Sep 2 09:04:53 VERBOSE[114695]: Sip read: INVITE sip:2001@softins.softins.co.uk;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200;branch=z9hG4bK5cea4fb8cda4dec7 From: "Tony Mountifield" ;tag=b49c41a0c48e2d80 To: Contact: Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 19313 INVITE User-Agent: Grandstream BT100 1.0.5.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 341 v=0 o=2000 8000 8000 IN IP4 192.168.0.200 s=SIP Call c=IN IP4 192.168.0.200 t=0 0 m=audio 5004 RTP/AVP 98 8 0 18 2 15 9 4 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=ptime:20 Sep 2 09:04:53 VERBOSE[114695]: 12 headers, 16 lines Sep 2 09:04:53 DEBUG[114695]: Allocating new SIP call for 41d72bfad13725a4@192.168.0.200 Sep 2 09:04:53 VERBOSE[114695]: Using latest request as basis request Sep 2 09:04:53 VERBOSE[114695]: Sending to 192.168.0.200 : 5060 (non-NAT) Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 98 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 8 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 0 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 18 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 2 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 15 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 9 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 4 Sep 2 09:04:53 VERBOSE[114695]: Peer audio RTP is at port 192.168.0.200:5004 Sep 2 09:04:53 DEBUG[114695]: Peer audio RTP is at port 192.168.0.200:5004 Sep 2 09:04:53 VERBOSE[114695]: Found description format iLBC Sep 2 09:04:53 VERBOSE[114695]: Found description format PCMA Sep 2 09:04:53 VERBOSE[114695]: Found description format PCMU Sep 2 09:04:53 VERBOSE[114695]: Found description format G729 Sep 2 09:04:53 VERBOSE[114695]: Found description format G726-32 Sep 2 09:04:53 VERBOSE[114695]: Found description format G728 Sep 2 09:04:53 VERBOSE[114695]: Found description format G722 Sep 2 09:04:53 VERBOSE[114695]: Found description format G723 Sep 2 09:04:53 VERBOSE[114695]: Capabilities: us - 0x8040e(GSM|ULAW|ALAW|ILBC|H263), peer - audio=0x51d(G723|ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0x40c(ULAW|ALAW|ILBC) Sep 2 09:04:53 VERBOSE[114695]: Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x1(G723) Sep 2 09:04:53 DEBUG[114695]: Setting NAT on RTP to 0 Sep 2 09:04:53 VERBOSE[114695]: Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.200;branch=z9hG4bK5cea4fb8cda4dec7 From: "Tony Mountifield" ;tag=b49c41a0c48e2d80 To: ;tag=as5c5b4ca1 Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 19313 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="025a6cf5" Content-Length: 0 to 192.168.0.200:5060 Sep 2 09:04:53 VERBOSE[114695]: Scheduling destruction of call '41d72bfad13725a4@192.168.0.200' in 15000 ms Sep 2 09:04:53 VERBOSE[114695]: Found user '2000' Sep 2 09:04:53 VERBOSE[114695]: Sip read: ACK sip:2001@softins.softins.co.uk;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200;branch=z9hG4bK5cea4fb8cda4dec7 From: "Tony Mountifield" ;tag=b49c41a0c48e2d80 To: ;tag=as5c5b4ca1 Contact: Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 19313 ACK User-Agent: Grandstream BT100 1.0.5.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Sep 2 09:04:53 VERBOSE[114695]: 11 headers, 0 lines Sep 2 09:04:53 DEBUG[114695]: Stopping retransmission on '41d72bfad13725a4@192.168.0.200' of Response 19313: Found Sep 2 09:04:53 VERBOSE[114695]: Sip read: INVITE sip:2001@softins.softins.co.uk;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200;branch=z9hG4bKca965cb976fff083 From: "Tony Mountifield" ;tag=b49c41a0c48e2d80 To: Contact: Proxy-Authorization: DIGEST username="2000", realm="asterisk", algorithm=MD5, uri="sip:2001@softins.softins.co.uk;user=phone", nonce="025a6cf5", response="effd3303edefee571247443cbc3142f0" Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 19314 INVITE User-Agent: Grandstream BT100 1.0.5.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 341 v=0 o=2000 8000 8000 IN IP4 192.168.0.200 s=SIP Call c=IN IP4 192.168.0.200 t=0 0 m=audio 5004 RTP/AVP 98 8 0 18 2 15 9 4 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=ptime:20 Sep 2 09:04:53 VERBOSE[114695]: 13 headers, 16 lines Sep 2 09:04:53 VERBOSE[114695]: Using latest request as basis request Sep 2 09:04:53 VERBOSE[114695]: Sending to 192.168.0.200 : 5060 (non-NAT) Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 98 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 8 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 0 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 18 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 2 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 15 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 9 Sep 2 09:04:53 VERBOSE[114695]: Found RTP audio format 4 Sep 2 09:04:53 VERBOSE[114695]: Peer audio RTP is at port 192.168.0.200:5004 Sep 2 09:04:53 DEBUG[114695]: Peer audio RTP is at port 192.168.0.200:5004 Sep 2 09:04:53 VERBOSE[114695]: Found description format iLBC Sep 2 09:04:53 VERBOSE[114695]: Found description format PCMA Sep 2 09:04:53 VERBOSE[114695]: Found description format PCMU Sep 2 09:04:53 VERBOSE[114695]: Found description format G729 Sep 2 09:04:53 VERBOSE[114695]: Found description format G726-32 Sep 2 09:04:53 VERBOSE[114695]: Found description format G728 Sep 2 09:04:53 VERBOSE[114695]: Found description format G722 Sep 2 09:04:53 VERBOSE[114695]: Found description format G723 Sep 2 09:04:53 VERBOSE[114695]: Capabilities: us - 0x8040e(GSM|ULAW|ALAW|ILBC|H263), peer - audio=0x51d(G723|ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0x40c(ULAW|ALAW|ILBC) Sep 2 09:04:53 VERBOSE[114695]: Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x1(G723) Sep 2 09:04:53 DEBUG[114695]: Setting NAT on RTP to 0 Sep 2 09:04:53 VERBOSE[114695]: Found user '2000' Sep 2 09:04:53 DEBUG[114695]: Check for res for 2000 Sep 2 09:04:53 DEBUG[114695]: Call from user '2000' is 1 out of 0 Sep 2 09:04:53 VERBOSE[114695]: Looking for 2001 in from-sip-internal Sep 2 09:04:53 DEBUG[114695]: build_route: Contact hop: Sep 2 09:04:53 VERBOSE[114695]: list_route: hop: Sep 2 09:04:53 VERBOSE[114695]: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200;branch=z9hG4bKca965cb976fff083 From: "Tony Mountifield" ;tag=b49c41a0c48e2d80 To: ;tag=as63d94195 Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 19314 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.200:5060 Sep 2 09:04:53 DEBUG[294930]: Launching 'Dial' Sep 2 09:04:53 DEBUG[294930]: Allocating new SIP call for (null) Sep 2 09:04:53 DEBUG[294930]: Setting NAT on RTP to 0 Sep 2 09:04:53 DEBUG[294930]: Outgoing Call for 2001 Sep 2 09:04:53 DEBUG[294930]: Call from user '2001' is 1 out of 0 Sep 2 09:04:53 VERBOSE[294930]: We're at 192.168.0.1 port 13716 Sep 2 09:04:53 VERBOSE[294930]: Answering/Requesting with root capability 8 Sep 2 09:04:53 VERBOSE[294930]: Answering with preferred capability 0x4(ULAW) Sep 2 09:04:53 VERBOSE[294930]: Answering with preferred capability 0x400(ILBC) Sep 2 09:04:53 VERBOSE[294930]: Answering with preferred capability 0x2(GSM) Sep 2 09:04:53 VERBOSE[294930]: 12 headers, 11 lines Sep 2 09:04:53 VERBOSE[294930]: Reliably Transmitting: INVITE sip:2001@192.168.0.201;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK44d62a30 From: "2000" ;tag=as15777eb1 To: Contact: Call-ID: 25754e603557407a5d7415f25d65645e@192.168.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 02 Sep 2004 08:04:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 231 v=0 o=root 13712 13712 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 13716 RTP/AVP 8 0 97 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - (no NAT) to 192.168.0.201:5060 Sep 2 09:04:53 VERBOSE[294930]: -- Called 2001 Sep 2 09:04:53 DEBUG[294930]: Set channel SIP/2001-de61 to read format ALAW Sep 2 09:04:53 DEBUG[294930]: Set channel SIP/2000-b5bf to write format ALAW Sep 2 09:04:53 DEBUG[294930]: Set channel SIP/2001-de61 to write format ALAW Sep 2 09:04:53 DEBUG[294930]: Set channel SIP/2000-b5bf to read format ALAW Sep 2 09:04:53 VERBOSE[114695]: Sip read: SIP/2.0 100 trying Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK44d62a30 From: "2000" ;tag=as15777eb1 To: Call-ID: 25754e603557407a5d7415f25d65645e@192.168.0.1 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.9 Content-Length: 0 Sep 2 09:04:53 VERBOSE[114695]: 8 headers, 0 lines Sep 2 09:04:53 DEBUG[114695]: (Provisional) Stopping retransmission (but retaining packet) on '25754e603557407a5d7415f25d65645e@192.168.0.1' Request 102: Found Sep 2 09:04:53 VERBOSE[114695]: Sip read: SIP/2.0 180 ringing Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK44d62a30 From: "2000" ;tag=as15777eb1 To: ;tag=877729714034fadd Call-ID: 25754e603557407a5d7415f25d65645e@192.168.0.1 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.9 Content-Length: 0 Sep 2 09:04:53 VERBOSE[114695]: 8 headers, 0 lines Sep 2 09:04:53 DEBUG[114695]: (Provisional) Stopping retransmission (but retaining packet) on '25754e603557407a5d7415f25d65645e@192.168.0.1' Request 102: Found Sep 2 09:04:53 VERBOSE[294930]: -- SIP/2001-de61 is ringing Sep 2 09:04:53 VERBOSE[294930]: Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.200;branch=z9hG4bKca965cb976fff083 From: "Tony Mountifield" ;tag=b49c41a0c48e2d80 To: ;tag=as63d94195 Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 19314 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.200:5060 Sep 2 09:04:55 VERBOSE[114695]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK44d62a30 From: "2000" ;tag=as15777eb1 To: ;tag=877729714034fadd Call-ID: 25754e603557407a5d7415f25d65645e@192.168.0.1 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.9 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 145 v=0 o=2001 8000 8000 IN IP4 192.168.0.201 s=SIP Call c=IN IP4 192.168.0.201 t=0 0 m=audio 5004 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 Sep 2 09:04:55 VERBOSE[114695]: 11 headers, 8 lines Sep 2 09:04:55 DEBUG[114695]: Acked pending invite 102 Sep 2 09:04:55 DEBUG[114695]: Stopping retransmission on '25754e603557407a5d7415f25d65645e@192.168.0.1' of Request 102: Found Sep 2 09:04:55 VERBOSE[114695]: Found RTP audio format 8 Sep 2 09:04:55 VERBOSE[114695]: Peer audio RTP is at port 192.168.0.201:5004 Sep 2 09:04:55 DEBUG[114695]: Peer audio RTP is at port 192.168.0.201:5004 Sep 2 09:04:55 VERBOSE[114695]: Found description format PCMA Sep 2 09:04:55 VERBOSE[114695]: Capabilities: us - 0x8040e(GSM|ULAW|ALAW|ILBC|H263), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW) Sep 2 09:04:55 VERBOSE[114695]: Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Sep 2 09:04:55 DEBUG[114695]: build_route: Contact hop: Sep 2 09:04:55 VERBOSE[114695]: list_route: hop: Sep 2 09:04:55 VERBOSE[114695]: set_destination: Parsing for address/port to send to Sep 2 09:04:55 VERBOSE[114695]: set_destination: set destination to 192.168.0.201, port 5060 Sep 2 09:04:55 VERBOSE[114695]: Transmitting: ACK sip:2001@192.168.0.201;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5d7bfd6a From: "2000" ;tag=as15777eb1 To: ;tag=877729714034fadd Contact: Call-ID: 25754e603557407a5d7415f25d65645e@192.168.0.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.201:5060 Sep 2 09:04:55 VERBOSE[294930]: -- SIP/2001-de61 answered SIP/2000-b5bf Sep 2 09:04:55 DEBUG[294930]: Set channel SIP/2000-b5bf to read format ALAW Sep 2 09:04:55 DEBUG[294930]: Set channel SIP/2001-de61 to write format ALAW Sep 2 09:04:55 DEBUG[294930]: Set channel SIP/2000-b5bf to write format ALAW Sep 2 09:04:55 DEBUG[294930]: Set channel SIP/2001-de61 to read format ALAW Sep 2 09:04:55 DEBUG[294930]: sip_answer(SIP/2000-b5bf) Sep 2 09:04:55 VERBOSE[294930]: We're at 192.168.0.1 port 16254 Sep 2 09:04:55 VERBOSE[294930]: Answering with preferred capability 0x8(ALAW) Sep 2 09:04:55 VERBOSE[294930]: Answering with preferred capability 0x4(ULAW) Sep 2 09:04:55 VERBOSE[294930]: Answering with preferred capability 0x400(ILBC) Sep 2 09:04:55 VERBOSE[294930]: Answering with preferred capability 0x2(GSM) Sep 2 09:04:55 VERBOSE[294930]: Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.200;branch=z9hG4bKca965cb976fff083 From: "Tony Mountifield" ;tag=b49c41a0c48e2d80 To: ;tag=as63d94195 Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 19314 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 231 v=0 o=root 13712 13712 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 16254 RTP/AVP 8 0 98 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.0.200:5060 Sep 2 09:04:55 VERBOSE[294930]: -- Attempting native bridge of SIP/2000-b5bf and SIP/2001-de61 Sep 2 09:04:55 DEBUG[294930]: Got a FRAME_CONTROL (4) frame on channel SIP/2001-de61 Sep 2 09:04:55 DEBUG[294930]: Bridge stops bridging channels SIP/2000-b5bf and SIP/2001-de61 Sep 2 09:04:55 DEBUG[294930]: Read from SIP/2001-de61 (4,4) Sep 2 09:04:55 VERBOSE[294930]: -- Attempting native bridge of SIP/2000-b5bf and SIP/2001-de61 Sep 2 09:04:55 VERBOSE[114695]: Sip read: ACK sip:2001@192.168.0.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200;branch=z9hG4bK20511664077bb23e From: "Tony Mountifield" ;tag=b49c41a0c48e2d80 To: ;tag=as63d94195 Contact: Proxy-Authorization: DIGEST username="2000", realm="asterisk", algorithm=MD5, uri="sip:2001@192.168.0.1", nonce="025a6cf5", response="96a4dac4ea7aa881c7c5e341e6401ab1" Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 19314 ACK User-Agent: Grandstream BT100 1.0.5.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Sep 2 09:04:55 VERBOSE[114695]: 12 headers, 0 lines Sep 2 09:04:55 DEBUG[114695]: Stopping retransmission on '41d72bfad13725a4@192.168.0.200' of Response 19314: Found Sep 2 09:04:55 DEBUG[294930]: Ooh, format changed from UNKN to ALAW Sep 2 09:04:55 DEBUG[294930]: Ooh, format changed from UNKN to ALAW Sep 2 09:05:11 DEBUG[294930]: Got RTCP report of 8 bytes Sep 2 09:05:11 VERBOSE[114695]: Sip read: BYE sip:2000@192.168.0.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.201;branch=z9hG4bKe09ee17509da5991 From: ;tag=877729714034fadd To: "2000" ;tag=as15777eb1 Contact: Call-ID: 25754e603557407a5d7415f25d65645e@192.168.0.1 CSeq: 15135 BYE User-Agent: Grandstream BT100 1.0.5.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Sep 2 09:05:11 VERBOSE[114695]: 11 headers, 0 lines Sep 2 09:05:11 VERBOSE[114695]: Sending to 192.168.0.201 : 5060 (non-NAT) Sep 2 09:05:11 VERBOSE[114695]: Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.201;branch=z9hG4bKe09ee17509da5991 From: ;tag=877729714034fadd To: "2000" ;tag=as15777eb1 Call-ID: 25754e603557407a5d7415f25d65645e@192.168.0.1 CSeq: 15135 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.201:5060 Sep 2 09:05:11 DEBUG[294930]: Didn't get a frame from channel: SIP/2001-de61 Sep 2 09:05:11 DEBUG[294930]: Bridge stops bridging channels SIP/2000-b5bf and SIP/2001-de61 Sep 2 09:05:11 DEBUG[294930]: Hanging up channel 'SIP/2001-de61' Sep 2 09:05:11 DEBUG[294930]: sip_hangup(SIP/2001-de61) Sep 2 09:05:11 DEBUG[294930]: update_user_counter(2001) - decrement outUse counter Sep 2 09:05:11 DEBUG[294930]: Exiting with DIALSTATUS=ANSWER. Sep 2 09:05:11 DEBUG[294930]: Spawn extension (from-sip-internal,2001,1) exited non-zero on 'SIP/2000-b5bf' Sep 2 09:05:11 DEBUG[294930]: Launching 'Hangup' Sep 2 09:05:11 DEBUG[294930]: Spawn extension (from-sip-internal,h,1) exited non-zero on 'SIP/2000-b5bf' Sep 2 09:05:11 VERBOSE[114695]: Destroying call '25754e603557407a5d7415f25d65645e@192.168.0.1' Sep 2 09:05:11 DEBUG[294930]: Hanging up channel 'SIP/2000-b5bf' Sep 2 09:05:11 DEBUG[294930]: sip_hangup(SIP/2000-b5bf) Sep 2 09:05:11 DEBUG[294930]: update_user_counter(2000) - decrement inUse counter Sep 2 09:05:11 VERBOSE[294930]: set_destination: Parsing for address/port to send to Sep 2 09:05:11 VERBOSE[294930]: set_destination: set destination to 192.168.0.200, port 5060 Sep 2 09:05:11 VERBOSE[294930]: Reliably Transmitting: BYE sip:2000@192.168.0.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0d913656 From: ;tag=as63d94195 To: "Tony Mountifield" ;tag=b49c41a0c48e2d80 Contact: Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.200:5060 Sep 2 09:05:11 VERBOSE[114695]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0d913656 From: ;tag=as63d94195 To: "Tony Mountifield" ;tag=b49c41a0c48e2d80 Call-ID: 41d72bfad13725a4@192.168.0.200 CSeq: 102 BYE User-Agent: Grandstream BT100 1.0.5.9 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Sep 2 09:05:11 VERBOSE[114695]: 10 headers, 0 lines Sep 2 09:05:11 DEBUG[114695]: Stopping retransmission on '41d72bfad13725a4@192.168.0.200' of Request 102: Found Sep 2 09:05:11 VERBOSE[114695]: Message is BYE Sep 2 09:05:11 VERBOSE[114695]: Destroying call '41d72bfad13725a4@192.168.0.200'