<--- SIP read from UDP://64.61.93.190:5060 ---> INVITE sip:15184520000@207.255.56.000:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bK9c4.4c428095.0 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bK9c4.f2721da5.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK27c0a73d;rport=5060 From: "Harrisburg PA" ;tag=as735bd392 To: Contact: Call-ID: 680b58b608df319d488d8d6e21437bce@64.61.93.170 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 68 Remote-Party-ID: "Harrisburg PA" ;privacy=off;screen=no Date: Fri, 18 Sep 2009 17:02:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 410 v=0 o=root 29667 29667 IN IP4 64.61.93.170 s=session c=IN IP4 64.61.93.170 t=0 0 m=audio 10652 RTP/AVP 0 8 3 97 111 5 7 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:7 LPC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (18 headers 19 lines) --- == Using SIP RTP CoS mark 5 Sending to 64.61.93.190 : 5060 (no NAT) Using INVITE request as basis request - 680b58b608df319d488d8d6e21437bce@64.61.93.170 No user '7177240000' in SIP users list Found peer 'voicepulse-primary' for '7177240000' from 64.61.93.190:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 111 Found RTP audio format 5 Found RTP audio format 7 Found RTP audio format 101 Peer audio RTP is at port 64.61.93.170:10652 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 iLBC for ID 97 Found audio description format G726-32 for ID 111 Found audio description format DVI4 for ID 5 Found audio description format LPC for ID 7 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xcae (gsm|ulaw|alaw|g726|adpcm|lpc10|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 64.61.93.170:10652 Looking for 15184520000 in trunkhandler_voicepulse (domain 207.255.56.000) list_route: hop: bob*CLI> <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bK9c4.4c428095.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bK9c4.f2721da5.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK27c0a73d;rport=5060 Record-Route: From: "Harrisburg PA" ;tag=as735bd392 To: Call-ID: 680b58b608df319d488d8d6e21437bce@64.61.93.170 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.16-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [15184520000@trunkhandler_voicepulse:1] NoOp("SIP/voicepulse-primary-08497470", "CALL FROM: Harrisburg PA <7177240000> To: 15184520000") in new stack -- Executing [15184520000@trunkhandler_voicepulse:2] Dial("SIP/voicepulse-primary-08497470", "SIP/voicepulse-primary/15188780000,45,r") in new stack == Using SIP RTP CoS mark 5 Audio is at 207.255.56.000 port 16834 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 64.61.93.190:5060: INVITE sip:15188780000@jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK67ba22cd;rport Max-Forwards: 70 From: "Harrisburg PA" ;tag=as0d776d0e To: Contact: Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.16-rc1 Remote-Party-ID: "Harrisburg PA" ;privacy=off;screen=no Date: Fri, 18 Sep 2009 17:02:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 317 v=0 o=root 674220992 674220992 IN IP4 207.255.56.000 s=Asterisk PBX 1.6.0.16-rc1 c=IN IP4 207.255.56.000 t=0 0 m=audio 16834 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/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 --- -- Called voicepulse-primary/15188780000 bob*CLI> <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bK9c4.4c428095.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bK9c4.f2721da5.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK27c0a73d;rport=5060 Record-Route: From: "Harrisburg PA" ;tag=as735bd392 To: ;tag=as7165db23 Call-ID: 680b58b608df319d488d8d6e21437bce@64.61.93.170 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.16-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK67ba22cd;rport=5060 From: "Harrisburg PA" ;tag=as0d776d0e To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.d1df Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="207.255.56.000", nonce="4ab3be6d852d82e43e4b5e8ecb9afcbb766c033a", qop="auth" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 64.61.93.190:5060: ACK sip:15188780000@jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK67ba22cd;rport Max-Forwards: 70 From: "Harrisburg PA" ;tag=as0d776d0e To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.d1df Contact: Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.16-rc1 Remote-Party-ID: "Harrisburg PA" ;privacy=off;screen=no Content-Length: 0 --- Audio is at 207.255.56.000 port 16834 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 64.61.93.190:5060: INVITE sip:15188780000@jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK7e7d5fb8;rport Max-Forwards: 70 From: "Harrisburg PA" ;tag=as0d776d0e To: Contact: Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0.16-rc1 Remote-Party-ID: "Harrisburg PA" ;privacy=off;screen=no Proxy-Authorization: Digest username="foo", realm="207.255.56.000", algorithm=MD5, uri="sip:15188780000@jfk-primary.voicepulse.com", nonce="4ab3be6d852d82e43e4b5e8ecb9afcbb766c033a", response="af456f463ed47c6499dc42ca09fd54dd", qop=auth, cnonce="75446a9a", nc=00000001 Date: Fri, 18 Sep 2009 17:02:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 317 v=0 o=root 674220992 674220993 IN IP4 207.255.56.000 s=Asterisk PBX 1.6.0.16-rc1 c=IN IP4 207.255.56.000 t=0 0 m=audio 16834 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/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 --- bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK7e7d5fb8;rport=5060 From: "Harrisburg PA" ;tag=as0d776d0e To: Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 103 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK7e7d5fb8;rport=5060 Record-Route: From: "Harrisburg PA" ;tag=as0d776d0e To: ;tag=as744877b6 Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 103 INVITE User-Agent: VoicePulse PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 22130 22130 IN IP4 64.61.93.188 s=session c=IN IP4 64.61.93.188 t=0 0 m=audio 17126 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 64.61.93.188:17126 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 telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 64.61.93.188:17126 -- SIP/voicepulse-primary-08469350 is making progress passing it to SIP/voicepulse-primary-08497470 bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK7e7d5fb8;rport=5060 Record-Route: From: "Harrisburg PA" ;tag=as0d776d0e To: ;tag=as744877b6 Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 103 INVITE User-Agent: VoicePulse PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 22130 22131 IN IP4 64.61.93.188 s=session c=IN IP4 64.61.93.188 t=0 0 m=audio 17126 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 64.61.93.188:17126 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 telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 64.61.93.188:17126 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 64.61.93.190, port 5060 Transmitting (no NAT) to 64.61.93.190:5060: ACK sip:+15188780000@64.61.93.188 SIP/2.0 Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK41401e83;rport Route: Max-Forwards: 70 From: "Harrisburg PA" ;tag=as0d776d0e To: ;tag=as744877b6 Contact: Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.0.16-rc1 Remote-Party-ID: "Harrisburg PA" ;privacy=off;screen=no Content-Length: 0 --- -- SIP/voicepulse-primary-08469350 answered SIP/voicepulse-primary-08497470 Audio is at 207.255.56.000 port 16550 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP bob*CLI> <--- Reliably Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bK9c4.4c428095.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bK9c4.f2721da5.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK27c0a73d;rport=5060 Record-Route: From: "Harrisburg PA" ;tag=as735bd392 To: ;tag=as7165db23 Call-ID: 680b58b608df319d488d8d6e21437bce@64.61.93.170 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.16-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 319 v=0 o=root 2000091937 2000091937 IN IP4 207.255.56.000 s=Asterisk PBX 1.6.0.16-rc1 c=IN IP4 207.255.56.000 t=0 0 m=audio 16550 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 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 <------------> -- Packet2Packet bridging SIP/voicepulse-primary-08497470 and SIP/voicepulse-primary-08469350 bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> ACK sip:15184520000@207.255.56.000:5060 SIP/2.0 Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bK9c4.4c428095.2 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bK9c4.f2721da5.2 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK6d7e9afb;rport=5060 From: "Harrisburg PA" ;tag=as735bd392 To: ;tag=as7165db23 Contact: Call-ID: 680b58b608df319d488d8d6e21437bce@64.61.93.170 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 68 Remote-Party-ID: "Harrisburg PA" ;privacy=off;screen=no Content-Length: 0 VP-hint: loose-routed <-------------> --- (14 headers 0 lines) --- bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> BYE sip:7177240000@207.255.56.000:5060 SIP/2.0 Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bK7e7bf3f7 Via: SIP/2.0/UDP 64.61.93.188:5060;branch=z9hG4bK7e7bf3f7;rport=5060 From: ;tag=as744877b6 To: "Harrisburg PA" ;tag=as0d776d0e Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 102 BYE User-Agent: VoicePulse PBX Max-Forwards: 69 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 VP-hint: loose-routed VP-hint: forwarded <-------------> --- (14 headers 0 lines) --- Sending to 64.61.93.190 : 5060 (no NAT) bob*CLI> <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bK7e7bf3f7;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.188:5060;branch=z9hG4bK7e7bf3f7;rport=5060 From: ;tag=as744877b6 To: "Harrisburg PA" ;tag=as0d776d0e Call-ID: 2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.0.16-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (trunkhandler_voicepulse, 15184520000, 2) exited non-zero on 'SIP/voicepulse-primary-08497470' Scheduling destruction of SIP dialog '680b58b608df319d488d8d6e21437bce@64.61.93.170' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 64.61.93.190, port 5060 Reliably Transmitting (no NAT) to 64.61.93.190:5060: BYE sip:7177240000@64.61.93.170 SIP/2.0 Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK7a3edece;rport Route: Max-Forwards: 70 From: ;tag=as7165db23 To: "Harrisburg PA" ;tag=as735bd392 Call-ID: 680b58b608df319d488d8d6e21437bce@64.61.93.170 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.0.16-rc1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Really destroying SIP dialog '2e9ba2da0d75ec3e284f1fb366d636d1@207.255.56.000' Method: BYE bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK7a3edece;rport=5060 From: ;tag=as7165db23 To: "Harrisburg PA" ;tag=as735bd392 Call-ID: 680b58b608df319d488d8d6e21437bce@64.61.93.170 CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '680b58b608df319d488d8d6e21437bce@64.61.93.170' Method: ACK Really destroying SIP dialog '4481df364be6c17959c9c9b761c91d60@127.0.0.1' Method: REGISTER Really destroying SIP dialog '68f3c19468ae7be44d80f876158b7ddf@127.0.0.1' Method: REGISTER [Sep 18 13:04:32] NOTICE[20154]: chan_sip.c:9816 sip_reregister: -- Re-registration for foo@jfk-primary.voicepulse.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 64.61.93.190:5060: REGISTER sip:jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK500db023;rport Max-Forwards: 70 From: ;tag=as5882a47a To: Call-ID: 4481df364be6c17959c9c9b761c91d60@127.0.0.1 CSeq: 106 REGISTER User-Agent: Asterisk PBX 1.6.0.16-rc1 Authorization: Digest username="foo", realm="jfk-primary.voicepulse.com", algorithm=MD5, uri="sip:jfk-primary.voicepulse.com", nonce="4ab3bd902c360e46df41c5586fee2c1cd6ba2d74", response="607a4dd07962236c862a234fd171b346", qop=auth, cnonce="023aa8cc", nc=00000004 Expires: 120 Contact: Event: registration Content-Length: 0 --- bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK500db023;rport=5060 From: ;tag=as5882a47a To: Call-ID: 4481df364be6c17959c9c9b761c91d60@127.0.0.1 CSeq: 106 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK500db023;rport=5060 From: ;tag=as5882a47a To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.0605 Call-ID: 4481df364be6c17959c9c9b761c91d60@127.0.0.1 CSeq: 106 REGISTER WWW-Authenticate: Digest realm="jfk-primary.voicepulse.com", nonce="4ab3becce4fc900e4aa25b1fe67060b5523d8bdd", qop="auth", stale=true Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Responding to challenge, registration to domain/host name jfk-primary.voicepulse.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 64.61.93.190:5060: REGISTER sip:jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK61097682;rport Max-Forwards: 70 From: ;tag=as4438564a To: Call-ID: 4481df364be6c17959c9c9b761c91d60@127.0.0.1 CSeq: 107 REGISTER User-Agent: Asterisk PBX 1.6.0.16-rc1 Authorization: Digest username="foo", realm="jfk-primary.voicepulse.com", algorithm=MD5, uri="sip:jfk-primary.voicepulse.com", nonce="4ab3becce4fc900e4aa25b1fe67060b5523d8bdd", response="cb423639f61fbfef33e8eedfc7210b96", qop=auth, cnonce="08fd1807", nc=00000001 Expires: 120 Contact: Event: registration Content-Length: 0 --- bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK61097682;rport=5060 From: ;tag=as4438564a To: Call-ID: 4481df364be6c17959c9c9b761c91d60@127.0.0.1 CSeq: 107 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- bob*CLI> <--- SIP read from UDP://64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.255.56.000:5060;branch=z9hG4bK61097682;rport=5060 From: ;tag=as4438564a To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.de3f Call-ID: 4481df364be6c17959c9c9b761c91d60@127.0.0.1 CSeq: 107 REGISTER Contact: ;expires=120;received="sip:207.255.56.000:5060" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '4481df364be6c17959c9c9b761c91d60@127.0.0.1' in 32000 ms (Method: REGISTER) [Sep 18 13:04:32] NOTICE[20154]: chan_sip.c:15981 handle_response_register: Outbound Registration: Expiry for jfk-primary.voicepulse.com is 120 sec (Scheduling reregistration in 105 s) [Sep 18 13:04:32] NOTICE[20154]: chan_sip.c:9816 sip_reregister: -- Re-registration for foo@jfk-backup.voicepulse.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 67.108.9.165:5060: REGISTER sip:jfk-backup.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 207.255.56.113:5060;branch=z9hG4bK667af78d;rport Max-Forwards: 70 From: ;tag=as43297000 To: Call-ID: 68f3c19468ae7be44d80f876158b7ddf@127.0.0.1 CSeq: 108 REGISTER User-Agent: Asterisk PBX 1.6.0.16-rc1 Authorization: Digest username="foo", realm="jfk-backup.voicepulse.com", algorithm=MD5, uri="sip:jfk-backup.voicepulse.com", nonce="4ab3bd4b000021586147080b336fb0a8fa980dcb401385ae", response="2803ac1c5738cee66f89d7a7d2e9eb4d", qop=auth, cnonce="34581301", nc=00000002 Expires: 120 Contact: Event: registration Content-Length: 0 --- bob*CLI> <--- SIP read from UDP://67.108.9.165:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 207.255.56.113:5060;branch=z9hG4bK667af78d;rport=5060 From: ;tag=as43297000 To: Call-ID: 68f3c19468ae7be44d80f876158b7ddf@127.0.0.1 CSeq: 108 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- bob*CLI> <--- SIP read from UDP://67.108.9.165:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 207.255.56.113:5060;branch=z9hG4bK667af78d;rport=5060 From: ;tag=as43297000 To: ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.c263 Call-ID: 68f3c19468ae7be44d80f876158b7ddf@127.0.0.1 CSeq: 108 REGISTER WWW-Authenticate: Digest realm="jfk-backup.voicepulse.com", nonce="4ab3bdb400002e01e536e462cfcada48c2ef93ada92c4f29", qop="auth", stale=true Server: OpenSIPS (1.4.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Responding to challenge, registration to domain/host name jfk-backup.voicepulse.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 67.108.9.165:5060: REGISTER sip:jfk-backup.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 207.255.56.113:5060;branch=z9hG4bK34dcd5a5;rport Max-Forwards: 70 From: ;tag=as60ff7b34 To: Call-ID: 68f3c19468ae7be44d80f876158b7ddf@127.0.0.1 CSeq: 109 REGISTER User-Agent: Asterisk PBX 1.6.0.16-rc1 Authorization: Digest username="foo", realm="jfk-backup.voicepulse.com", algorithm=MD5, uri="sip:jfk-backup.voicepulse.com", nonce="4ab3bdb400002e01e536e462cfcada48c2ef93ada92c4f29", response="3294407a0ca82f43b0f7e8b1f671334a", qop=auth, cnonce="716f8e11", nc=00000001 Expires: 120 Contact: Event: registration Content-Length: 0 --- bob*CLI> <--- SIP read from UDP://67.108.9.165:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 207.255.56.113:5060;branch=z9hG4bK34dcd5a5;rport=5060 From: ;tag=as60ff7b34 To: Call-ID: 68f3c19468ae7be44d80f876158b7ddf@127.0.0.1 CSeq: 109 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- bob*CLI> <--- SIP read from UDP://67.108.9.165:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.255.56.113:5060;branch=z9hG4bK34dcd5a5;rport=5060 From: ;tag=as60ff7b34 To: ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.bd64 Call-ID: 68f3c19468ae7be44d80f876158b7ddf@127.0.0.1 CSeq: 109 REGISTER Contact: ;expires=120;received="sip:207.255.56.113:5060" Server: OpenSIPS (1.4.2-notls (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '68f3c19468ae7be44d80f876158b7ddf@127.0.0.1' in 32000 ms (Method: REGISTER)