root@asterisk-13:/# asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv Asterisk 13.0.1, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 13.0.1 currently running on asterisk-13 (pid = 1238) asterisk-13*CLI> sip set debug peer 708 SIP Debugging Enabled for IP: 192.168.5.108 asterisk-13*CLI> bridge technology unsuspend native_rtp Unsuspended bridge technology 'native_rtp' -- Accepting AUTHENTICATED call from 27.111.14.68:4569: -- > requested format = alaw, -- > requested prefs = (alaw|g722|ulaw|g729|gsm), -- > actual format = ulaw, -- > host prefs = (ulaw|alaw|siren14|siren7|g722|slin16|slin|g726|g726aal2|adpcm|gsm|ilbc|speex|g729|speex16|testlaw|g719|opus|jpeg|png|h261...), -- > priority = mine -- Executing [099129107@from-099129107:1] Set("IAX2/099129107-12469", "CALLERID(number)=021624717") in new stack -- Executing [099129107@from-099129107:2] Goto("IAX2/099129107-12469", "default,707,1") in new stack -- Goto (default,707,1) -- Executing [707@default:1] Gosub("IAX2/099129107-12469", "707,stdexten(SIP/707)") in new stack -- Executing [707@default:50000] NoOp("IAX2/099129107-12469", "Start stdexten") in new stack -- Executing [707@default:50001] Answer("IAX2/099129107-12469", "") in new stack -- Executing [707@default:50002] Set("IAX2/099129107-12469", "LOCAL(ext)=707") in new stack -- Executing [707@default:50003] Set("IAX2/099129107-12469", "LOCAL(dev)=SIP/707") in new stack -- Executing [707@default:50004] Set("IAX2/099129107-12469", "LOCAL(cntx)=") in new stack -- Executing [707@default:50005] Set("IAX2/099129107-12469", "LOCAL(mbx)=707") in new stack -- Executing [707@default:50006] Dial("IAX2/099129107-12469", "SIP/707,20") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/707 -- SIP/707-00000012 is ringing -- SIP/707-00000012 answered IAX2/099129107-12469 -- Channel IAX2/099129107-12469 joined 'simple_bridge' basic-bridge -- Channel SIP/707-00000012 joined 'simple_bridge' basic-bridge > 0x7f2a58017390 -- Probation passed - setting RTP source address to 192.168.5.107:16438 -- Started music on hold, class 'default', on channel 'IAX2/099129107-12469' == Using SIP RTP CoS mark 5 -- Executing [708@numberplan-IAX2-SetDDI:1] NoOp("SIP/707-00000013", "callerid num is 707") in new stack -- Executing [708@numberplan-IAX2-SetDDI:2] GotoIf("SIP/707-00000013", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:3] GotoIf("SIP/707-00000013", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:4] GotoIf("SIP/707-00000013", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:5] GotoIf("SIP/707-00000013", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:6] GotoIf("SIP/707-00000013", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:7] GotoIf("SIP/707-00000013", "1?yesDDI") in new stack -- Goto (numberplan-IAX2-SetDDI,708,17) -- Executing [708@numberplan-IAX2-SetDDI:17] Set("SIP/707-00000013", "IAXCHAN=099129107") in new stack -- Executing [708@numberplan-IAX2-SetDDI:18] Goto("SIP/707-00000013", "numberplan-cost-efficient,708,1") in new stack -- Goto (numberplan-cost-efficient,708,1) -- Executing [708@numberplan-cost-efficient:1] Gosub("SIP/707-00000013", "708,stdexten(SIP/708)") in new stack -- Executing [708@numberplan-cost-efficient:50000] NoOp("SIP/707-00000013", "Start stdexten") in new stack -- Executing [708@numberplan-cost-efficient:50001] Answer("SIP/707-00000013", "") in new stack > 0x7f2a54765d10 -- Probation passed - setting RTP source address to 192.168.5.107:16440 -- Executing [708@numberplan-cost-efficient:50002] Set("SIP/707-00000013", "LOCAL(ext)=708") in new stack -- Executing [708@numberplan-cost-efficient:50003] Set("SIP/707-00000013", "LOCAL(dev)=SIP/708") in new stack -- Executing [708@numberplan-cost-efficient:50004] Set("SIP/707-00000013", "LOCAL(cntx)=") in new stack -- Executing [708@numberplan-cost-efficient:50005] Set("SIP/707-00000013", "LOCAL(mbx)=708") in new stack -- Executing [708@numberplan-cost-efficient:50006] Dial("SIP/707-00000013", "SIP/708,20") in new stack == Using SIP RTP CoS mark 5 Audio is at 18722 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding codec g723 to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec g729 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.5.108:5060: INVITE sip:708@192.168.5.108:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK66a27e54 Max-Forwards: 70 From: "Chris" ;tag=as59f5f8c3 To: Contact: Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.0.1 Date: Tue, 02 Dec 2014 21:13:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 898 v=0 o=root 1371140854 1371140854 IN IP4 192.168.5.43 s=Asterisk PBX 13.0.1 c=IN IP4 192.168.5.43 t=0 0 m=audio 18722 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- -- Called SIP/708 <--- SIP read from UDP:192.168.5.108:5060 ---> SIP/2.0 100 Trying To: From: "Chris" ;tag=as59f5f8c3 Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK66a27e54 Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.5.108:5060 ---> SIP/2.0 180 Ringing To: ;tag=a4e8d2cc231fd198i0 From: "Chris" ;tag=as59f5f8c3 Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK66a27e54 Contact: "Adrian Smith" Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/708-00000014 is ringing <--- SIP read from UDP:192.168.5.108:5060 ---> SIP/2.0 200 OK To: ;tag=a4e8d2cc231fd198i0 From: "Chris" ;tag=as59f5f8c3 Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK66a27e54 Contact: "Adrian Smith" Server: Linksys/SPA942-6.1.5(a) Content-Length: 206 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 13173 13173 IN IP4 192.168.5.108 s=- c=IN IP4 192.168.5.108 t=0 0 m=audio 16432 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 failed to extend from 64 to 98 Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.5.108:16432 sip_route_dump: route/path hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.5.108:5060 Transmitting (no NAT) to 192.168.5.108:5060: ACK sip:708@192.168.5.108:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK78d9216d Max-Forwards: 70 From: "Chris" ;tag=as59f5f8c3 To: ;tag=a4e8d2cc231fd198i0 Contact: Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.0.1 Content-Length: 0 --- -- SIP/708-00000014 answered SIP/707-00000013 -- Channel SIP/707-00000013 joined 'simple_bridge' basic-bridge <96b59eb0-8fd3-4fc2-b432-28fdbc2c7ac3> -- Channel SIP/708-00000014 joined 'simple_bridge' basic-bridge <96b59eb0-8fd3-4fc2-b432-28fdbc2c7ac3> > Bridge 96b59eb0-8fd3-4fc2-b432-28fdbc2c7ac3: switching from simple_bridge technology to native_rtp set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.5.108:5060 Audio is at 18722 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding codec g723 to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec g729 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.5.108:5060: INVITE sip:708@192.168.5.108:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK54e2f692 Max-Forwards: 70 From: "Chris" ;tag=as59f5f8c3 To: ;tag=a4e8d2cc231fd198i0 Contact: Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 13.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 900 v=0 o=root 1371140854 1371140855 IN IP4 192.168.5.107 s=Asterisk PBX 13.0.1 c=IN IP4 192.168.5.107 t=0 0 m=audio 16440 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.5.108:5060 ---> SIP/2.0 200 OK To: ;tag=a4e8d2cc231fd198i0 From: "Chris" ;tag=as59f5f8c3 Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 103 INVITE Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK54e2f692 Contact: "Adrian Smith" Server: Linksys/SPA942-6.1.5(a) Content-Length: 206 Content-Type: application/sdp v=0 o=- 13173 13174 IN IP4 192.168.5.108 s=- c=IN IP4 192.168.5.108 t=0 0 m=audio 16432 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (10 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 failed to extend from 64 to 98 Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.5.108:16432 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.5.108:5060 Transmitting (no NAT) to 192.168.5.108:5060: ACK sip:708@192.168.5.108:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK21c7bda9 Max-Forwards: 70 From: "Chris" ;tag=as59f5f8c3 To: ;tag=a4e8d2cc231fd198i0 Contact: Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 13.0.1 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.5.108:5060 Audio is at 18722 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding codec g723 to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec g729 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.5.108:5060: INVITE sip:708@192.168.5.108:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK5f37d5c2 Max-Forwards: 70 From: "Chris" ;tag=as59f5f8c3 To: ;tag=a4e8d2cc231fd198i0 Contact: Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 13.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 900 v=0 o=root 1371140854 1371140856 IN IP4 192.168.5.107 s=Asterisk PBX 13.0.1 c=IN IP4 192.168.5.107 t=0 0 m=audio 16440 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.5.108:5060 ---> SIP/2.0 200 OK To: ;tag=a4e8d2cc231fd198i0 From: "Chris" ;tag=as59f5f8c3 Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 104 INVITE Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK5f37d5c2 Contact: "Adrian Smith" Server: Linksys/SPA942-6.1.5(a) Content-Length: 206 Content-Type: application/sdp v=0 o=- 13173 13175 IN IP4 192.168.5.108 s=- c=IN IP4 192.168.5.108 t=0 0 m=audio 16432 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (10 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 failed to extend from 64 to 98 Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.5.108:16432 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.5.108:5060 Transmitting (no NAT) to 192.168.5.108:5060: ACK sip:708@192.168.5.108:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK6d233ae5 Max-Forwards: 70 From: "Chris" ;tag=as59f5f8c3 To: ;tag=a4e8d2cc231fd198i0 Contact: Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 13.0.1 Content-Length: 0 --- -- Channel IAX2/099129107-12469 left 'simple_bridge' basic-bridge -- Channel IAX2/099129107-12469 swapped with SIP/707-00000013 into 'native_rtp' basic-bridge <96b59eb0-8fd3-4fc2-b432-28fdbc2c7ac3> -- Channel SIP/707-00000013 left 'native_rtp' basic-bridge <96b59eb0-8fd3-4fc2-b432-28fdbc2c7ac3> > Bridge 96b59eb0-8fd3-4fc2-b432-28fdbc2c7ac3: switching from native_rtp technology to simple_bridge -- Channel SIP/707-00000012 left 'simple_bridge' basic-bridge -- Stopped music on hold on IAX2/099129107-12469 == Spawn extension (numberplan-cost-efficient, 708, 50006) exited non-zero on 'SIP/707-00000013' <--- SIP read from UDP:192.168.5.108:5060 ---> BYE sip:707@192.168.5.43:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.108:5060;branch=z9hG4bK-74e7d82e From: ;tag=a4e8d2cc231fd198i0 To: "Chris" ;tag=as59f5f8c3 Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 101 BYE Max-Forwards: 70 User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.5.108:5060 (no NAT) Scheduling destruction of SIP dialog '219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.5.108:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.108:5060;branch=z9hG4bK-74e7d82e;received=192.168.5.108 From: ;tag=a4e8d2cc231fd198i0 To: "Chris" ;tag=as59f5f8c3 Call-ID: 219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060 CSeq: 101 BYE Server: Asterisk PBX 13.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/708-00000014 left 'simple_bridge' basic-bridge <96b59eb0-8fd3-4fc2-b432-28fdbc2c7ac3> -- Channel IAX2/099129107-12469 left 'simple_bridge' basic-bridge <96b59eb0-8fd3-4fc2-b432-28fdbc2c7ac3> == Spawn extension (default, 707, 50006) exited non-zero on 'IAX2/099129107-12469' -- Hungup 'IAX2/099129107-12469' asterisk-13*CLI> bridge technology suspend native_rtp Suspended bridge technology 'native_rtp' -- Accepting AUTHENTICATED call from 27.111.14.68:4569: -- > requested format = alaw, -- > requested prefs = (alaw|g722|ulaw|g729|gsm), -- > actual format = ulaw, -- > host prefs = (ulaw|alaw|siren14|siren7|g722|slin16|slin|g726|g726aal2|adpcm|gsm|ilbc|speex|g729|speex16|testlaw|g719|opus|jpeg|png|h261...), -- > priority = mine -- Executing [099129107@from-099129107:1] Set("IAX2/099129107-9574", "CALLERID(number)=021624717") in new stack -- Executing [099129107@from-099129107:2] Goto("IAX2/099129107-9574", "default,707,1") in new stack -- Goto (default,707,1) -- Executing [707@default:1] Gosub("IAX2/099129107-9574", "707,stdexten(SIP/707)") in new stack -- Executing [707@default:50000] NoOp("IAX2/099129107-9574", "Start stdexten") in new stack -- Executing [707@default:50001] Answer("IAX2/099129107-9574", "") in new stack -- Executing [707@default:50002] Set("IAX2/099129107-9574", "LOCAL(ext)=707") in new stack -- Executing [707@default:50003] Set("IAX2/099129107-9574", "LOCAL(dev)=SIP/707") in new stack -- Executing [707@default:50004] Set("IAX2/099129107-9574", "LOCAL(cntx)=") in new stack -- Executing [707@default:50005] Set("IAX2/099129107-9574", "LOCAL(mbx)=707") in new stack -- Executing [707@default:50006] Dial("IAX2/099129107-9574", "SIP/707,20") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/707 -- SIP/707-00000015 is ringing -- SIP/707-00000015 answered IAX2/099129107-9574 -- Channel IAX2/099129107-9574 joined 'simple_bridge' basic-bridge -- Channel SIP/707-00000015 joined 'simple_bridge' basic-bridge > 0x7f2a6000a140 -- Probation passed - setting RTP source address to 192.168.5.107:16442 -- Started music on hold, class 'default', on channel 'IAX2/099129107-9574' == Using SIP RTP CoS mark 5 -- Executing [708@numberplan-IAX2-SetDDI:1] NoOp("SIP/707-00000016", "callerid num is 707") in new stack -- Executing [708@numberplan-IAX2-SetDDI:2] GotoIf("SIP/707-00000016", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:3] GotoIf("SIP/707-00000016", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:4] GotoIf("SIP/707-00000016", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:5] GotoIf("SIP/707-00000016", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:6] GotoIf("SIP/707-00000016", "0?yesDDI") in new stack -- Executing [708@numberplan-IAX2-SetDDI:7] GotoIf("SIP/707-00000016", "1?yesDDI") in new stack -- Goto (numberplan-IAX2-SetDDI,708,17) -- Executing [708@numberplan-IAX2-SetDDI:17] Set("SIP/707-00000016", "IAXCHAN=099129107") in new stack -- Executing [708@numberplan-IAX2-SetDDI:18] Goto("SIP/707-00000016", "numberplan-cost-efficient,708,1") in new stack -- Goto (numberplan-cost-efficient,708,1) -- Executing [708@numberplan-cost-efficient:1] Gosub("SIP/707-00000016", "708,stdexten(SIP/708)") in new stack -- Executing [708@numberplan-cost-efficient:50000] NoOp("SIP/707-00000016", "Start stdexten") in new stack -- Executing [708@numberplan-cost-efficient:50001] Answer("SIP/707-00000016", "") in new stack > 0x7f2a54765d10 -- Probation passed - setting RTP source address to 192.168.5.107:16444 -- Executing [708@numberplan-cost-efficient:50002] Set("SIP/707-00000016", "LOCAL(ext)=708") in new stack -- Executing [708@numberplan-cost-efficient:50003] Set("SIP/707-00000016", "LOCAL(dev)=SIP/708") in new stack -- Executing [708@numberplan-cost-efficient:50004] Set("SIP/707-00000016", "LOCAL(cntx)=") in new stack -- Executing [708@numberplan-cost-efficient:50005] Set("SIP/707-00000016", "LOCAL(mbx)=708") in new stack -- Executing [708@numberplan-cost-efficient:50006] Dial("SIP/707-00000016", "SIP/708,20") in new stack == Using SIP RTP CoS mark 5 Audio is at 11962 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding codec g723 to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec g729 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.5.108:5060: INVITE sip:708@192.168.5.108:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK21e64345 Max-Forwards: 70 From: "Chris" ;tag=as3b1a0548 To: Contact: Call-ID: 2c068b4701dcb1b520bb17225a206477@192.168.5.43:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.0.1 Date: Tue, 02 Dec 2014 21:14:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 898 v=0 o=root 1920080285 1920080285 IN IP4 192.168.5.43 s=Asterisk PBX 13.0.1 c=IN IP4 192.168.5.43 t=0 0 m=audio 11962 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- -- Called SIP/708 <--- SIP read from UDP:192.168.5.108:5060 ---> SIP/2.0 100 Trying To: From: "Chris" ;tag=as3b1a0548 Call-ID: 2c068b4701dcb1b520bb17225a206477@192.168.5.43:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK21e64345 Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.5.108:5060 ---> SIP/2.0 180 Ringing To: ;tag=de5ca339fbce0810i0 From: "Chris" ;tag=as3b1a0548 Call-ID: 2c068b4701dcb1b520bb17225a206477@192.168.5.43:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK21e64345 Contact: "Adrian Smith" Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/708-00000017 is ringing <--- SIP read from UDP:192.168.5.108:5060 ---> SIP/2.0 200 OK To: ;tag=de5ca339fbce0810i0 From: "Chris" ;tag=as3b1a0548 Call-ID: 2c068b4701dcb1b520bb17225a206477@192.168.5.43:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK21e64345 Contact: "Adrian Smith" Server: Linksys/SPA942-6.1.5(a) Content-Length: 206 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 18294 18294 IN IP4 192.168.5.108 s=- c=IN IP4 192.168.5.108 t=0 0 m=audio 16434 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 failed to extend from 64 to 98 Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.5.108:16434 sip_route_dump: route/path hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.5.108:5060 Transmitting (no NAT) to 192.168.5.108:5060: ACK sip:708@192.168.5.108:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.43:5060;branch=z9hG4bK13764d41 Max-Forwards: 70 From: "Chris" ;tag=as3b1a0548 To: ;tag=de5ca339fbce0810i0 Contact: Call-ID: 2c068b4701dcb1b520bb17225a206477@192.168.5.43:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.0.1 Content-Length: 0 --- -- SIP/708-00000017 answered SIP/707-00000016 -- Channel SIP/707-00000016 joined 'simple_bridge' basic-bridge <094a07f2-d4cf-4865-9f75-d22a20c28b74> -- Channel SIP/708-00000017 joined 'simple_bridge' basic-bridge <094a07f2-d4cf-4865-9f75-d22a20c28b74> > 0x7f2a541c74e0 -- Probation passed - setting RTP source address to 192.168.5.108:16434 Really destroying SIP dialog '219d4fdd5bf3b04e5aaf22ac6e5c68ea@192.168.5.43:5060' Method: BYE -- Channel IAX2/099129107-9574 left 'simple_bridge' basic-bridge -- Channel IAX2/099129107-9574 swapped with SIP/707-00000016 into 'simple_bridge' basic-bridge <094a07f2-d4cf-4865-9f75-d22a20c28b74> -- Channel SIP/707-00000016 left 'simple_bridge' basic-bridge <094a07f2-d4cf-4865-9f75-d22a20c28b74> -- Channel SIP/707-00000015 left 'simple_bridge' basic-bridge -- Stopped music on hold on IAX2/099129107-9574 == Spawn extension (numberplan-cost-efficient, 708, 50006) exited non-zero on 'SIP/707-00000016' <--- SIP read from UDP:192.168.5.108:5060 ---> BYE sip:707@192.168.5.43:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.108:5060;branch=z9hG4bK-57f10913 From: ;tag=de5ca339fbce0810i0 To: "Chris" ;tag=as3b1a0548 Call-ID: 2c068b4701dcb1b520bb17225a206477@192.168.5.43:5060 CSeq: 101 BYE Max-Forwards: 70 User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.5.108:5060 (no NAT) Scheduling destruction of SIP dialog '2c068b4701dcb1b520bb17225a206477@192.168.5.43:5060' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.5.108:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.108:5060;branch=z9hG4bK-57f10913;received=192.168.5.108 From: ;tag=de5ca339fbce0810i0 To: "Chris" ;tag=as3b1a0548 Call-ID: 2c068b4701dcb1b520bb17225a206477@192.168.5.43:5060 CSeq: 101 BYE Server: Asterisk PBX 13.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/708-00000017 left 'simple_bridge' basic-bridge <094a07f2-d4cf-4865-9f75-d22a20c28b74> -- Channel IAX2/099129107-9574 left 'simple_bridge' basic-bridge <094a07f2-d4cf-4865-9f75-d22a20c28b74> == Spawn extension (default, 707, 50006) exited non-zero on 'IAX2/099129107-9574' -- Hungup 'IAX2/099129107-9574'