __ast_pbx_run: Spawn extension (phones-sip,dialed-number,1) exited non-zero on 'SIP/153-0000000a' == Spawn extension (phones-sip, dialed-number, 1) exited non-zero on 'SIP/153-0000000a' [Apr 12 18:01:03] DEBUG[8487]: channel.c:2605 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/153-0000000a' [Apr 12 18:01:03] DEBUG[8487]: channel.c:2733 ast_hangup: Hanging up channel 'SIP/153-0000000a' [Apr 12 18:01:03] DEBUG[8487]: chan_sip.c:5827 sip_hangup: Hangup call SIP/153-0000000a, SIP callid ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg. [Apr 12 18:01:03] DEBUG[8487]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x1349808' asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK75fcfa4b From: "153" ;tag=as1220bb53 To: ;tag=14AD594C-1856 Date: Tue, 12 Apr 2011 16:01:03 GMT Call-ID: 676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 105 BYE <-------------> --- (9 headers 0 lines) --- [Apr 12 18:01:03] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060' of Request 105: Match Found Really destroying SIP dialog '676ab3ba3dd9065546d9d8183d966dac@asterisk-public-ip:5060' Method: INVITE [Apr 12 18:01:03] DEBUG[8401]: rtp_engine.c:292 instance_destructor: Destroyed RTP instance '0x13807b8' asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> <-------------> asterisk*CLI> [Apr 12 18:01:08] DEBUG[8401]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 47b97a090ebd905a191f73ea222d85ea@127.0.1.1:0 - OPTIONS (No RTP) [Apr 12 18:01:08] DEBUG[8401]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method OPTIONS - callid 6727e49d3550bcbe243e12e476b52e7b@asterisk-public-ip:5060 Reliably Transmitting (no NAT) to 192.168.1.151:5060: OPTIONS sip:152@192.168.1.151:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4f0ca126 Max-Forwards: 70 From: "asterisk" ;tag=as1f53e3d5 To: Contact: Call-ID: 6727e49d3550bcbe243e12e476b52e7b@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 16:01:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4f0ca126 From: "asterisk" ;tag=as1f53e3d5 To: ;tag=4229407113 Call-ID: 6727e49d3550bcbe243e12e476b52e7b@asterisk-public-ip:5060 CSeq: 102 OPTIONS Contact: "152" Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 asterisk*CLI> <-------------> --- (12 headers 0 lines) --- [Apr 12 18:01:08] DEBUG[8401]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '6727e49d3550bcbe243e12e476b52e7b@asterisk-public-ip:5060' of Request 102: Match Found Really destroying SIP dialog '6727e49d3550bcbe243e12e476b52e7b@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI> Really destroying SIP dialog 'ZDAyMmQ2ODI0YzBkOWVmMmFmYzMwNjAwNWU4OGIzODg.' Method: BYE [Apr 12 18:01:09] DEBUG[8401]: rtp_engine.c:292 instance_destructor: Destroyed RTP instance '0x1349808' asterisk*CLI> Disconnected from Asterisk server