== Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk 1.0.0, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk 1.0.0 currently running on xplanner (pid = 31455) xplanner*CLI> Verbosity was 0 and is now 3 Kxplanner*CLI> quithelp!clear15GquitK!script 7111.log15Gsip debug ip 192.168.0.19 xplanner*CLI> SIP Debugging Enabled for IP: 192.168.0.19 Kxplanner*CLI> Sip read: INVITE sip:5857501515@192.168.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK46ed8c471 Max-Forwards: 4 Content-Length: 472 To: 5857501515 From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097817 INVITE Supported: timer Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Content-Type: application/sdp Contact: L4-7111 Supported: replaces User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 752981766 IN IP4 192.168.0.19 s=SIP Call c=IN IP4 192.168.0.19 t=0 0 m=audio 3000 RTP/AVP 0 18 101 102 107 104 105 106 4 8 103 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 telephone-event/8000 a=fmtp:103 0-15 a=ptime:30 a=silenceSupp:on - - - - 23 headers, 20 lines Using latest request as basis request Sending to 192.168.0.19 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 103 Peer audio RTP is at port 192.168.0.19:3000 Found description format PCMU Found description format G729 Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x14d(G723|ULAW|ALAW|SLINR|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK46ed8c471 From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as5146e09d Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097817 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="3021038b" Content-Length: 0 to 192.168.0.19:5060 Scheduling destruction of call '3e1015a79ec26897820f337b0e3a1d50@192.168.0.19' in 15000 ms Found user '7111' Kxplanner*CLI> Sip read: ACK sip:5857501515@192.168.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK46ed8c471 Max-Forwards: 4 Content-Length: 0 To: 5857501515 ;tag=as5146e09d From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097817 ACK User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 9 headers, 0 lines Kxplanner*CLI> Sip read: INVITE sip:5857501515@192.168.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKef2a488f7 Max-Forwards: 4 Content-Length: 472 To: 5857501515 From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097818 INVITE Supported: timer Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Contact: L4-7111 Content-Type: application/sdp Supported: replaces Proxy-Authorization:Digest response="ba0957c427ddb5e82b978e8e8c80414d",username="7111",realm="asterisk",nonce="3021038b",uri="sip:5857501515@192.168.0.101:5060" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 752981766 IN IP4 192.168.0.19 s=SIP Call c=IN IP4 192.168.0.19 t=0 0 m=audio 3000 RTP/AVP 0 18 101 102 107 104 105 106 4 8 103 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 telephone-event/8000 a=fmtp:103 0-15 a=ptime:30 a=silenceSupp:on - - - - 24 headers, 20 lines Using latest request as basis request Sending to 192.168.0.19 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 103 Peer audio RTP is at port 192.168.0.19:3000 Found description format PCMU Found description format G729 Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x14d(G723|ULAW|ALAW|SLINR|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '7111' Looking for 5857501515 in sip list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKef2a488f7 From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as68fe0317 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097818 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> -- Executing Dial("SIP/7111-cc58", "SIP/5857501515|10") in new stack Kxplanner*CLI> -- Called 5857501515 Kxplanner*CLI> -- SIP/5857501515-46a6 is ringing Kxplanner*CLI> Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKef2a488f7 From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as68fe0317 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097818 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> -- SIP/5857501515-46a6 answered SIP/7111-cc58 Kxplanner*CLI> We're at 192.168.0.101 port 10550 Kxplanner*CLI> Answering with capability 0x2(GSM) Kxplanner*CLI> Answering with capability 0x4(ULAW) Kxplanner*CLI> Answering with capability 0x8(ALAW) Kxplanner*CLI> Answering with non-codec capability 0x1(G723) Kxplanner*CLI> Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKef2a488f7 From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as68fe0317 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097818 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 31684 31684 IN IP4 192.168.0.101 s=session c=IN IP4 192.168.0.101 t=0 0 m=audio 10550 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 - - - - to 192.168.0.19:5060 Kxplanner*CLI> -- Attempting native bridge of SIP/7111-cc58 and SIP/5857501515-46a6 Kxplanner*CLI> Sip read: ACK sip:5857501515@192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK6c0f46f91 Max-Forwards: 4 Content-Length: 0 To: 5857501515 ;tag=as68fe0317 From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097818 ACK Contact: L4-7111 Proxy-Authorization:Digest response="3d329cdd700a61f410625d857c633da4",username="7111",realm="asterisk",nonce="3021038b",uri="sip:5857501515@192.168.0.101" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 11 headers, 0 lines set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.19, port 5060 We're at 192.168.0.101 port 10550 Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) 11 headers, 11 lines Reliably Transmitting: INVITE sip:7111@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK4558b70b From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 239 v=0 o=root 31684 31685 IN IP4 192.168.0.17 s=session c=IN IP4 192.168.0.17 t=0 0 m=audio 3000 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 - - - - (no NAT) to 192.168.0.19:5060 Kxplanner*CLI> Sip read: SIP/2.0 100 Trying Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 102 INVITE From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK4558b70b Content-Length: 0 User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 8 headers, 0 lines Kxplanner*CLI> Sip read: SIP/2.0 200 OK Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 102 INVITE From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK4558b70b Content-Length: 247 Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Content-Type: application/sdp Supported: replaces Contact: L4-7111 User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 752981766 IN IP4 192.168.0.19 s=SIP Call c=IN IP4 192.168.0.19 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:30 a=sendrecv a=silenceSupp:on - - - - 21 headers, 12 lines Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.19:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0xc(ULAW|ALAW)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.19, port 5060 Transmitting: ACK sip:7111@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK6fe4a677 From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.19:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.19, port 5060 Kxplanner*CLI> We're at 192.168.0.101 port 10550 Kxplanner*CLI> Answering with capability 0x1(G723) Kxplanner*CLI> Answering with capability 0x4(ULAW) Kxplanner*CLI> Answering with capability 0x8(ALAW) Kxplanner*CLI> Answering with capability 0x40(SLINR) Kxplanner*CLI> Answering with capability 0x100(G729A) Kxplanner*CLI> Answering with non-codec capability 0x1(G723) Kxplanner*CLI> 11 headers, 14 lines Kxplanner*CLI> Reliably Transmitting: INVITE sip:7111@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK0a3349ca From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 314 v=0 o=root 31684 31686 IN IP4 192.168.0.17 s=session c=IN IP4 192.168.0.17 t=0 0 m=audio 3000 RTP/AVP 4 0 8 10 18 101 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:10 L16/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 192.168.0.19:5060 Kxplanner*CLI> Sip read: SIP/2.0 100 Trying Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 103 INVITE From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK0a3349ca Content-Length: 0 User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 8 headers, 0 lines Kxplanner*CLI> Sip read: SIP/2.0 200 OK Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 103 INVITE From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK0a3349ca Content-Length: 317 Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Content-Type: application/sdp Supported: replaces Contact: L4-7111 User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 752981767 IN IP4 192.168.0.19 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 3000 RTP/AVP 4 0 8 10 18 101 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:10 L16/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=ptime:30 a=sendrecv a=silenceSupp:on - - - - 21 headers, 15 lines Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 10 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:3000 Found description format G723 Found description format PCMU Found description format PCMA Found description format L16 Found description format G729 Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x14d(G723|ULAW|ALAW|SLINR|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) -- Started music on hold, class 'default', on SIP/5857501515-46a6 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.19, port 5060 Transmitting: ACK sip:7111@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK3928a7d1 From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.19:5060 Kxplanner*CLI> set_destination: Parsing for address/port to send to Kxplanner*CLI> set_destination: set destination to 192.168.0.19, port 5060 Kxplanner*CLI> We're at 192.168.0.101 port 10550 Kxplanner*CLI> Answering with capability 0x4(ULAW) Kxplanner*CLI> Answering with capability 0x8(ALAW) Kxplanner*CLI> Answering with non-codec capability 0x1(G723) Kxplanner*CLI> 11 headers, 11 lines Kxplanner*CLI> Reliably Transmitting: INVITE sip:7111@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK20749262 From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 239 v=0 o=root 31684 31687 IN IP4 192.168.0.17 s=session c=IN IP4 192.168.0.17 t=0 0 m=audio 3000 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 - - - - (no NAT) to 192.168.0.19:5060 Kxplanner*CLI> Sip read: SIP/2.0 100 Trying Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 104 INVITE From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK20749262 Content-Length: 0 User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 8 headers, 0 lines Kxplanner*CLI> Sip read: SIP/2.0 200 OK Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 104 INVITE From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK20749262 Content-Length: 242 Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Content-Type: application/sdp Supported: replaces Contact: L4-7111 User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 752981767 IN IP4 192.168.0.19 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:30 a=sendrecv a=silenceSupp:on - - - - 21 headers, 12 lines Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0xc(ULAW|ALAW)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) -- Stopped music on hold on SIP/5857501515-46a6 -- Started music on hold, class 'default', on SIP/5857501515-46a6 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.19, port 5060 Transmitting: ACK sip:7111@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK3f6bbccd From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.19:5060 Kxplanner*CLI> Sip read: INVITE sip:7112@192.168.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK312a1f865 Max-Forwards: 4 Content-Length: 473 To: 7112 From: L4-7111 ;tag=4dae3002bb6cc27 Call-ID: b69422d57a64139b4d1a06f740f286e0@192.168.0.19 CSeq: 1172999279 INVITE Supported: timer Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Content-Type: application/sdp Contact: L4-7111 Supported: replaces User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 1870490455 IN IP4 192.168.0.19 s=SIP Call c=IN IP4 192.168.0.19 t=0 0 m=audio 3002 RTP/AVP 0 18 101 102 107 104 105 106 4 8 103 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 telephone-event/8000 a=fmtp:103 0-15 a=ptime:30 a=silenceSupp:on - - - - 23 headers, 20 lines Using latest request as basis request Sending to 192.168.0.19 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 103 Peer audio RTP is at port 192.168.0.19:3002 Found description format PCMU Found description format G729 Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x14d(G723|ULAW|ALAW|SLINR|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK312a1f865 From: L4-7111 ;tag=4dae3002bb6cc27 To: 7112 ;tag=as00d4e2a9 Call-ID: b69422d57a64139b4d1a06f740f286e0@192.168.0.19 CSeq: 1172999279 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5462187d" Content-Length: 0 to 192.168.0.19:5060 Scheduling destruction of call 'b69422d57a64139b4d1a06f740f286e0@192.168.0.19' in 15000 ms Found user '7111' Kxplanner*CLI> Sip read: ACK sip:7112@192.168.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK312a1f865 Max-Forwards: 4 Content-Length: 0 To: 7112 ;tag=as00d4e2a9 From: L4-7111 ;tag=4dae3002bb6cc27 Call-ID: b69422d57a64139b4d1a06f740f286e0@192.168.0.19 CSeq: 1172999279 ACK User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 9 headers, 0 lines Kxplanner*CLI> Sip read: INVITE sip:7112@192.168.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK3b04c74af Max-Forwards: 4 Content-Length: 473 To: 7112 From: L4-7111 ;tag=4dae3002bb6cc27 Call-ID: b69422d57a64139b4d1a06f740f286e0@192.168.0.19 CSeq: 1172999280 INVITE Supported: timer Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Contact: L4-7111 Content-Type: application/sdp Supported: replaces Proxy-Authorization:Digest response="3798d4c14c1fd2dc86b444688488df10",username="7111",realm="asterisk",nonce="5462187d",uri="sip:7112@192.168.0.101:5060" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 1870490455 IN IP4 192.168.0.19 s=SIP Call c=IN IP4 192.168.0.19 t=0 0 m=audio 3002 RTP/AVP 0 18 101 102 107 104 105 106 4 8 103 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 telephone-event/8000 a=fmtp:103 0-15 a=ptime:30 a=silenceSupp:on - - - - 24 headers, 20 lines Using latest request as basis request Sending to 192.168.0.19 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 103 Peer audio RTP is at port 192.168.0.19:3002 Found description format PCMU Found description format G729 Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x14d(G723|ULAW|ALAW|SLINR|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '7111' Looking for 7112 in sip list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK3b04c74af From: L4-7111 ;tag=4dae3002bb6cc27 To: 7112 ;tag=as24cc7e7d Call-ID: b69422d57a64139b4d1a06f740f286e0@192.168.0.19 CSeq: 1172999280 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> -- Executing Dial("SIP/7111-cdf3", "SIP/7112|15") in new stack Kxplanner*CLI> -- Called 7112 Kxplanner*CLI> -- SIP/7112-0d8a is ringing Kxplanner*CLI> Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK3b04c74af From: L4-7111 ;tag=4dae3002bb6cc27 To: 7112 ;tag=as24cc7e7d Call-ID: b69422d57a64139b4d1a06f740f286e0@192.168.0.19 CSeq: 1172999280 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> Sip read: REFER sip:5857501515@192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKecfd17ff4 Max-Forwards: 4 Content-Length: 0 To: 5857501515 ;tag=as68fe0317 From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097819 REFER Supported: timer Contact: L4-7111 Refer-To: sip:7112@192.168.0.101:5060?Replaces=b69422d57a64139b4d1a06f740f286e0%40192.168.0.19%3bto-tag%3das24cc7e7d%3bfrom-tag%3d4dae3002bb6cc27 Referred-By: sip:7111@192.168.0.101:5060 Proxy-Authorization:Digest response="c7fab335fcfd33a4fe30b3412abdafab",username="7111",realm="asterisk",nonce="3021038b",uri="sip:5857501515@192.168.0.101" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 14 headers, 0 lines Looking for 7112 in sip Looking for 7111 in sip -- Stopped music on hold on SIP/5857501515-46a6 Transmitting (no NAT): SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKecfd17ff4 From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as68fe0317 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097819 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Reliably Transmitting: NOTIFY sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK3f6bbccd From: "Adrian" ;tag=as68fe0317 To: Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Event: refer;id=2131097819 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK (no NAT) to 192.168.0.19:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.19, port 5060 Reliably Transmitting: BYE sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK520f7a80 From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 106 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.19:5060 Kxplanner*CLI> Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK3b04c74af From: L4-7111 ;tag=4dae3002bb6cc27 To: 7112 ;tag=as24cc7e7d Call-ID: b69422d57a64139b4d1a06f740f286e0@192.168.0.19 CSeq: 1172999280 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> == Spawn extension (sip, 5857501515, 1) exited non-zero on 'SIP/7111-cc58' Kxplanner*CLI> Sip read: ACK sip:7112@192.168.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK3b04c74af Max-Forwards: 4 Content-Length: 0 To: 7112 ;tag=as24cc7e7d From: L4-7111 ;tag=4dae3002bb6cc27 Call-ID: b69422d57a64139b4d1a06f740f286e0@192.168.0.19 CSeq: 1172999280 ACK Proxy-Authorization:Digest response="ad5f046515e7c5873eaa9bec0b2d336b",username="7111",realm="asterisk",nonce="5462187d",uri="sip:7112@192.168.0.101:5060" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 10 headers, 0 lines Destroying call 'b69422d57a64139b4d1a06f740f286e0@192.168.0.19' Kxplanner*CLI> Retransmitting #1 (no NAT): NOTIFY sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK3f6bbccd From: "Adrian" ;tag=as68fe0317 To: Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Event: refer;id=2131097819 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK to 192.168.0.19:5060 Retransmitting #1 (no NAT): BYE sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK520f7a80 From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 106 BYE User-Agent: Asterisk PBX Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> Retransmitting #2 (no NAT): NOTIFY sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK3f6bbccd From: "Adrian" ;tag=as68fe0317 To: Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Event: refer;id=2131097819 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK to 192.168.0.19:5060 Retransmitting #2 (no NAT): BYE sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK520f7a80 From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 106 BYE User-Agent: Asterisk PBX Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> Retransmitting #3 (no NAT): NOTIFY sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK3f6bbccd From: "Adrian" ;tag=as68fe0317 To: Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Event: refer;id=2131097819 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK to 192.168.0.19:5060 Retransmitting #3 (no NAT): BYE sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK520f7a80 From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 106 BYE User-Agent: Asterisk PBX Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> Retransmitting #4 (no NAT): NOTIFY sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK3f6bbccd From: "Adrian" ;tag=as68fe0317 To: Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Event: refer;id=2131097819 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK to 192.168.0.19:5060 Retransmitting #4 (no NAT): BYE sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK520f7a80 From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 106 BYE User-Agent: Asterisk PBX Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> Retransmitting #5 (no NAT): NOTIFY sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK3f6bbccd From: "Adrian" ;tag=as68fe0317 To: Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Event: refer;id=2131097819 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK to 192.168.0.19:5060 Retransmitting #5 (no NAT): BYE sip:7111@ SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK520f7a80 From: 5857501515 ;tag=as68fe0317 To: L4-7111 ;tag=c1659e73110d7f6 Contact: Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 106 BYE User-Agent: Asterisk PBX Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> Oct 5 15:24:01 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 for seqno 105 (Non-critical Request) Oct 5 15:24:01 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 for seqno 106 (Non-critical Request) Destroying call '3e1015a79ec26897820f337b0e3a1d50@192.168.0.19' Kxplanner*CLI> -- SIP/7112-0d8a answered SIP/5857501515-46a6 -- Attempting native bridge of SIP/5857501515-46a6 and SIP/7112-0d8a Kxplanner*CLI> == Spawn extension (sip, 7112, 1) exited non-zero on 'SIP/5857501515-46a6' Kxplanner*CLI> Sip read: INVITE sip:5857501515@192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK440f5606a Max-Forwards: 4 Content-Length: 472 To: 5857501515 ;tag=as68fe0317 From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097820 INVITE Supported: timer Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Content-Type: application/sdp Contact: L4-7111 Supported: replaces Proxy-Authorization:Digest response="45484bf2cf3282f3cbf5ec35c47c3759",username="7111",realm="asterisk",nonce="3021038b",uri="sip:5857501515@192.168.0.101" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 752981768 IN IP4 192.168.0.19 s=SIP Call c=IN IP4 192.168.0.19 t=0 0 m=audio 3000 RTP/AVP 0 18 101 102 107 104 105 106 4 8 103 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - 24 headers, 20 lines Using latest request as basis request Sending to 192.168.0.19 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 103 Peer audio RTP is at port 192.168.0.19:3000 Found description format PCMU Found description format G729 Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x14d(G723|ULAW|ALAW|SLINR|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK440f5606a From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as68fe0317 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097820 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="16a79172" Content-Length: 0 to 192.168.0.19:5060 Scheduling destruction of call '3e1015a79ec26897820f337b0e3a1d50@192.168.0.19' in 15000 ms Found user '7111' Kxplanner*CLI> Sip read: ACK sip:5857501515@192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bK440f5606a Max-Forwards: 4 Content-Length: 0 To: 5857501515 ;tag=as68fe0317 From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097820 ACK Proxy-Authorization:Digest response="b9aea48ed00b83b5f69874cf7cbd1798",username="7111",realm="asterisk",nonce="16a79172",uri="sip:5857501515@192.168.0.101" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 10 headers, 0 lines Kxplanner*CLI> Sip read: INVITE sip:5857501515@192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKa50e4f0f4 Max-Forwards: 4 Content-Length: 472 To: 5857501515 ;tag=as68fe0317 From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097821 INVITE Supported: timer Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Contact: L4-7111 Content-Type: application/sdp Supported: replaces Proxy-Authorization:Digest response="a90ee8b1c55720f7ec488deb4722387d",username="7111",realm="asterisk",nonce="16a79172",uri="sip:5857501515@192.168.0.101" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 752981768 IN IP4 192.168.0.19 s=SIP Call c=IN IP4 192.168.0.19 t=0 0 m=audio 3000 RTP/AVP 0 18 101 102 107 104 105 106 4 8 103 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - 24 headers, 20 lines Using latest request as basis request Sending to 192.168.0.19 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 103 Peer audio RTP is at port 192.168.0.19:3000 Found description format PCMU Found description format G729 Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x14d(G723|ULAW|ALAW|SLINR|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '7111' Looking for 5857501515 in sip list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKa50e4f0f4 From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as68fe0317 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097821 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> -- Executing Dial("SIP/7111-bac2", "SIP/5857501515|10") in new stack Kxplanner*CLI> -- Called 5857501515 Kxplanner*CLI> -- SIP/5857501515-74ec is ringing Kxplanner*CLI> Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKa50e4f0f4 From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as68fe0317 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097821 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> Sip read: CANCEL sip:5857501515@192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKa50e4f0f4 Max-Forwards: 4 Content-Length: 0 To: 5857501515 ;tag=as68fe0317 From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097821 CANCEL Supported: timer Supported: replaces Proxy-Authorization:Digest response="e5aa000da4cf88cc14f6f1324dc6d96e",username="7111",realm="asterisk",nonce="16a79172",uri="sip:5857501515@192.168.0.101" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 12 headers, 0 lines Sending to 192.168.0.19 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKa50e4f0f4 From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as68fe0317 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097821 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Kxplanner*CLI> == Spawn extension (sip, 5857501515, 1) exited non-zero on 'SIP/7111-bac2' Kxplanner*CLI> Destroying call '3e1015a79ec26897820f337b0e3a1d50@192.168.0.19' Kxplanner*CLI> Sip read: BYE sip:5857501515@192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKbfbfe860c Max-Forwards: 4 Content-Length: 0 To: 5857501515 ;tag=as68fe0317 From: L4-7111 ;tag=c1659e73110d7f6 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097822 BYE Supported: timer Supported: replaces Proxy-Authorization:Digest response="067d9dbf72085d82f3ed76f2bdc95307",username="7111",realm="asterisk",nonce="16a79172",uri="sip:5857501515@192.168.0.101" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 12 headers, 0 lines Sending to 192.168.0.19 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.19;branch=z9hG4bKbfbfe860c From: L4-7111 ;tag=c1659e73110d7f6 To: 5857501515 ;tag=as68fe0317 Call-ID: 3e1015a79ec26897820f337b0e3a1d50@192.168.0.19 CSeq: 2131097822 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.19:5060 Destroying call '3e1015a79ec26897820f337b0e3a1d50@192.168.0.19' Kxplanner*CLI> stop now xplanner*CLI> Executing last minute cleanups Asterisk cleanly ending (0).