== Parsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-HEAD-06/01/04-12:56:14, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-HEAD-06/01/04-12:56:14 currently running on asteriskpbx (pid = 13417) asteriskpbx*CLI> sip debug asteriskpbx*CLI> SIP Debugging Enabled asteriskpbx*CLI> set verbose 0 ################ Pickup phone, Line M1 selected, dial 123 ################ Sip read: INVITE sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C2D27 Session-Expires: 1800 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1325 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK,INFO Supported: 100rel,timer User-Agent: ACT P103SLD 02.07 Content-Type: application/sdp Content-Length: 255 v=0 o=username 278200477 278200477 IN IP4 192.168.16.8 s=ACT P103SLD 02.07 c=IN IP4 192.168.16.8 t=0 0 m=audio 41000 RTP/AVP 0 18 4 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtcp:41001 a=direction:both a=sendrecv 14 headers, 12 lines Using latest request as basis request asteriskpbx*CLI> Sending to 192.168.16.8 : 5060 (non-NAT) Found RTP audio format 0 asteriskpbx*CLI> Found RTP audio format 18 Found RTP audio format 4 asteriskpbx*CLI> Peer RTP is at port 192.168.16.8:0 Found description format PCMU Found description format G729 Found description format G723 Capabilities: us - 0x8040e(GSM|ULAW|ALAW|ILBC|H263), peer - audio=0x105(G723|ULAW|G729A)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) asteriskpbx*CLI> Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C2D27 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as2b735f2d Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1325 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="33dd35a3" Content-Length: 0 to 192.168.16.8:5060 asteriskpbx*CLI> Scheduling destruction of call 'CALL_ID12_00D0E9013F27_T7089446@192.168.16.8' in 15000 ms Found user '2008' asteriskpbx*CLI> Sip read: ACK sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C2D27 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as2b735f2d Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1325 ACK Contact: Max-Forwards: 70 Content-Length: 0 asteriskpbx*CLI> 9 headers, 0 lines asteriskpbx*CLI> Sip read: INVITE sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C2D3E Session-Expires: 1800 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1326 INVITE Proxy-Authorization: Digest username="2008", realm="asterisk", nonce="33dd35a3", opaque="", uri="sip:123@192.168.16.254", response="4ea8ffeaae9d63c93e703bbf0e0ad039" Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK,INFO Supported: 100rel,timer User-Agent: ACT P103SLD 02.07 Content-Type: application/sdp Content-Length: 255 v=0 o=username 278200477 278200477 IN IP4 192.168.16.8 s=ACT P103SLD 02.07 c=IN IP4 192.168.16.8 t=0 0 m=audio 41000 RTP/AVP 0 18 4 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtcp:41001 a=direction:both a=sendrecv asteriskpbx*CLI> 15 headers, 12 lines Using latest request as basis request Sending to 192.168.16.8 : 5060 (non-NAT) Found RTP audio format 0 asteriskpbx*CLI> Found RTP audio format 18 Found RTP audio format 4 asteriskpbx*CLI> Peer RTP is at port 192.168.16.8:0 Found description format PCMU asteriskpbx*CLI> Found description format G729 asteriskpbx*CLI> Found description format G723 asteriskpbx*CLI> Capabilities: us - 0x8040e(GSM|ULAW|ALAW|ILBC|H263), peer - audio=0x105(G723|ULAW|G729A)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) asteriskpbx*CLI> Found user '2008' Looking for 123 in ctx-mobile asteriskpbx*CLI> list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C2D3E From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1326 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.16.8:5060 We're at 192.168.16.254 port 18160 asteriskpbx*CLI> Answering with preferred capability 0x2(GSM) Answering with preferred capability 0x400(ILBC) Answering with capability 0x4(ULAW) asteriskpbx*CLI> Answering with capability 0x8(ALAW) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C2D3E From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1326 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 237 vasteriskpbx*CLI> =0 o=root 13417 13417 IN IP4 192.168.16.254 s=session c=IN IP4 192.168.16.254 t=0 0 m=audio 18160 RTP/AVP 3 97 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - to 192.168.16.8:5060 asteriskpbx*CLI> Sip read: ACK sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C2D5A From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1326 ACK Contact: Max-Forwards: 70 Content-Length: 0 asteriskpbx*CLI> 9 headers, 0 lines ########### Press M2 (line 2) on handset This will HOLD M1 (line 1) ########### Sip read: INVITE sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C44A4 Session-Expires: 1800 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1327 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK,INFO Supported: 100rel,timer User-Agent: ACT P103SLD 02.07 Content-Type: application/sdp Content-Length: 218 v=0 o=username 278200477 278200478 IN IP4 192.168.16.8 s=ACT P103SLD 02.07 c=IN IP4 0.0.0.0 t=0 0 m=audio 41000 RTP/AVP 0 18 4 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=sendonly 14 headers, 10 lines asteriskpbx*CLI> Using latest request as basis request Sending to 192.168.16.8 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Peer RTP is at port 0.0.0.0:0 Found description format PCMU Found description format G729 Found description format G723 Capabilities: us - 0x8040e(GSM|ULAW|ALAW|ILBC|H263), peer - audio=0x105(G723|ULAW|G729A)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) asteriskpbx*CLI> We're at 192.168.16.254 port 18160 Answering with capability 0x4(ULAW) asteriskpbx*CLI> Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C44A4 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1327 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 164 v=0 o=root 13417 13418 IN IP4 192.168.16.254 s=session c=IN IP4 192.168.16.254 t=0 0 m=audio 18160 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - to 192.168.16.8:5060 asteriskpbx*CLI> Sip read: ACK sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C450D From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1327 ACK Contact: Max-Forwards: 70 Content-Length: 0 asteriskpbx*CLI> 9 headers, 0 lines ######### Dial 1701 on M2 (new line) ######### Sip read: INVITE sip:1701@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C5A53 Session-Expires: 1800 From: "phoneH" ;tag=00D0E9013F27_T7101010 To: Call-ID: CALL_ID13_00D0E9013F27_T7101010@192.168.16.8 CSeq: 1328 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK,INFO Supported: 100rel,timer User-Agent: ACT P103SLD 02.07 Content-Type: application/sdp Content-Length: 255 v=0 o=username 278257377 278257377 IN IP4 192.168.16.8 s=ACT P103SLD 02.07 c=IN IP4 192.168.16.8 t=0 0 m=audio 41000 RTP/AVP 0 18 4 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtcp:41001 a=direction:both a=sendrecv 14 headers, 12 lines Using latest request as basis request asteriskpbx*CLI> Sending to 192.168.16.8 : 5060 (non-NAT) asteriskpbx*CLI> Found RTP audio format 0 asteriskpbx*CLI> Found RTP audio format 18 asteriskpbx*CLI> Found RTP audio format 4 Peer RTP is at port 192.168.16.8:0 asteriskpbx*CLI> Found description format PCMU asteriskpbx*CLI> Found description format G729 asteriskpbx*CLI> Found description format G723 asteriskpbx*CLI> Capabilities: us - 0x8040e(GSM|ULAW|ALAW|ILBC|H263), peer - audio=0x105(G723|ULAW|G729A)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) asteriskpbx*CLI> Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C5A53 From: "phoneH" ;tag=00D0E9013F27_T7101010 To: ;tag=as0785d4a9 Call-ID: CALL_ID13_00D0E9013F27_T7101010@192.168.16.8 CSeq: 1328 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="7897d9bb" asteriskpbx*CLI> Content-Length: 0 to 192.168.16.8:5060 Scheduling destruction of call 'CALL_ID13_00D0E9013F27_T7101010@192.168.16.8' in 15000 ms Found user '2008' asteriskpbx*CLI> Sip read: ACK sip:1701@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C5A53 From: "phoneH" ;tag=00D0E9013F27_T7101010 To: ;tag=as0785d4a9 Call-ID: CALL_ID13_00D0E9013F27_T7101010@192.168.16.8 CSeq: 1328 ACK Contact: Max-Forwards: 70 Content-Length: 0 asteriskpbx*CLI> 9 headers, 0 lines asteriskpbx*CLI> Sip read: INVITE sip:1701@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C5A69 Session-Expires: 1800 From: "phoneH" ;tag=00D0E9013F27_T7101010 To: Call-ID: CALL_ID13_00D0E9013F27_T7101010@192.168.16.8 CSeq: 1329 INVITE Proxy-Authorization: Digest username="2008", realm="asterisk", nonce="7897d9bb", opaque="", uri="sip:1701@192.168.16.254", response="5898214c909a4451556db2bee7bc0244" Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK,INFO Supported: 100rel,timer User-Agent: ACT P103SLD 02.07 Content-Type: application/sdp Content-Length: 255 v=0 o=username 278257377 278257377 IN IP4 192.168.16.8 s=ACT P103SLD 02.07 c=IN IP4 192.168.16.8 t=0 0 m=audio 41000 RTP/AVP 0 18 4 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtcp:41001 a=direction:both a=sendrecv asteriskpbx*CLI> 15 headers, 12 lines Using latest request as basis request asteriskpbx*CLI> Sending to 192.168.16.8 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Peer RTP is at port 192.168.16.8:0 Found description format PCMU Found description format G729 Found description format G723 Capabilities: us - 0x8040e(GSM|ULAW|ALAW|ILBC|H263), peer - audio=0x105(G723|ULAW|G729A)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) asteriskpbx*CLI> Found user '2008' Looking for 1701 in ctx-mobile list_route: hop: asteriskpbx*CLI> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C5A69 From: "phoneH" ;tag=00D0E9013F27_T7101010 To: ;tag=as58c37c69 Call-ID: CALL_ID13_00D0E9013F27_T7101010@192.168.16.8 CSeq: 1329 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.16.8:5060 asteriskpbx*CLI> We're at 192.168.16.254 port 15118 asteriskpbx*CLI> Answering with preferred capability 0x2(GSM) Answering with preferred capability 0x400(ILBC) Answering with capability 0x4(ULAW) asteriskpbx*CLI> Answering with capability 0x8(ALAW) asteriskpbx*CLI> Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C5A69 From: "phoneH" ;tag=00D0E9013F27_T7101010 To: ;tag=as58c37c69 Call-ID: CALL_ID13_00D0E9013F27_T7101010@192.168.16.8 CSeq: 1329 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 237 v=0 o=root 13417 13417 IN IP4 192.168.16.254 s=session c=IN IP4 192.168.16.254 t=0 0 m=audio 15118 RTP/AVP 3 97 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - to 192.168.16.8:5060 asteriskpbx*CLI> Sip read: ACK sip:1701@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C5A84 From: "phoneH" ;tag=00D0E9013F27_T7101010 To: ;tag=as58c37c69 Call-ID: CALL_ID13_00D0E9013F27_T7101010@192.168.16.8 CSeq: 1329 ACK Contact: Max-Forwards: 70 Content-Length: 0 asteriskpbx*CLI> 9 headers, 0 lines ########### Hangup phone Line M2 will clear, Line M1 still flashing ########### Sip read: BYE sip:1701@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C723C From: "phoneH" ;tag=00D0E9013F27_T7101010 To: ;tag=as58c37c69 Call-ID: CALL_ID13_00D0E9013F27_T7101010@192.168.16.8 CSeq: 1330 BYE Contact: Max-Forwards: 70 Content-Length: 0 asteriskpbx*CLI> 9 headers, 0 lines asteriskpbx*CLI> Sending to 192.168.16.8 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C723C From: "phoneH" ;tag=00D0E9013F27_T7101010 To: ;tag=as58c37c69 Call-ID: CALL_ID13_00D0E9013F27_T7101010@192.168.16.8 CSeq: 1330 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.16.8:5060 asteriskpbx*CLI> Destroying call 'CALL_ID13_00D0E9013F27_T7101010@192.168.16.8' ######### Pickup phone. Try and collect held call on M1 (line 1) by pressing M1 ######### Sip read: INVITE sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C9991 Session-Expires: 1800 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1331 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK,INFO Supported: 100rel,timer User-Agent: ACT P103SLD 02.07 Content-Type: application/sdp Content-Length: 169 v=0 o=username 278200477 278200479 IN IP4 192.168.16.8 s=ACT P103SLD 02.07 c=IN IP4 192.168.16.8 t=0 0 m=audio 41000 RTP/AVP 0 a=rtpmap:0 PCMU/8000/1 a=sendrecv 14 headers, 8 lines asteriskpbx*CLI> Sip read: INVITE sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C9991 Session-Expires: 1800 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1331 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK,INFO Supported: 100rel,timer User-Agent: ACT P103SLD 02.07 Content-Type: application/sdp Content-Length: 169 v=0 o=username 278200477 278200479 IN IP4 192.168.16.8 s=ACT P103SLD 02.07 c=IN IP4 192.168.16.8 t=0 0 m=audio 41000 RTP/AVP 0 a=rtpmap:0 PCMU/8000/1 a=sendrecv 14 headers, 8 lines asteriskpbx*CLI> Sip read: REGISTER sip:192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.192:5060;branch=z9hG4bK_00D0E9013F25_T0D4118B2 Max-Forwards: 70 From: ;tag=00D0E9013F25_T222369971 To: Call-ID: REGISTER_00D0E9013F25_T45056270@192.168.16.192 CSeq: 1973 REGISTER Contact: Expires: 3600 User-Agent: ACT P103SLD 02.07 Accept: application/nat-info Content-Length: 0 12 headers, 0 lines asteriskpbx*CLI> Using latest request as basis request Sending to 192.168.16.192 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.192:5060;branch=z9hG4bK_00D0E9013F25_T0D4118B2 From: ;tag=00D0E9013F25_T222369971 To: ;tag=as19bfd537 Call-ID: REGISTER_00D0E9013F25_T45056270@192.168.16.192 CSeq: 1973 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.16.192:5060 asteriskpbx*CLI> Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.16.192:5060;branch=z9hG4bK_00D0E9013F25_T0D4118B2 From: ;tag=00D0E9013F25_T222369971 To: ;tag=as19bfd537 Call-ID: REGISTER_00D0E9013F25_T45056270@192.168.16.192 CSeq: 1973 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="330d2198" Content-Length: 0 to 192.168.16.192:5060 asteriskpbx*CLI> Scheduling destruction of call 'REGISTER_00D0E9013F25_T45056270@192.168.16.192' in 15000 ms asteriskpbx*CLI> Sip read: REGISTER sip:192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.192:5060;branch=z9hG4bK_00D0E9013F25_T0D4118C6 Max-Forwards: 70 From: ;tag=00D0E9013F25_T222369991 To: Call-ID: REGISTER_00D0E9013F25_T45056270@192.168.16.192 CSeq: 1974 REGISTER Contact: Expires: 3600 User-Agent: ACT P103SLD 02.07 Proxy-Authorization: Digest username="2192", realm="asterisk", nonce="330d2198", algorithm=md5, opaque="", uri="sip:192.168.16.254", response="686e3ff75d195254a0afaedaf1d386cb" Accept: application/nat-info Content-Length: 0 asteriskpbx*CLI> 13 headers, 0 lines Using latest request as basis request Sending to 192.168.16.192 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.192:5060;branch=z9hG4bK_00D0E9013F25_T0D4118C6 From: ;tag=00D0E9013F25_T222369991 To: ;tag=as19bfd537 Call-ID: REGISTER_00D0E9013F25_T45056270@192.168.16.192 CSeq: 1974 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.16.192:5060 asteriskpbx*CLI> Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.192:5060;branch=z9hG4bK_00D0E9013F25_T0D4118C6 From: ;tag=00D0E9013F25_T222369991 To: ;tag=as19bfd537 Call-ID: REGISTER_00D0E9013F25_T45056270@192.168.16.192 CSeq: 1974 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: ;expires=3600 Date: Tue, 01 Jun 2004 13:34:33 GMT Content-Length: 0 to 192.168.16.192:5060 asteriskpbx*CLI> Scheduling destruction of call 'REGISTER_00D0E9013F25_T45056270@192.168.16.192' in 15000 ms asteriskpbx*CLI> Sip read: INVITE sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C9991 Session-Expires: 1800 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1331 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK,INFO Supported: 100rel,timer User-Agent: ACT P103SLD 02.07 Content-Type: application/sdp Content-Length: 169 v=0 o=username 278200477 278200479 IN IP4 192.168.16.8 s=ACT P103SLD 02.07 c=IN IP4 192.168.16.8 t=0 0 m=audio 41000 RTP/AVP 0 a=rtpmap:0 PCMU/8000/1 a=sendrecv asteriskpbx*CLI> 14 headers, 8 lines asteriskpbx*CLI> Sip read: INVITE sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006C9991 Session-Expires: 1800 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1331 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,PRACK,INFO Supported: 100rel,timer User-Agent: ACT P103SLD 02.07 Content-Type: application/sdp Content-Length: 169 v=0 o=username 278200477 278200479 IN IP4 192.168.16.8 s=ACT P103SLD 02.07 c=IN IP4 192.168.16.8 t=0 0 m=audio 41000 RTP/AVP 0 a=rtpmap:0 PCMU/8000/1 a=sendrecv 14 headers, 8 lines asteriskpbx*CLI> Sip read: BYE sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006CB113 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1332 BYE Contact: Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines asteriskpbx*CLI> Sip read: BYE sip:123@192.168.16.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK_00D0E9013F27_T006CB113 From: "phoneH" ;tag=00D0E9013F27_T7089446 To: ;tag=as645aed84 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 1332 BYE Contact: Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines asteriskpbx*CLI> Destroying call 'REGISTER_00D0E9013F25_T45056270@192.168.16.192' ############### After waiting for retries, handset finally gets Busy signal then hangs up. Hangup handset ############### ############### However, the application is still going strong ############### asteriskpbx*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/2008-926b (ctx-mobile 123 3 ) Up SayUnixTime |GB|'beep' ABd 'digits/at' IMp 1 active channel(s) asteriskpbx*CLI> soft hangup SIP/2008-926b Requested Hangup on channel 'SIP/2008-926b' set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.16.8, port 5060 Reliably Transmitting: BYE sip:2008@192.168.16.8:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.254:5060;branch=z9hG4bK060532eb From: ;tag=as645aed84 To: "phoneH" ;tag=00D0E9013F27_T7089446 Contact: Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.16.8:5060 asteriskpbx*CLI> Sip read: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.16.254:5060;branch=z9hG4bK060532eb From: ;tag=as645aed84 To: "phoneH" ;tag=00D0E9013F27_T7089446 Call-ID: CALL_ID12_00D0E9013F27_T7089446@192.168.16.8 CSeq: 102 BYE Content-Length: 0 7 headers, 0 lines Message is BYE Destroying call 'CALL_ID12_00D0E9013F27_T7089446@192.168.16.8' asteriskpbx*CLI> exit Executing last minute cleanups