<-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '55ae66469f7f@24.123.23.170' Method: REGISTER set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.93, port 5060 Audio is at 24.123.23.170 port 17286 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.93:5060: INVITE sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK42c538af From: ;tag=as68aa18d6 To: 528 ;tag=90493555b415b8be1762433957f5a495 Contact: Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 3998 4000 IN IP4 24.123.23.170 s=session c=IN IP4 24.123.23.170 t=0 0 m=audio 17286 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=recvonly --- Scheduling destruction of SIP dialog '079076751e0b5ba32413ad190c372a4a@24.123.23.170' in 32000 ms (Method: ACK) == Spawn extension (smvoice-sip, 204, 6) exited non-zero on 'SIP/526-b7720560' unifiedpaging*CLI> <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 200 OK To: 528 ;tag=90493555b415b8be1762433957f5a495 Contact: Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK42c538af From: ;tag=as68aa18d6 Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170 CSeq: 103 INVITE Content-Length: 241 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK v=0 o=- 3998 103875 IN IP4 192.168.1.93 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 30028 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16,17,18 a=silenceSupp:off - - - - a=ptime:20 a=sendonly <-------------> --- (10 headers 12 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.93, port 5060 Transmitting (no NAT) to 192.168.1.93:5060: ACK sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2668c44b From: ;tag=as68aa18d6 To: 528 ;tag=90493555b415b8be1762433957f5a495 Contact: Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.93, port 5060 Reliably Transmitting (no NAT) to 192.168.1.93:5060: BYE sip:528@192.168.1.93:5060 SIP/2.0 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2b453c9f From: ;tag=as68aa18d6 To: 528 ;tag=90493555b415b8be1762433957f5a495 Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '079076751e0b5ba32413ad190c372a4a@24.123.23.170' in 32000 ms (Method: ACK) Retransmitting #3 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx61d03b960da7358c2333efdaf0622dbe;received=192.168.1.93 From: 528 ;tag=f055e63c670dfe04684b84461a029087 To: ;tag=as68237d31 Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170 CSeq: 465812 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 3998 4000 IN IP4 192.168.1.91 s=session c=IN IP4 192.168.1.91 t=0 0 m=audio 10010 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 - - - - a=ptime:20 a=sendrecv --- unifiedpaging*CLI> <--- SIP read from 192.168.1.93:5060 ---> SIP/2.0 200 OK To: 528 ;tag=90493555b415b8be1762433957f5a495 Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2b453c9f From: ;tag=as68aa18d6 Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170 CSeq: 104 BYE Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '079076751e0b5ba32413ad190c372a4a@24.123.23.170' Method: ACK Retransmitting #4 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx61d03b960da7358c2333efdaf0622dbe;received=192.168.1.93 From: 528 ;tag=f055e63c670dfe04684b84461a029087 To: ;tag=as68237d31 Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170 CSeq: 465812 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 3998 4000 IN IP4 192.168.1.91 s=session c=IN IP4 192.168.1.91 t=0 0 m=audio 10010 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 - - - - a=ptime:20 a=sendrecv --- Retransmitting #5 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx61d03b960da7358c2333efdaf0622dbe;received=192.168.1.93 From: 528 ;tag=f055e63c670dfe04684b84461a029087 To: ;tag=as68237d31 Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170 CSeq: 465812 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 3998 4000 IN IP4 192.168.1.91 s=session c=IN IP4 192.168.1.91 t=0 0 m=audio 10010 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 - - - - a=ptime:20 a=sendrecv --- [Feb 15 16:41:49] NOTICE[4029]: chan_sip.c:7055 sip_reregister: -- Re-registration for 3173241052@sip.broadvoice.com [Feb 15 16:41:49] NOTICE[4029]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) Retransmitting #6 (no NAT) to 192.168.1.93:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx61d03b960da7358c2333efdaf0622dbe;received=192.168.1.93 From: 528 ;tag=f055e63c670dfe04684b84461a029087 To: ;tag=as68237d31 Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170 CSeq: 465812 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 3998 4000 IN IP4 192.168.1.91 s=session c=IN IP4 192.168.1.91 t=0 0 m=audio 10010 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 - - - - a=ptime:20 a=sendrecv --- [Feb 15 16:41:54] WARNING[4029]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission 386a40f7388a7a353e329d8124252dda@24.123.23.170 for seqno 465812 (Critical Response) [Feb 15 16:41:54] WARNING[4029]: chan_sip.c:1898 retrans_pkt: Hanging up call 386a40f7388a7a353e329d8124252dda@24.123.23.170 - no reply to our critical packet. == Spawn extension (smvoice-sip, 204, 6) exited non-zero on 'SIP/550-b7734378' Really destroying SIP dialog '386a40f7388a7a353e329d8124252dda@24.123.23.170' Method: ACK unifiedpaging*CLI> <--- SIP read from 192.168.1.93:5060 ---> BYE sip:550@24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKvdfc86f6d4ca22066c8a692b67e2c3c12 Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170 CSeq: 465813 BYE From: 528 ;tag=f055e63c670dfe04684b84461a029087 To: ;tag=as68237d31 User-Agent: Uniden SIP Phone p2 Ver BS4.77 Max-Forwards: 70 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- Transmitting (no NAT) to 192.168.1.93:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKvdfc86f6d4ca22066c8a692b67e2c3c12;received=192.168.1.93 From: 528 ;tag=f055e63c670dfe04684b84461a029087 To: ;tag=as68237d31 Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170 CSeq: 465813 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> unifiedpaging*CLI> <--- SIP read from 192.168.1.93:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKd79f74b3cb3cbbf05a003f0e07299adab Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 37380 REGISTER From: ;tag=94d2aaa50f89390af2cf3ace3ef31f76 To: Contact: ;expires=60;q=0.50 User-Agent: Uniden SIP Phone p2 Ver BS4.77 Max-Forwards: 70 Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK Authorization: Digest realm="asterisk", nonce="7273933d", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="b81b38903c6cb734d1cb18f7e251e60a" <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.93 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.93:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKd79f74b3cb3cbbf05a003f0e07299adab;received=192.168.1.93 From: ;tag=94d2aaa50f89390af2cf3ace3ef31f76 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 37380 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> unifiedpaging*CLI> <--- Transmitting (no NAT) to 192.168.1.93:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKd79f74b3cb3cbbf05a003f0e07299adab;received=192.168.1.93 From: ;tag=94d2aaa50f89390af2cf3ace3ef31f76 To: ;tag=as0e0dd9da Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 37380 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="21a1c665" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER) unifiedpaging*CLI> <--- SIP read from 192.168.1.93:5060 ---> REGISTER sip:24.123.23.170 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKua02f3e059cfd16f0ca668dfc24bc03f6 CSeq: 37381 REGISTER Call-ID: 55ae66469f7f@24.123.23.170 From: ;tag=94d2aaa50f89390af2cf3ace3ef31f76 To: Contact: ;expires=60;q=0.50 User-Agent: Uniden SIP Phone p2 Ver BS4.77 Max-Forwards: 70 Content-Length: 0 Supported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK Authorization: Digest realm="asterisk", nonce="21a1c665", algorithm=MD5, uri="sip:24.123.23.170", username="528", response="5606d9374716cefc019277ae5e024a07" <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.93 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.93:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKua02f3e059cfd16f0ca668dfc24bc03f6;received=192.168.1.93 From: ;tag=94d2aaa50f89390af2cf3ace3ef31f76 To: Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 37381 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> unifiedpaging*CLI> <--- Transmitting (no NAT) to 192.168.1.93:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKua02f3e059cfd16f0ca668dfc24bc03f6;received=192.168.1.93 From: ;tag=94d2aaa50f89390af2cf3ace3ef31f76 To: ;tag=as0e0dd9da Call-ID: 55ae66469f7f@24.123.23.170 CSeq: 37381 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Thu, 15 Feb 2007 21:42:05 GMT Content-Length: 0