siphub01*CLI> Verbosity is at least 3 siphub01*CLI> [Apr 21 14:39:56] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@_INBOUND:1] SIPAddHeader("SIP/_SIP01-00000000", ""Alert-Info:\;info=alert-external\;x-line-id=0"") in new stack siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@_INBOUND:2] Gosub("SIP/_SIP01-00000000", "_MYEXTEN,+31332990003,1") in new stack siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@_MYEXTEN:1] Goto("SIP/_SIP01-00000000", "BoxWares,+31332990003,1") in new stack siphub01*CLI> [Apr 21 14:39:56] -- Goto (BoxWares,+31332990003,1) siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@BoxWares:1] GotoIfTime("SIP/_SIP01-00000000", "00:00-07:00,mon-fri,*,*?gesloten,1") in new stack siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@BoxWares:2] GotoIfTime("SIP/_SIP01-00000000", "18:00-00:00,mon-fri,*,*?gesloten,1") in new stack siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@BoxWares:3] GotoIfTime("SIP/_SIP01-00000000", "*,sat,*,*?gesloten,1") in new stack siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@BoxWares:4] GotoIfTime("SIP/_SIP01-00000000", "*,sun,*,*?gesloten,1") in new stack siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@BoxWares:5] GotoIf("SIP/_SIP01-00000000", "0?gesloten,1") in new stack siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@BoxWares:6] Dial("SIP/_SIP01-00000000", "SIP/BoxWares.17,8") in new stack siphub01*CLI> [Apr 21 14:39:56] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] -- Called BoxWares.17 siphub01*CLI> [Apr 21 14:39:56] -- Got SIP response 486 "Busy Here" back from 10.51.1.5:5060 siphub01*CLI> [Apr 21 14:39:56] -- SIP/BoxWares.17-00000001 is busy siphub01*CLI> [Apr 21 14:39:56] == Everyone is busy/congested at this time (1:1/0/0) siphub01*CLI> [Apr 21 14:39:56] -- Executing [+31332990003@BoxWares:7] Dial("SIP/_SIP01-00000000", "SIP/BoxWares.17&SIP/BoxWares.18&SIP/BoxWares.19&SIP/BoxWares.20&SIP/BoxWares.21&SIP/BoxWares.22&SIP/BoxWares.23") in new stack siphub01*CLI> [Apr 21 14:39:56] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] -- Called BoxWares.17 siphub01*CLI> [Apr 21 14:39:56] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] -- Called BoxWares.18 siphub01*CLI> [Apr 21 14:39:56] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] -- Called BoxWares.19 siphub01*CLI> [Apr 21 14:39:56] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] -- Called BoxWares.20 siphub01*CLI> [Apr 21 14:39:56] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] -- Called BoxWares.21 siphub01*CLI> [Apr 21 14:39:56] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] -- Called BoxWares.22 siphub01*CLI> [Apr 21 14:39:56] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:39:56] -- Called BoxWares.23 siphub01*CLI> [Apr 21 14:39:56] -- SIP/BoxWares.20-00000005 is ringing siphub01*CLI> [Apr 21 14:39:56] -- SIP/BoxWares.23-00000008 is ringing siphub01*CLI> [Apr 21 14:39:56] -- SIP/BoxWares.19-00000004 is ringing siphub01*CLI> [Apr 21 14:39:56] -- SIP/BoxWares.21-00000006 is ringing siphub01*CLI> [Apr 21 14:39:56] -- Got SIP response 486 "Busy Here" back from 10.51.1.5:5060 siphub01*CLI> [Apr 21 14:39:56] -- SIP/BoxWares.17-00000002 is busy siphub01*CLI> [Apr 21 14:39:56] -- SIP/BoxWares.22-00000007 is ringing siphub01*CLI> [Apr 21 14:39:56] -- SIP/BoxWares.18-00000003 is ringing siphub01*CLI> [Apr 21 14:40:05] -- SIP/BoxWares.18-00000003 connected line has changed. Saving it until answer for SIP/_SIP01-00000000 siphub01*CLI> [Apr 21 14:40:05] -- SIP/BoxWares.18-00000003 answered SIP/_SIP01-00000000 siphub01*CLI> [Apr 21 14:40:05] -- Remotely bridging SIP/_SIP01-00000000 and SIP/BoxWares.18-00000003 siphub01*CLI> core set verbose 10 siphub01*CLI> core set debug 10 siphub01*CLI> g729 show version siphub01*CLI> g729 show licenses siphub01*CLI> sip set debug on siphub01*CLI> Verbosity was 3 and is now 10 Core debug was 0 and is now 10 Digium G.729A Module Version 1.8.0_3.1.5 (optimized for nocona_64) 0/0 encoders/decoders of 20 licensed channels are currently in use Licenses Found: File: G729-UQ4SW82GKK24.lic -- Key: G729-UQ4SW82GKK24 -- Host-ID: e6:53:63:2e:34:2a:7d:ed:59:a8:67:6a:f6:39:54:1f:be:00:00:d7 -- Channels: 18 (Expires: 2031-04-08) (OK) File: G729-NRF2DJZZCBHZ.lic -- Key: G729-NRF2DJZZCBHZ -- Host-ID: e6:53:63:2e:34:2a:7d:ed:59:a8:67:6a:f6:39:54:1f:be:00:00:d7 -- Channels: 2 (Expires: 2031-01-16) (OK) SIP Debugging enabled siphub01*CLI> [Apr 21 14:40:08] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:08] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:09] DEBUG[1021]: chan_sip.c:3631 __sip_autodestruct: Auto destroying SIP dialog '1843967320-5060-2@BA.FB.B.BA' siphub01*CLI> [Apr 21 14:40:09] DEBUG[1021]: chan_sip.c:5602 sip_destroy: Destroying SIP dialog 1843967320-5060-2@BA.FB.B.BA [Apr 21 14:40:09] Really destroying SIP dialog '1843967320-5060-2@BA.FB.B.BA' Method: REGISTER siphub01*CLI> [Apr 21 14:40:09] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:09] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:10] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:10] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:3631 __sip_autodestruct: Auto destroying SIP dialog '0bffcd20299b07be10beb9d66377c980@172.20.54.10:5060' siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:5602 sip_destroy: Destroying SIP dialog 0bffcd20299b07be10beb9d66377c980@172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:11] Really destroying SIP dialog '0bffcd20299b07be10beb9d66377c980@172.20.54.10:5060' Method: OPTIONS siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:11] <--- SIP read from UDP:10.1.1.20:5060 ---> INVITE sip:00651977655@siphub01.boxwares.wan SIP/2.0 Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-c506af9f From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 To: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Jeroen Draadloos" Expires: 240 User-Agent: Cisco/WIP310-5.0.13 Content-Length: 280 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE Allow-Events: dialog Supported: replaces Content-Type: application/sdp v=0 o=- 2425390 2425390 IN IP4 10.1.1.20 s=- c=IN IP4 10.1.1.20 t=0 0 m=audio 16384 RTP/AVP 0 2 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:40 a=sendrecv <-------------> siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 52]: INVITE sip:00651977655@siphub01.boxwares.wan SIP/2.0 [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 55]: Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-c506af9f siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 87]: From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 43]: To: siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 36]: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 101 INVITE siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 60]: Contact: "Jeroen Draadloos" siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 12]: Expires: 240 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 31]: User-Agent: Cisco/WIP310-5.0.13 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 19]: Content-Length: 280 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 72]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 20]: Allow-Events: dialog siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 13 [ 19]: Supported: replaces siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 14 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 15 [ 0]: siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 36]: o=- 2425390 2425390 IN IP4 10.1.1.20 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 3]: s=- siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 18]: c=IN IP4 10.1.1.20 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 34]: m=audio 16384 RTP/AVP 0 2 8 18 101 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 23]: a=rtpmap:2 G726-32/8000 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 22]: a=rtpmap:18 G729a/8000 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 11 [ 15]: a=fmtp:101 0-15 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 12 [ 10]: a=ptime:40 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 13 [ 10]: a=sendrecv [Apr 21 14:40:11] --- (15 headers 14 lines) --- siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: a8d24c89-98f0a9b8@10.1.1.20 (Checking From) --From tag ba1ed6559922ca5co0 --To-tag siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: acl.c:715 ast_ouraddrfor: For destination '10.1.1.20', our source address is '172.20.20.110'. siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:11] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for a8d24c89-98f0a9b8@10.1.1.20 - INVITE (No RTP) siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: sip/reqresp_parser.c:1584 parse_sip_options: Begin: parsing SIP "Supported: replaces" siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: sip/reqresp_parser.c:1600 parse_sip_options: Found SIP option: -replaces- siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: sip/reqresp_parser.c:1608 parse_sip_options: Matched SIP option: replaces siphub01*CLI> [Apr 21 14:40:11] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20:5060' gives... siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '5060'. siphub01*CLI> [Apr 21 14:40:12] Sending to 10.1.1.20:5060 (no NAT) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:21172 handle_request_invite: Initializing initreq for method INVITE - callid a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:12] Using INVITE request as basis request - a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:12] Found peer 'BoxWares.25' for 'BoxWares.25' from 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off siphub01*CLI> [Apr 21 14:40:12] <--- Reliably Transmitting (no NAT) to 10.1.1.20:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-c506af9f;received=10.1.1.20 From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 To: ;tag=as6798d6e2 Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 101 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="telbox", nonce="57c1f1ab" Content-Length: 0 <------------> siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #90 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:12] Scheduling destruction of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' in 32000 ms (Method: INVITE) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:12] <--- SIP read from UDP:10.1.1.20:5060 ---> ACK sip:00651977655@siphub01.boxwares.wan SIP/2.0 Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-c506af9f From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 To: ;tag=as6798d6e2 Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 101 ACK Max-Forwards: 70 Contact: "Jeroen Draadloos" User-Agent: Cisco/WIP310-5.0.13 Content-Length: 0 Allow-Events: dialog <-------------> siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 49]: ACK sip:00651977655@siphub01.boxwares.wan SIP/2.0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 55]: Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-c506af9f siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 87]: From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 58]: To: ;tag=as6798d6e2 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 36]: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 13]: CSeq: 101 ACK siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 60]: Contact: "Jeroen Draadloos" siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 31]: User-Agent: Cisco/WIP310-5.0.13 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 17]: Content-Length: 0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 20]: Allow-Events: dialog siphub01*CLI> [Apr 21 14:40:12] --- (11 headers 0 lines) --- siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: a8d24c89-98f0a9b8@10.1.1.20 (Checking From) --From tag ba1ed6559922ca5co0 --To-tag as6798d6e2 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received ACK (6) - Command in SIP ACK siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:3747 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #90 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:3780 __sip_ack: Stopping retransmission on 'a8d24c89-98f0a9b8@10.1.1.20' of Response 101: Match Found siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:12] <--- SIP read from UDP:10.1.1.20:5060 ---> INVITE sip:00651977655@siphub01.boxwares.wan SIP/2.0 Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-7ff52e13 From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 To: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="BoxWares.25",realm="telbox",nonce="57c1f1ab",uri="sip:00651977655@siphub01.boxwares.wan",algorithm=MD5,response="041de242f2971fd043928f6367894fe1" Contact: "Jeroen Draadloos" Expires: 240 User-Agent: Cisco/WIP310-5.0.13 Content-Length: 280 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE Allow-Events: dialog Supported: replaces Content-Type: application/sdp v=0 o=- 2425390 2425390 IN IP4 10.1.1.20 s=- c=IN IP4 10.1.1.20 t=0 0 m=audio 16384 RTP/AVP 0 2 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:40 a=sendrecv <-------------> siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 52]: INVITE sip:00651977655@siphub01.boxwares.wan SIP/2.0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 55]: Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-7ff52e13 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 87]: From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 43]: To: siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 36]: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 102 INVITE siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [178]: Authorization: Digest username="BoxWares.25",realm="telbox",nonce="57c1f1ab",uri="sip:00651977655@siphub01.boxwares.wan",algorithm=MD5,response="041de242f2971fd043928f6367894fe1" siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 60]: Contact: "Jeroen Draadloos" siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 12]: Expires: 240 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 31]: User-Agent: Cisco/WIP310-5.0.13 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 19]: Content-Length: 280 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 72]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 13 [ 20]: Allow-Events: dialog siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 14 [ 19]: Supported: replaces siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 15 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 16 [ 0]: siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 36]: o=- 2425390 2425390 IN IP4 10.1.1.20 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 3]: s=- siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 18]: c=IN IP4 10.1.1.20 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 34]: m=audio 16384 RTP/AVP 0 2 8 18 101 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 23]: a=rtpmap:2 G726-32/8000 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 22]: a=rtpmap:18 G729a/8000 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 11 [ 15]: a=fmtp:101 0-15 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 12 [ 10]: a=ptime:40 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 13 [ 10]: a=sendrecv siphub01*CLI> [Apr 21 14:40:12] --- (16 headers 14 lines) --- siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: a8d24c89-98f0a9b8@10.1.1.20 (Checking From) --From tag ba1ed6559922ca5co0 --To-tag siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting 'siphub01.boxwares.wan' gives... siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host 'siphub01.boxwares.wan' and port '(null)'. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting 'siphub01.boxwares.wan' gives... siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host 'siphub01.boxwares.wan' and port '(null)'. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20:5060' gives... siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '5060'. siphub01*CLI> [Apr 21 14:40:12] Sending to 10.1.1.20:5060 (no NAT) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:21172 handle_request_invite: Initializing initreq for method INVITE - callid a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:12] Using INVITE request as basis request - a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:12] Found peer 'BoxWares.25' for 'BoxWares.25' from 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:344 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xddc7a8' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: res_rtp_asterisk.c:472 ast_rtp_new: Allocated port 26568 for RTP instance '0xddc7a8' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:353 ast_rtp_instance_new: RTP instance '0xddc7a8' is setup and ready to go siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: res_rtp_asterisk.c:2370 ast_rtp_prop_set: Setup RTCP on RTP instance '0xddc7a8' siphub01*CLI> [Apr 21 14:40:12] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:4683 do_setnat: Setting NAT on RTP to Off siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP o=- 2425390 2425390 IN IP4 10.1.1.20... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP s=-... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20' gives... siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '(null)'. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP c=IN IP4 10.1.1.20... OK. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:12] Found RTP audio format 0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:12] Found RTP audio format 2 siphub01*CLI> [Apr 21 14:40:12] Found RTP audio format 8 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:12] Found RTP audio format 18 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:12] Found RTP audio format 101 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:12] Found audio description format PCMU for ID 0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. siphub01*CLI> [Apr 21 14:40:12] Found audio description format G726-32 for ID 2 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK. siphub01*CLI> [Apr 21 14:40:12] Found audio description format PCMA for ID 8 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. siphub01*CLI> [Apr 21 14:40:12] Found audio description format G729a for ID 18 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729a/8000... OK. siphub01*CLI> [Apr 21 14:40:12] Found audio description format telephone-event for ID 101 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=ptime:40... OK. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 2 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:12] Capabilities: us - 0x100 (g729), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Apr 21 14:40:12] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xddc7a8' siphub01*CLI> [Apr 21 14:40:12] Peer audio RTP is at port 10.1.1.20:16384 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x7fa2e8006560 to 0xddc970 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 2 from 0x7fa2e8006560 to 0xddc970 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0x7fa2e8006560 to 0xddc970 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0x7fa2e8006560 to 0xddc970 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa2e8006560 to 0xddc970 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:8619 process_sdp: We're settling with these formats: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:21320 handle_request_invite: Checking SIP call limits for device BoxWares.25 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:5454 update_call_counter: Updating call counter for incoming call siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:5559 update_call_counter: Call from peer 'BoxWares.25' is 1 out of 6 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1012]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - BoxWares.25 [Apr 21 14:40:12] Looking for 00651977655 in BoxWares (domain siphub01.boxwares.wan) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1012]: chan_sip.c:24969 sip_devicestate: Checking device state for peer BoxWares.25 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1012]: devicestate.c:458 do_state_change: Changing state for SIP/BoxWares.25 - state 2 (In use) [Apr 21 14:40:12] DEBUG[1012]: devicestate.c:438 devstate_event: device 'SIP/BoxWares.25' state '2' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1036]: app_queue.c:1330 handle_statechange: Device 'SIP/BoxWares.25' changed to state '2' (In use) but we don't care because they're not a member of any queue. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:6557 sip_new: *** Our native formats are 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:6558 sip_new: *** Joint capabilities are 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:6559 sip_new: *** Our capabilities are 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:6560 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:6590 sip_new: This channel will not be able to handle video. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:13287 build_route: build_route: Contact hop: "Jeroen Draadloos" siphub01*CLI> [Apr 21 14:40:12] list_route: hop: siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:21612 handle_request_invite: SIP/BoxWares.25-00000009: New call is still down.... Trying... siphub01*CLI> [Apr 21 14:40:12] <--- Transmitting (no NAT) to 10.1.1.20:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-7ff52e13;received=10.1.1.20 From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 To: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 102 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1012]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - BoxWares.25 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1012]: chan_sip.c:24969 sip_devicestate: Checking device state for peer BoxWares.25 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1012]: devicestate.c:458 do_state_change: Changing state for SIP/BoxWares.25 - state 2 (In use) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1012]: devicestate.c:438 devstate_event: device 'SIP/BoxWares.25' state '2' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1036]: app_queue.c:1330 handle_statechange: Device 'SIP/BoxWares.25' changed to state '2' (In use) but we don't care because they're not a member of any queue. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'EXTEN' is '00651977655' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: func_strings.c:738 filter: c1=48, c2=57 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: func_strings.c:752 filter: Allowed: 0123456789 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3895 pbx_substitute_variables_helper_full: Function result is '00651977655' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'Goto' siphub01*CLI> [Apr 21 14:40:12] -- Executing [00651977655@BoxWares:1] Goto("SIP/BoxWares.25-00000009", "_OUTBOUND,00651977655,1") in new stack siphub01*CLI> [Apr 21 14:40:12] -- Goto (_OUTBOUND,00651977655,1) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'EXTEN' is '00651977655' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'Goto' siphub01*CLI> [Apr 21 14:40:12] -- Executing [00651977655@_OUTBOUND:1] Goto("SIP/BoxWares.25-00000009", "_OUTBOUND_now,+31651977655,1") in new stack siphub01*CLI> [Apr 21 14:40:12] -- Goto (_OUTBOUND_now,+31651977655,1) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'SIPAddHeader' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:1] SIPAddHeader("SIP/BoxWares.25-00000009", ""Alert-Info:\;info=alert-external\;x-line-id=0"") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:27810 sip_addheader: SIP Header added ""Alert-Info:\;info=alert-external\;x-line-id=0"" as __SIPADDHEADER01 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'Asserted' is '+31332990003' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'NoOp' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:2] NoOp("SIP/BoxWares.25-00000009", "Asserted=+31332990003") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'Asserted' is '+31332990003' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3963 pbx_substitute_variables_helper_full: Expression result is '0' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'ExecIf' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:3] ExecIf("SIP/BoxWares.25-00000009", "0?Hangup(21)") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'Asserted' is '+31332990003' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'Set' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:4] Set("SIP/BoxWares.25-00000009", "CALLERID(all)=+31332990003") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'Asserted' is '+31332990003' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'Asserted' is '+31332990003' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'SIPAddHeader' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:5] SIPAddHeader("SIP/BoxWares.25-00000009", "P-Asserted-Identity: "+31332990003" ") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:27810 sip_addheader: SIP Header added "P-Asserted-Identity: "+31332990003" " as __SIPADDHEADER02 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3093 ast_str_retrieve_variable: Result of 'Anonymous' is NULL siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3963 pbx_substitute_variables_helper_full: Expression result is '0' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'GotoIf' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:6] GotoIf("SIP/BoxWares.25-00000009", "0?:continue") in new stack siphub01*CLI> [Apr 21 14:40:12] -- Goto (_OUTBOUND_now,+31651977655,9) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'EXTEN' is '+31651977655' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: func_strings.c:738 filter: c1=48, c2=57 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: func_strings.c:752 filter: Allowed: 0123456789 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3895 pbx_substitute_variables_helper_full: Function result is '31651977655' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'Gosub' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:9] Gosub("SIP/BoxWares.25-00000009", "_MYEXTEN,+31651977655,1") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: app_stack.c:360 gosub_exec: Channel SIP/BoxWares.25-00000009 has no datastore, so we're allocating one. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'Return' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_MYEXTEN:1] Return("SIP/BoxWares.25-00000009", "") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'Accountcode' is 'BoxWares' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'NoOp' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:10] NoOp("SIP/BoxWares.25-00000009", "Accountcode=BoxWares") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'Accountcode' is 'BoxWares' [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3963 pbx_substitute_variables_helper_full: Expression result is '0' [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'ExecIf' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:11] ExecIf("SIP/BoxWares.25-00000009", "0?Hangup(21)") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'Accountcode' is 'BoxWares' [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'Set' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:12] Set("SIP/BoxWares.25-00000009", "CDR(accountcode)=BoxWares") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'Set' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:13] Set("SIP/BoxWares.25-00000009", "CDR(amaflags)=BILLING") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'Set' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:14] Set("SIP/BoxWares.25-00000009", "CDR(userfield)=Tele2.out") in new stack siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3096 ast_str_retrieve_variable: Result of 'EXTEN' is '+31651977655' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: func_strings.c:738 filter: c1=48, c2=57 [Apr 21 14:40:12] DEBUG[1092]: func_strings.c:752 filter: Allowed: 0123456789 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: pbx.c:3895 pbx_substitute_variables_helper_full: Function result is '31651977655' [Apr 21 14:40:12] DEBUG[1092]: pbx.c:4067 pbx_extension_helper: Launching 'Dial' siphub01*CLI> [Apr 21 14:40:12] -- Executing [+31651977655@_OUTBOUND_now:15] Dial("SIP/BoxWares.25-00000009", "SIP/_SIP01/+31651977655,60,L(14400000)") in new stack siphub01*CLI> [Apr 21 14:40:12] -- Setting call duration limit to 14400.000 seconds. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:25071 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] == Using UDPTL CoS mark 5 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 016f8b3e00f5ecec68da5318688a7403@127.0.0.2:0 - INVITE (No RTP) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: rtp_engine.c:344 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xda8138' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: res_rtp_asterisk.c:472 ast_rtp_new: Allocated port 17582 for RTP instance '0xda8138' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: rtp_engine.c:353 ast_rtp_instance_new: RTP instance '0xda8138' is setup and ready to go siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: res_rtp_asterisk.c:2370 ast_rtp_prop_set: Setup RTCP on RTP instance '0xda8138' siphub01*CLI> [Apr 21 14:40:12] == Using SIP RTP CoS mark 5 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:4683 do_setnat: Setting NAT on RTP to Off siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:4691 do_setnat: Setting NAT on UDPTL to Off siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:2894 obproxy_get: OBPROXY: Not applying OBproxy to this call siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: acl.c:715 ast_ouraddrfor: For destination '172.20.54.10', our source address is '172.20.20.110'. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:6557 sip_new: *** Our native formats are 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:6558 sip_new: *** Joint capabilities are 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:6559 sip_new: *** Our capabilities are 0x108 (alaw|g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:6560 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:6562 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:6590 sip_new: This channel will not be able to handle video. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: rtp_engine.c:1446 ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 'SIP/_SIP01-0000000a' with that of 'SIP/BoxWares.25-00000009' siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable DIALEDTIME. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable DIALSTATUS. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable GOSUB_RETVAL. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5896 ast_channel_inherit_variables: Copying hard-transferable variable SIPADDHEADER02. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5896 ast_channel_inherit_variables: Copying hard-transferable variable SIPADDHEADER01. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable Asserted. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable Accountcode. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable SIPCALLID. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: channel.c:5900 ast_channel_inherit_variables: Not copying variable SIPURI. siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:5212 sip_call: Outgoing Call for +31651977655 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:5454 update_call_counter: Updating call counter for outgoing call siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:11438 transmit_invite: Adding SIP Header "P-Asserted-Identity" with content :"+31332990003" : siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:11438 transmit_invite: Adding SIP Header "Alert-Info" with content :;info=alert-external;x-line-id=0: siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:10570 add_sdp: ** Our capability: 0x108 (alaw|g729) Video flag: False Text flag: False siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:12] Audio is at 5060 siphub01*CLI> [Apr 21 14:40:12] Adding codec 0x100 (g729) to SDP siphub01*CLI> [Apr 21 14:40:12] Adding codec 0x8 (alaw) to SDP siphub01*CLI> [Apr 21 14:40:12] Adding non-codec 0x1 (telephone-event) to SDP siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:10680 add_sdp: -- Done with adding codecs to SDP siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:10819 add_sdp: Done building SDP. Settling with this capability: 0x108 (alaw|g729) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method INVITE - callid 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 0 [ 49]: INVITE sip:+31651977655@172.20.54.10:5060 SIP/2.0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 1 [ 58]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7ba9cdf7 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 2 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 3 [ 68]: From: "+31332990003" ;tag=as6f2fe75b siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 4 [ 40]: To: siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 5 [ 46]: Contact: siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 6 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 7 [ 16]: CSeq: 102 INVITE siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 8 [ 18]: User-Agent: TelBox siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 9 [ 35]: Date: Thu, 21 Apr 2011 12:40:12 GMT siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 11 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 12 [ 84]: P-Asserted-Identity: "+31332990003" siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 13 [ 59]: Alert-Info: ;info=alert-external;x-line-id=0 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 14 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:12] Reliably Transmitting (no NAT) to 172.20.54.10:5060: INVITE sip:+31651977655@172.20.54.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7ba9cdf7 Max-Forwards: 70 From: "+31332990003" ;tag=as6f2fe75b To: Contact: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 102 INVITE User-Agent: TelBox Date: Thu, 21 Apr 2011 12:40:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "+31332990003" Alert-Info: ;info=alert-external;x-line-id=0 Content-Type: application/sdp Content-Length: 283 v=0 o=root 126127222 126127222 IN IP4 172.20.20.110 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.20.110 t=0 0 m=audio 17582 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #93 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1092]: chan_sip.c:3089 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:12] -- Called _SIP01/+31651977655 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:12] <--- SIP read from UDP:172.20.54.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7ba9cdf7;received=172.20.20.110 From: "+31332990003" ;tag=as6f2fe75b To: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 102 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <-------------> siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 81]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7ba9cdf7;received=172.20.20.110 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 68]: From: "+31332990003" ;tag=as6f2fe75b siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 40]: To: siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 102 INVITE siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 14]: Server: TelBox siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 45]: Contact: siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 17]: Content-Length: 0 siphub01*CLI> [Apr 21 14:40:12] --- (11 headers 0 lines) --- siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 (Checking To) --From tag as6f2fe75b --To-tag siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:3814 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #93 - INVITE (got response) siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:3821 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Request 102: Found [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:18794 handle_response_invite: SIP response 100 to standard invite siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:12] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:13] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:13] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:14] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:14] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:15] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:15] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:15] <--- SIP read from UDP:10.1.1.41:5060 ---> <-------------> siphub01*CLI> [Apr 21 14:40:15] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:15] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:16] <--- SIP read from UDP:172.20.54.10:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7ba9cdf7;received=172.20.20.110 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 102 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 282 v=0 o=root 2051453966 2051453966 IN IP4 172.20.54.10 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.54.10 t=0 0 m=audio 5630 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 28]: SIP/2.0 183 Session Progress siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 81]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7ba9cdf7;received=172.20.20.110 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 68]: From: "+31332990003" ;tag=as6f2fe75b siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 55]: To: ;tag=as75358887 [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 102 INVITE siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 14]: Server: TelBox [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 45]: Contact: siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 19]: Content-Length: 282 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 0]: siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 48]: o=root 2051453966 2051453966 IN IP4 172.20.54.10 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 22]: s=Asterisk PBX 1.8.3.2 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 21]: c=IN IP4 172.20.54.10 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 29]: m=audio 5630 RTP/AVP 18 8 101 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 21]: a=rtpmap:18 G729/8000 [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 19]: a=fmtp:18 annexb=no siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 15]: a=fmtp:101 0-16 [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 11 [ 10]: a=ptime:20 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 12 [ 10]: a=sendrecv [Apr 21 14:40:16] --- (12 headers 13 lines) --- siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 (Checking To) --From tag as6f2fe75b --To-tag as75358887 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:3821 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Request 102: Found siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:18794 handle_response_invite: SIP response 183 to standard invite siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP o=root 2051453966 2051453966 IN IP4 172.20.54.10... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP s=Asterisk PBX 1.8.3.2... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10' gives... siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '(null)'. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP c=IN IP4 172.20.54.10... OK. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:16] Found RTP audio format 18 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:16] Found RTP audio format 8 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:16] Found RTP audio format 101 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:16] Found audio description format G729 for ID 18 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:16] Found audio description format PCMA for ID 8 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. siphub01*CLI> [Apr 21 14:40:16] Found audio description format telephone-event for ID 101 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:16] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729) siphub01*CLI> [Apr 21 14:40:16] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xda8138' siphub01*CLI> [Apr 21 14:40:16] Peer audio RTP is at port 172.20.54.10:5630 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0x7fa2e80066a0 to 0xda8300 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0x7fa2e80066a0 to 0xda8300 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa2e80066a0 to 0xda8300 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8619 process_sdp: We're settling with these formats: 0x108 (alaw|g729) siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:8624 process_sdp: We have an owner, now see if we need to change this call siphub01*CLI> [Apr 21 14:40:16] -- SIP/_SIP01-0000000a is making progress passing it to SIP/BoxWares.25-00000009 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1092]: rtp_engine.c:1531 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/BoxWares.25-00000009' with that of 'SIP/_SIP01-0000000a' siphub01*CLI> [Apr 21 14:40:16] DEBUG[1092]: chan_sip.c:10924 transmit_response_with_sdp: Setting framing from config on incoming call [Apr 21 14:40:16] DEBUG[1092]: chan_sip.c:10570 add_sdp: ** Our capability: 0x100 (g729) Video flag: True Text flag: True siphub01*CLI> [Apr 21 14:40:16] DEBUG[1092]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x0 (nothing) siphub01*CLI> [Apr 21 14:40:16] Audio is at 5060 [Apr 21 14:40:16] Adding codec 0x100 (g729) to SDP siphub01*CLI> [Apr 21 14:40:16] Adding non-codec 0x1 (telephone-event) to SDP siphub01*CLI> [Apr 21 14:40:16] DEBUG[1092]: chan_sip.c:10680 add_sdp: -- Done with adding codecs to SDP siphub01*CLI> [Apr 21 14:40:16] DEBUG[1092]: chan_sip.c:10819 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Apr 21 14:40:16] <--- Transmitting (no NAT) to 10.1.1.20:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-7ff52e13;received=10.1.1.20 From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 To: ;tag=as2138f877 Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 102 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 1948114604 1948114604 IN IP4 172.20.20.110 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.20.110 t=0 0 m=audio 26568 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=sendrecv <------------> siphub01*CLI> [Apr 21 14:40:16] DEBUG[1092]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 183' onto UDP socket destined for 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:16] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:17] DEBUG[1092]: res_rtp_asterisk.c:1239 ast_rtp_write: Ooh, format changed from unknown to g729 [Apr 21 14:40:17] DEBUG[1092]: res_rtp_asterisk.c:1270 ast_rtp_write: Created smoother: format: g729 ms: 40 len: 40 [Apr 21 14:40:17] DEBUG[1092]: res_rtp_asterisk.c:1140 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0xddc7a8' siphub01*CLI> [Apr 21 14:40:17] DEBUG[1092]: res_rtp_asterisk.c:1239 ast_rtp_write: Ooh, format changed from unknown to g729 [Apr 21 14:40:17] DEBUG[1092]: res_rtp_asterisk.c:1270 ast_rtp_write: Created smoother: format: g729 ms: 40 len: 40 siphub01*CLI> [Apr 21 14:40:17] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:17] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:18] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:18] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:19] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:19] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:21] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:21] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 64 bytes siphub01*CLI> [Apr 21 14:40:22] <--- SIP read from UDP:172.20.54.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7ba9cdf7;received=172.20.20.110 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 102 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 282 v=0 o=root 2051453966 2051453967 IN IP4 172.20.54.10 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.54.10 t=0 0 m=audio 5630 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 81]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7ba9cdf7;received=172.20.20.110 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 68]: From: "+31332990003" ;tag=as6f2fe75b [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 55]: To: ;tag=as75358887 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 14]: Server: TelBox [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 26]: Supported: replaces, timer [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 45]: Contact: [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 29]: Content-Type: application/sdp [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 19]: Content-Length: 282 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 0]: [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 48]: o=root 2051453966 2051453967 IN IP4 172.20.54.10 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 22]: s=Asterisk PBX 1.8.3.2 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 21]: c=IN IP4 172.20.54.10 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 29]: m=audio 5630 RTP/AVP 18 8 101 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 21]: a=rtpmap:18 G729/8000 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 19]: a=fmtp:18 annexb=no [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 15]: a=fmtp:101 0-16 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 11 [ 10]: a=ptime:20 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 12 [ 10]: a=sendrecv [Apr 21 14:40:22] --- (12 headers 13 lines) --- [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 (Checking To) --From tag as6f2fe75b --To-tag as75358887 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3742 __sip_ack: Acked pending invite 102 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' of Request 102: Match Found [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:18794 handle_response_invite: SIP response 200 to standard invite [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP o=root 2051453966 2051453967 IN IP4 172.20.54.10... UNSUPPORTED. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP s=Asterisk PBX 1.8.3.2... UNSUPPORTED. [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10' gives... [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '(null)'. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP c=IN IP4 172.20.54.10... OK. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [Apr 21 14:40:22] Found RTP audio format 18 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x7fa2e80066a0 [Apr 21 14:40:22] Found RTP audio format 8 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fa2e80066a0 [Apr 21 14:40:22] Found RTP audio format 101 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa2e80066a0 [Apr 21 14:40:22] Found audio description format G729 for ID 18 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Apr 21 14:40:22] Found audio description format PCMA for ID 8 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Apr 21 14:40:22] Found audio description format telephone-event for ID 101 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x7fa2e80066a0 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x7fa2e80066a0 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa2e80066a0 [Apr 21 14:40:22] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729) [Apr 21 14:40:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xda8138' [Apr 21 14:40:22] Peer audio RTP is at port 172.20.54.10:5630 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0x7fa2e80066a0 to 0xda8300 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0x7fa2e80066a0 to 0xda8300 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa2e80066a0 to 0xda8300 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8619 process_sdp: We're settling with these formats: 0x108 (alaw|g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8624 process_sdp: We have an owner, now see if we need to change this call [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:5454 update_call_counter: Updating call counter for outgoing call siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:13287 build_route: build_route: Contact hop: siphub01*CLI> [Apr 21 14:40:22] list_route: hop: siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:9580 reqprep: Strict routing enforced for session 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] set_destination: Parsing for address/port to send to [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. [Apr 21 14:40:22] set_destination: set destination to 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:22] Transmitting (no NAT) to 172.20.54.10:5060: ACK sip:+31651977655@172.20.54.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK61d17cbe Max-Forwards: 70 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Contact: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 102 ACK User-Agent: TelBox Content-Length: 0 --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'ACK sip:+31' onto UDP socket destined for 172.20.54.10:5060 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1012]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - _SIP01 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1012]: chan_sip.c:24969 sip_devicestate: Checking device state for peer _SIP01 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1012]: devicestate.c:458 do_state_change: Changing state for SIP/_SIP01 - state 1 (Not in use) [Apr 21 14:40:22] DEBUG[1012]: devicestate.c:438 devstate_event: device 'SIP/_SIP01' state '1' [Apr 21 14:40:22] -- SIP/_SIP01-0000000a answered SIP/BoxWares.25-00000009 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1036]: app_queue.c:1330 handle_statechange: Device 'SIP/_SIP01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: rtp_engine.c:1531 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/BoxWares.25-00000009' with that of 'SIP/_SIP01-0000000a' siphub01*CLI> [Apr 21 14:40:22] DEBUG[1012]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - BoxWares.25 [Apr 21 14:40:22] DEBUG[1012]: chan_sip.c:24969 sip_devicestate: Checking device state for peer BoxWares.25 [Apr 21 14:40:22] DEBUG[1012]: devicestate.c:458 do_state_change: Changing state for SIP/BoxWares.25 - state 2 (In use) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1012]: devicestate.c:438 devstate_event: device 'SIP/BoxWares.25' state '2' siphub01*CLI> [Apr 21 14:40:22] DEBUG[1036]: app_queue.c:1330 handle_statechange: Device 'SIP/BoxWares.25' changed to state '2' (In use) but we don't care because they're not a member of any queue. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:6028 sip_answer: SIP answering channel: SIP/BoxWares.25-00000009 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10924 transmit_response_with_sdp: Setting framing from config on incoming call siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10570 add_sdp: ** Our capability: 0x100 (g729) Video flag: True Text flag: True siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x0 (nothing) siphub01*CLI> [Apr 21 14:40:22] Audio is at 5060 siphub01*CLI> [Apr 21 14:40:22] Adding codec 0x100 (g729) to SDP siphub01*CLI> [Apr 21 14:40:22] Adding non-codec 0x1 (telephone-event) to SDP siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10680 add_sdp: -- Done with adding codecs to SDP siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10819 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] <--- Reliably Transmitting (no NAT) to 10.1.1.20:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-7ff52e13;received=10.1.1.20 From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 To: ;tag=as2138f877 Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 102 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 1948114604 1948114605 IN IP4 172.20.20.110 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.20.110 t=0 0 m=audio 26568 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=sendrecv <------------> siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #98 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: features.c:3517 ast_bridge_call: bridge answer set, chan answer set siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: channel.c:6062 ast_set_owners_and_peers: setting peeraccount to BoxWares for SIP/BoxWares.25-00000009 from data on channel SIP/_SIP01-0000000a siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [Apr 21 14:40:22] DEBUG[1092]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update siphub01*CLI> [Apr 21 14:40:22] -- Remotely bridging SIP/BoxWares.25-00000009 and SIP/_SIP01-0000000a siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:27695 sip_set_rtp_peer: Deferring reinvite on SIP 'a8d24c89-98f0a9b8@10.1.1.20' - It's audio will be redirected to IP 172.20.54.10:5630 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:27692 sip_set_rtp_peer: Sending reinvite on SIP '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' - It's audio soon redirected to IP 10.1.1.20:16384 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:9580 reqprep: Strict routing enforced for session 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:22] set_destination: set destination to 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10570 add_sdp: ** Our capability: 0x108 (alaw|g729) Video flag: True Text flag: True [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x100 (g729) [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10575 add_sdp: ** Our native-bridge filtered capablity: 0x108 (alaw|g729) siphub01*CLI> [Apr 21 14:40:22] Audio is at 5060 siphub01*CLI> [Apr 21 14:40:22] Adding codec 0x100 (g729) to SDP siphub01*CLI> [Apr 21 14:40:22] Adding codec 0x8 (alaw) to SDP [Apr 21 14:40:22] Adding non-codec 0x1 (telephone-event) to SDP siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10680 add_sdp: -- Done with adding codecs to SDP siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10819 add_sdp: Done building SDP. Settling with this capability: 0x108 (alaw|g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:2821 initialize_initreq: Initializing already initialized SIP dialog 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 (presumably reinvite) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 0 [ 49]: INVITE sip:+31651977655@172.20.54.10:5060 SIP/2.0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 1 [ 58]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6e97b77b siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 2 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 3 [ 68]: From: "+31332990003" ;tag=as6f2fe75b siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 4 [ 55]: To: ;tag=as75358887 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 5 [ 46]: Contact: siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 6 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 7 [ 16]: CSeq: 103 INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 8 [ 18]: User-Agent: TelBox siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 10 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 12 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:22] Reliably Transmitting (no NAT) to 172.20.54.10:5060: INVITE sip:+31651977655@172.20.54.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6e97b77b Max-Forwards: 70 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Contact: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 103 INVITE User-Agent: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 275 v=0 o=root 126127222 126127223 IN IP4 10.1.1.20 s=Asterisk PBX 1.8.3.2 c=IN IP4 10.1.1.20 t=0 0 m=audio 16384 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #99 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:3089 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] <--- SIP read from UDP:172.20.54.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6e97b77b;received=172.20.20.110 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 103 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <-------------> siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 81]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6e97b77b;received=172.20.20.110 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 68]: From: "+31332990003" ;tag=as6f2fe75b siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 55]: To: ;tag=as75358887 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 103 INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 14]: Server: TelBox siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 26]: Supported: replaces, timer [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 45]: Contact: [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 17]: Content-Length: 0 siphub01*CLI> [Apr 21 14:40:22] --- (11 headers 0 lines) --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 (Checking To) --From tag as6f2fe75b --To-tag as75358887 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3814 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #99 - INVITE (got response) [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3821 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Request 103: Found [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:18792 handle_response_invite: SIP response 100 to RE-invite on outgoing call 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: INVITE [Apr 21 14:40:22] <--- SIP read from UDP:172.20.54.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6e97b77b;received=172.20.20.110 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 103 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 282 v=0 o=root 2051453966 2051453968 IN IP4 172.20.54.10 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.54.10 t=0 0 m=audio 5630 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 81]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6e97b77b;received=172.20.20.110 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 68]: From: "+31332990003" ;tag=as6f2fe75b [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 55]: To: ;tag=as75358887 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 103 INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 14]: Server: TelBox [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 26]: Supported: replaces, timer [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 45]: Contact: [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 29]: Content-Type: application/sdp [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 19]: Content-Length: 282 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 0]: [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 48]: o=root 2051453966 2051453968 IN IP4 172.20.54.10 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 22]: s=Asterisk PBX 1.8.3.2 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 21]: c=IN IP4 172.20.54.10 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 29]: m=audio 5630 RTP/AVP 18 8 101 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 21]: a=rtpmap:18 G729/8000 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 19]: a=fmtp:18 annexb=no [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 15]: a=fmtp:101 0-16 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 11 [ 10]: a=ptime:20 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 12 [ 10]: a=sendrecv [Apr 21 14:40:22] --- (12 headers 13 lines) --- [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 (Checking To) --From tag as6f2fe75b --To-tag as75358887 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3742 __sip_ack: Acked pending invite 103 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' of Request 103: Match Found [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:18792 handle_response_invite: SIP response 200 to RE-invite on outgoing call 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP o=root 2051453966 2051453968 IN IP4 172.20.54.10... UNSUPPORTED. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP s=Asterisk PBX 1.8.3.2... UNSUPPORTED. [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10' gives... [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '(null)'. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP c=IN IP4 172.20.54.10... OK. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [Apr 21 14:40:22] Found RTP audio format 18 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x7fa2e80066a0 [Apr 21 14:40:22] Found RTP audio format 8 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fa2e80066a0 [Apr 21 14:40:22] Found RTP audio format 101 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa2e80066a0 [Apr 21 14:40:22] Found audio description format G729 for ID 18 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Apr 21 14:40:22] Found audio description format PCMA for ID 8 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Apr 21 14:40:22] Found audio description format telephone-event for ID 101 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x7fa2e80066a0 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x7fa2e80066a0 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa2e80066a0 [Apr 21 14:40:22] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729) [Apr 21 14:40:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Apr 21 14:40:22] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xda8138' [Apr 21 14:40:22] Peer audio RTP is at port 172.20.54.10:5630 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0x7fa2e80066a0 to 0xda8300 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0x7fa2e80066a0 to 0xda8300 [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa2e80066a0 to 0xda8300 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8619 process_sdp: We're settling with these formats: 0x108 (alaw|g729) [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8624 process_sdp: We have an owner, now see if we need to change this call [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:5454 update_call_counter: Updating call counter for outgoing call [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:9580 reqprep: Strict routing enforced for session 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 [Apr 21 14:40:22] set_destination: Parsing for address/port to send to [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. [Apr 21 14:40:22] set_destination: set destination to 172.20.54.10:5060 [Apr 21 14:40:22] Transmitting (no NAT) to 172.20.54.10:5060: ACK sip:+31651977655@172.20.54.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK29811832 Max-Forwards: 70 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Contact: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 103 ACK User-Agent: TelBox Content-Length: 0 --- [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'ACK sip:+31' onto UDP socket destined for 172.20.54.10:5060 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] <--- SIP read from UDP:10.51.1.16:5060 ---> <-------------> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] <--- SIP read from UDP:10.1.1.20:5060 ---> ACK sip:00651977655@172.20.20.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-a7432a0f From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 To: ;tag=as2138f877 Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="BoxWares.25",realm="telbox",nonce="57c1f1ab",uri="sip:00651977655@siphub01.boxwares.wan",algorithm=MD5,response="64bc6ac73ef2047faa28e92e231f90a4" Contact: "Jeroen Draadloos" User-Agent: Cisco/WIP310-5.0.13 Content-Length: 0 Allow-Events: dialog <-------------> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 46]: ACK sip:00651977655@172.20.20.110:5060 SIP/2.0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 55]: Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK-a7432a0f [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 87]: From: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 58]: To: ;tag=as2138f877 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 36]: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 13]: CSeq: 102 ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [178]: Authorization: Digest username="BoxWares.25",realm="telbox",nonce="57c1f1ab",uri="sip:00651977655@siphub01.boxwares.wan",algorithm=MD5,response="64bc6ac73ef2047faa28e92e231f90a4" [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 60]: Contact: "Jeroen Draadloos" [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 31]: User-Agent: Cisco/WIP310-5.0.13 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 17]: Content-Length: 0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 20]: Allow-Events: dialog [Apr 21 14:40:22] --- (12 headers 0 lines) --- [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: a8d24c89-98f0a9b8@10.1.1.20 (Checking From) --From tag ba1ed6559922ca5co0 --To-tag as2138f877 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received ACK (6) - Command in SIP ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3747 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #98 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3780 __sip_ack: Stopping retransmission on 'a8d24c89-98f0a9b8@10.1.1.20' of Response 102: Match Found [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:18630 check_pendings: Sending pending reinvite on 'a8d24c89-98f0a9b8@10.1.1.20' [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:9580 reqprep: Strict routing enforced for session a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:22] set_destination: Parsing for address/port to send to [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20:5060' gives... [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '5060'. [Apr 21 14:40:22] set_destination: set destination to 10.1.1.20:5060 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:10570 add_sdp: ** Our capability: 0x100 (g729) Video flag: True Text flag: True [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x0 (nothing) [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:10575 add_sdp: ** Our native-bridge filtered capablity: 0x100 (g729) [Apr 21 14:40:22] Audio is at 5060 [Apr 21 14:40:22] Adding codec 0x100 (g729) to SDP [Apr 21 14:40:22] Adding non-codec 0x1 (telephone-event) to SDP [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:10680 add_sdp: -- Done with adding codecs to SDP [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:10819 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:2821 initialize_initreq: Initializing already initialized SIP dialog a8d24c89-98f0a9b8@10.1.1.20 (presumably reinvite) [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 45]: INVITE sip:BoxWares.25@10.1.1.20:5060 SIP/2.0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 58]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK48a5a674 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 60]: From: ;tag=as2138f877 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 85]: To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 45]: Contact: [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 36]: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 18]: User-Agent: TelBox [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 26]: Supported: replaces, timer [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Apr 21 14:40:22] Reliably Transmitting (no NAT) to 10.1.1.20:5060: INVITE sip:BoxWares.25@10.1.1.20:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK48a5a674 Max-Forwards: 70 From: ;tag=as2138f877 To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 Contact: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 102 INVITE User-Agent: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 258 v=0 o=root 1948114604 1948114606 IN IP4 172.20.54.10 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.54.10 t=0 0 m=audio 5630 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=sendrecv --- [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #100 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 10.1.1.20:5060 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:22] <--- SIP read from UDP:10.1.1.20:5060 ---> SIP/2.0 200 OK To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 From: ;tag=as2138f877 Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 102 INVITE Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK48a5a674 Contact: "Jeroen Draadloos" Server: Cisco/WIP310-5.0.13 Content-Length: 226 Content-Type: application/sdp v=0 o=- 2426510 2426510 IN IP4 10.1.1.20 s=- c=IN IP4 10.1.1.20 t=0 0 m=audio 16384 RTP/AVP 18 101 a=rtpmap:18 G729a/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:40 a=sendrecv <-------------> siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 14]: SIP/2.0 200 OK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 85]: To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 60]: From: ;tag=as2138f877 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 36]: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 16]: CSeq: 102 INVITE [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 58]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK48a5a674 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 60]: Contact: "Jeroen Draadloos" siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 27]: Server: Cisco/WIP310-5.0.13 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 19]: Content-Length: 226 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 29]: Content-Type: application/sdp [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 0]: [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 36]: o=- 2426510 2426510 IN IP4 10.1.1.20 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 3]: s=- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 18]: c=IN IP4 10.1.1.20 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 28]: m=audio 16384 RTP/AVP 18 101 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 22]: a=rtpmap:18 G729a/8000 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 19]: a=fmtp:18 annexb=no siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 15]: a=fmtp:101 0-15 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 10]: a=ptime:40 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 11 [ 10]: a=sendrecv siphub01*CLI> [Apr 21 14:40:22] --- (10 headers 12 lines) --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: a8d24c89-98f0a9b8@10.1.1.20 (Checking To) --From tag as2138f877 --To-tag ba1ed6559922ca5co0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3742 __sip_ack: Acked pending invite 102 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3747 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #100 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3780 __sip_ack: Stopping retransmission on 'a8d24c89-98f0a9b8@10.1.1.20' of Request 102: Match Found siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:18792 handle_response_invite: SIP response 200 to RE-invite on outgoing call a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP o=- 2426510 2426510 IN IP4 10.1.1.20... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP s=-... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '(null)'. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP c=IN IP4 10.1.1.20... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [Apr 21 14:40:22] Found RTP audio format 18 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x7fa2e80066a0 [Apr 21 14:40:22] Found RTP audio format 101 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa2e80066a0 [Apr 21 14:40:22] Found audio description format G729a for ID 18 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729a/8000... OK. [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:22] Found audio description format telephone-event for ID 101 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=ptime:40... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa2e80066a0 [Apr 21 14:40:22] Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xddc7a8' [Apr 21 14:40:22] Peer audio RTP is at port 10.1.1.20:16384 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0x7fa2e80066a0 to 0xddc970 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa2e80066a0 to 0xddc970 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8619 process_sdp: We're settling with these formats: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8624 process_sdp: We have an owner, now see if we need to change this call siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:5454 update_call_counter: Updating call counter for incoming call siphub01*CLI> [Apr 21 14:40:22] DEBUG[1012]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - BoxWares.25 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1012]: chan_sip.c:24969 sip_devicestate: Checking device state for peer BoxWares.25 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1012]: devicestate.c:458 do_state_change: Changing state for SIP/BoxWares.25 - state 2 (In use) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1012]: devicestate.c:438 devstate_event: device 'SIP/BoxWares.25' state '2' siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20:5060' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '5060'. [Apr 21 14:40:22] DEBUG[1036]: app_queue.c:1330 handle_statechange: Device 'SIP/BoxWares.25' changed to state '2' (In use) but we don't care because they're not a member of any queue. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:9580 reqprep: Strict routing enforced for session a8d24c89-98f0a9b8@10.1.1.20 [Apr 21 14:40:22] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20:5060' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '5060'. siphub01*CLI> [Apr 21 14:40:22] set_destination: set destination to 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:22] Transmitting (no NAT) to 10.1.1.20:5060: ACK sip:BoxWares.25@10.1.1.20:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK0a9a8dec Max-Forwards: 70 From: ;tag=as2138f877 To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 Contact: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 102 ACK User-Agent: TelBox Content-Length: 0 --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'ACK sip:Box' onto UDP socket destined for 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: rtp_engine.c:1096 remote_bridge_loop: Oooh, 'SIP/BoxWares.25-00000009' changed end address to 10.1.1.20:16384 (format g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: rtp_engine.c:1099 remote_bridge_loop: Oooh, 'SIP/BoxWares.25-00000009' was 10.1.1.20:16384/(format unknown) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:27692 sip_set_rtp_peer: Sending reinvite on SIP '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' - It's audio soon redirected to IP 10.1.1.20:16384 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:9580 reqprep: Strict routing enforced for session 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:22] set_destination: set destination to 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10570 add_sdp: ** Our capability: 0x108 (alaw|g729) Video flag: True Text flag: True siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10575 add_sdp: ** Our native-bridge filtered capablity: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] Audio is at 5060 siphub01*CLI> [Apr 21 14:40:22] Adding codec 0x100 (g729) to SDP siphub01*CLI> [Apr 21 14:40:22] Adding non-codec 0x1 (telephone-event) to SDP siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10680 add_sdp: -- Done with adding codecs to SDP siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10819 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:2821 initialize_initreq: Initializing already initialized SIP dialog 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 (presumably reinvite) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 0 [ 49]: INVITE sip:+31651977655@172.20.54.10:5060 SIP/2.0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 1 [ 58]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK025ea91f [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 2 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 3 [ 68]: From: "+31332990003" ;tag=as6f2fe75b siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 4 [ 55]: To: ;tag=as75358887 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 5 [ 46]: Contact: siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 6 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 7 [ 16]: CSeq: 104 INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 8 [ 18]: User-Agent: TelBox siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 10 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 12 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:22] Reliably Transmitting (no NAT) to 172.20.54.10:5060: INVITE sip:+31651977655@172.20.54.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK025ea91f Max-Forwards: 70 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Contact: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 104 INVITE User-Agent: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 251 v=0 o=root 126127222 126127224 IN IP4 10.1.1.20 s=Asterisk PBX 1.8.3.2 c=IN IP4 10.1.1.20 t=0 0 m=audio 16384 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=sendrecv --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #101 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:3089 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:22] <--- SIP read from UDP:172.20.54.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK025ea91f;received=172.20.20.110 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 104 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <-------------> siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 81]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK025ea91f;received=172.20.20.110 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 68]: From: "+31332990003" ;tag=as6f2fe75b siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 55]: To: ;tag=as75358887 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 104 INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 14]: Server: TelBox siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 45]: Contact: siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 17]: Content-Length: 0 [Apr 21 14:40:22] --- (11 headers 0 lines) --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 (Checking To) --From tag as6f2fe75b --To-tag as75358887 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3814 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #101 - INVITE (got response) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3821 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Request 104: Found siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:18792 handle_response_invite: SIP response 100 to RE-invite on outgoing call 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK [Apr 21 14:40:22] <--- SIP read from UDP:172.20.54.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK025ea91f;received=172.20.20.110 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 104 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 258 v=0 o=root 2051453966 2051453969 IN IP4 172.20.54.10 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.54.10 t=0 0 m=audio 5630 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=sendrecv <-------------> siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 14]: SIP/2.0 200 OK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 81]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK025ea91f;received=172.20.20.110 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 68]: From: "+31332990003" ;tag=as6f2fe75b siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 55]: To: ;tag=as75358887 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 104 INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 14]: Server: TelBox siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 45]: Contact: [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 19]: Content-Length: 258 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 0]: siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 48]: o=root 2051453966 2051453969 IN IP4 172.20.54.10 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 22]: s=Asterisk PBX 1.8.3.2 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 21]: c=IN IP4 172.20.54.10 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 27]: m=audio 5630 RTP/AVP 18 101 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 21]: a=rtpmap:18 G729/8000 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 19]: a=fmtp:18 annexb=no siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 15]: a=fmtp:101 0-16 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 10]: a=ptime:40 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 11 [ 10]: a=sendrecv siphub01*CLI> [Apr 21 14:40:22] --- (12 headers 12 lines) --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 (Checking To) --From tag as6f2fe75b --To-tag as75358887 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3742 __sip_ack: Acked pending invite 104 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' of Request 104: Match Found siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:18792 handle_response_invite: SIP response 200 to RE-invite on outgoing call 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP o=root 2051453966 2051453969 IN IP4 172.20.54.10... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP s=Asterisk PBX 1.8.3.2... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '(null)'. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP c=IN IP4 172.20.54.10... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [Apr 21 14:40:22] Found RTP audio format 18 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:22] Found RTP audio format 101 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa2e80066a0 [Apr 21 14:40:22] Found audio description format G729 for ID 18 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Apr 21 14:40:22] Found audio description format telephone-event for ID 101 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=ptime:40... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa2e80066a0 [Apr 21 14:40:22] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xda8138' [Apr 21 14:40:22] Peer audio RTP is at port 172.20.54.10:5630 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0x7fa2e80066a0 to 0xda8300 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa2e80066a0 to 0xda8300 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8619 process_sdp: We're settling with these formats: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:8624 process_sdp: We have an owner, now see if we need to change this call siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:5454 update_call_counter: Updating call counter for outgoing call siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:9580 reqprep: Strict routing enforced for session 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:22] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:22] set_destination: set destination to 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:22] Transmitting (no NAT) to 172.20.54.10:5060: ACK sip:+31651977655@172.20.54.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK03a41d9a Max-Forwards: 70 From: "+31332990003" ;tag=as6f2fe75b To: ;tag=as75358887 Contact: Call-ID: 4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060 CSeq: 104 ACK User-Agent: TelBox Content-Length: 0 --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'ACK sip:+31' onto UDP socket destined for 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: rtp_engine.c:1062 remote_bridge_loop: Oooh, 'SIP/_SIP01-0000000a' changed end address to 172.20.54.10:5630 (format g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: rtp_engine.c:1065 remote_bridge_loop: Oooh, 'SIP/_SIP01-0000000a' changed end vaddress to (null) (format g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: rtp_engine.c:1068 remote_bridge_loop: Oooh, 'SIP/_SIP01-0000000a' changed end taddress to (null) (format g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: rtp_engine.c:1071 remote_bridge_loop: Oooh, 'SIP/_SIP01-0000000a' was 172.20.54.10:5630/(format unknown) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: rtp_engine.c:1074 remote_bridge_loop: Oooh, 'SIP/_SIP01-0000000a' was (null)/(format unknown) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: rtp_engine.c:1077 remote_bridge_loop: Oooh, 'SIP/_SIP01-0000000a' was (null)/(format unknown) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:27692 sip_set_rtp_peer: Sending reinvite on SIP 'a8d24c89-98f0a9b8@10.1.1.20' - It's audio soon redirected to IP 172.20.54.10:5630 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:9580 reqprep: Strict routing enforced for session a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:22] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20:5060' gives... siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '5060'. siphub01*CLI> [Apr 21 14:40:22] set_destination: set destination to 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10570 add_sdp: ** Our capability: 0x100 (g729) Video flag: True Text flag: True siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x0 (nothing) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10575 add_sdp: ** Our native-bridge filtered capablity: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] Audio is at 5060 siphub01*CLI> [Apr 21 14:40:22] Adding codec 0x100 (g729) to SDP siphub01*CLI> [Apr 21 14:40:22] Adding non-codec 0x1 (telephone-event) to SDP siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10680 add_sdp: -- Done with adding codecs to SDP siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:10819 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:2821 initialize_initreq: Initializing already initialized SIP dialog a8d24c89-98f0a9b8@10.1.1.20 (presumably reinvite) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 0 [ 45]: INVITE sip:BoxWares.25@10.1.1.20:5060 SIP/2.0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 1 [ 58]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6f469c72 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 2 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 3 [ 60]: From: ;tag=as2138f877 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 4 [ 85]: To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 5 [ 45]: Contact: siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 6 [ 36]: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 7 [ 16]: CSeq: 103 INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 8 [ 18]: User-Agent: TelBox siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 10 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:7816 parse_request: Header 12 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:22] Reliably Transmitting (no NAT) to 10.1.1.20:5060: INVITE sip:BoxWares.25@10.1.1.20:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6f469c72 Max-Forwards: 70 From: ;tag=as2138f877 To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 Contact: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 103 INVITE User-Agent: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 258 v=0 o=root 1948114604 1948114607 IN IP4 172.20.54.10 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.54.10 t=0 0 m=audio 5630 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=sendrecv --- siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #102 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1092]: chan_sip.c:3089 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:22] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:23] <--- SIP read from UDP:10.1.1.20:5060 ---> SIP/2.0 200 OK To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 From: ;tag=as2138f877 Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 103 INVITE Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6f469c72 Contact: "Jeroen Draadloos" Server: Cisco/WIP310-5.0.13 Content-Length: 226 Content-Type: application/sdp v=0 o=- 2426520 2426520 IN IP4 10.1.1.20 s=- c=IN IP4 10.1.1.20 t=0 0 m=audio 16384 RTP/AVP 18 101 a=rtpmap:18 G729a/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:40 a=sendrecv <-------------> siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 14]: SIP/2.0 200 OK siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 85]: To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 60]: From: ;tag=as2138f877 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 36]: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 16]: CSeq: 103 INVITE siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 58]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK6f469c72 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 60]: Contact: "Jeroen Draadloos" siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 27]: Server: Cisco/WIP310-5.0.13 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 19]: Content-Length: 226 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 0]: siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 36]: o=- 2426520 2426520 IN IP4 10.1.1.20 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 3]: s=- siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 18]: c=IN IP4 10.1.1.20 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 28]: m=audio 16384 RTP/AVP 18 101 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 22]: a=rtpmap:18 G729a/8000 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 19]: a=fmtp:18 annexb=no siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 15]: a=fmtp:101 0-15 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 10]: a=ptime:40 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 11 [ 10]: a=sendrecv siphub01*CLI> [Apr 21 14:40:23] --- (10 headers 12 lines) --- siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: a8d24c89-98f0a9b8@10.1.1.20 (Checking To) --From tag as2138f877 --To-tag ba1ed6559922ca5co0 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:3742 __sip_ack: Acked pending invite 103 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:3747 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #102 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:3780 __sip_ack: Stopping retransmission on 'a8d24c89-98f0a9b8@10.1.1.20' of Request 103: Match Found siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:18792 handle_response_invite: SIP response 200 to RE-invite on outgoing call a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP o=- 2426520 2426520 IN IP4 10.1.1.20... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP s=-... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20' gives... siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '(null)'. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP c=IN IP4 10.1.1.20... OK. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:23] Found RTP audio format 18 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:23] Found RTP audio format 101 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:23] Found audio description format G729a for ID 18 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729a/8000... OK. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:23] Found audio description format telephone-event for ID 101 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=ptime:40... OK. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:23] Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) siphub01*CLI> [Apr 21 14:40:23] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xddc7a8' siphub01*CLI> [Apr 21 14:40:23] Peer audio RTP is at port 10.1.1.20:16384 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0x7fa2e80066a0 to 0xddc970 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa2e80066a0 to 0xddc970 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8619 process_sdp: We're settling with these formats: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:8624 process_sdp: We have an owner, now see if we need to change this call siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:5454 update_call_counter: Updating call counter for incoming call siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20:5060' gives... siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '5060'. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1012]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - BoxWares.25 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1012]: chan_sip.c:24969 sip_devicestate: Checking device state for peer BoxWares.25 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1012]: devicestate.c:458 do_state_change: Changing state for SIP/BoxWares.25 - state 2 (In use) siphub01*CLI> [Apr 21 14:40:23] DEBUG[1012]: devicestate.c:438 devstate_event: device 'SIP/BoxWares.25' state '2' siphub01*CLI> [Apr 21 14:40:23] DEBUG[1036]: app_queue.c:1330 handle_statechange: Device 'SIP/BoxWares.25' changed to state '2' (In use) but we don't care because they're not a member of any queue. siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:9580 reqprep: Strict routing enforced for session a8d24c89-98f0a9b8@10.1.1.20 siphub01*CLI> [Apr 21 14:40:23] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.20:5060' gives... siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.20' and port '5060'. siphub01*CLI> [Apr 21 14:40:23] set_destination: set destination to 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:23] Transmitting (no NAT) to 10.1.1.20:5060: ACK sip:BoxWares.25@10.1.1.20:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK112b74ec Max-Forwards: 70 From: ;tag=as2138f877 To: "Jeroen Draadloos" ;tag=ba1ed6559922ca5co0 Contact: Call-ID: a8d24c89-98f0a9b8@10.1.1.20 CSeq: 103 ACK User-Agent: TelBox Content-Length: 0 --- siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'ACK sip:Box' onto UDP socket destined for 10.1.1.20:5060 siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:23] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:24] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:24] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:24] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:24] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:25] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:25] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:25] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:25] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:26] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:26] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:26] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:26] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:27] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:27] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:27] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:27] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:27] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:27] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:27] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:27] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:28] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:28] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:28] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:28] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:29] <--- SIP read from UDP:10.1.1.41:5060 ---> REGISTER sip:siphub01.boxwares.wan SIP/2.0 Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bK5ce900fab396beb2655f275cbe16c6f1;rport From: "Siemens_Test_1" ;tag=585578558 To: "Siemens_Test_1" Call-ID: 326654607@10_1_1_41 CSeq: 874 REGISTER Contact: Max-Forwards: 70 User-Agent: A580 IP021920000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 42]: REGISTER sip:siphub01.boxwares.wan SIP/2.0 siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 84]: Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bK5ce900fab396beb2655f275cbe16c6f1;rport siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 76]: From: "Siemens_Test_1" ;tag=585578558 siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 60]: To: "Siemens_Test_1" siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 28]: Call-ID: 326654607@10_1_1_41 siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 18]: CSeq: 874 REGISTER siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 41]: Contact: siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 31]: User-Agent: A580 IP021920000000 siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 12]: Expires: 180 siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 17]: Content-Length: 0 [Apr 21 14:40:29] --- (12 headers 0 lines) --- siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 326654607@10_1_1_41 (Checking From) --From tag 585578558 --To-tag siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: acl.c:715 ast_ouraddrfor: For destination '10.1.1.41', our source address is '172.20.20.110'. siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 326654607@10_1_1_41 - REGISTER (No RTP) siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:23480 handle_request_register: Initializing initreq for method REGISTER - callid 326654607@10_1_1_41 siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.41:5060' gives... siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.41' and port '5060'. [Apr 21 14:40:29] Sending to 10.1.1.41:5060 (no NAT) siphub01*CLI> [Apr 21 14:40:29] <--- Transmitting (no NAT) to 10.1.1.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bK5ce900fab396beb2655f275cbe16c6f1;received=10.1.1.41;rport=5060 From: "Siemens_Test_1" ;tag=585578558 To: "Siemens_Test_1" ;tag=as70b6a7c8 Call-ID: 326654607@10_1_1_41 CSeq: 874 REGISTER Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH S siphub01*CLI> upported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="telbox", nonce="0492fe6b" Content-Length: 0 <------------> siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.1.1.41:5060 siphub01*CLI> [Apr 21 14:40:29] Scheduling destruction of SIP dialog '326654607@10_1_1_41' in 32000 ms (Method: REGISTER) siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:29] <--- SIP read from UDP:10.1.1.41:5060 ---> REGISTER sip:siphub01.boxwares.wan SIP/2.0 Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bK91886761bef7923579ccf4570826c2;rport From: "Siemens_Test_1" ;tag=585578558 To: "Siemens_Test_1" Call-ID: 326654607@10_1_1_41 CSeq: 875 REGISTER Contact: Authorization: Digest username="BoxWares.51", realm="telbox", algorithm=MD5, uri="sip:siphub01.boxwares.wan", nonce="0492fe6b", response="3d1725ee0165f373977145a5f3c83819" Max-Forwards: 70 User-Agent: A580 IP021920000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 42]: REGISTER sip:siphub01.boxwares.wan SIP/2.0 [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 82]: Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bK91886761bef7923579ccf4570826c2;rport [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 76]: From: "Siemens_Test_1" ;tag=585578558 [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 60]: To: "Siemens_Test_1" [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 28]: Call-ID: 326654607@10_1_1_41 [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 18]: CSeq: 875 REGISTER [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 41]: Contact: siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [171]: Authorization: Digest username="BoxWares.51", realm="telbox", algorithm=MD5, uri="sip:siphub01.boxwares.wan", nonce="0492fe6b", response="3d1725ee0165f373977145a5f3c83819" [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 16]: Max-Forwards: 70 [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 31]: User-Agent: A580 IP021920000000 [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 12]: Expires: 180 [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 72]: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 17]: Content-Length: 0 [Apr 21 14:40:29] --- (13 headers 0 lines) --- [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 326654607@10_1_1_41 (Checking From) --From tag 585578558 --To-tag siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting 'siphub01.boxwares.wan' gives... [Apr 21 14:40:29] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host 'siphub01.boxwares.wan' and port '(null)'. [Apr 21 14:40:29] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting 'siphub01.boxwares.wan' gives... [Apr 21 14:40:29] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host 'siphub01.boxwares.wan' and port '(null)'. [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:23480 handle_request_register: Initializing initreq for method REGISTER - callid 326654607@10_1_1_41 siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.41:5060' gives... [Apr 21 14:40:29] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.41' and port '5060'. [Apr 21 14:40:29] Sending to 10.1.1.41:5060 (no NAT) [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:13089 parse_register_contact: Store REGISTER's Contact header for call routing. [Apr 21 14:40:29] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.41:5060' gives... [ siphub01*CLI> Apr 21 14:40:29] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.41' and port '5060'. [Apr 21 14:40:29] > Saved useragent "A580 IP021920000000" for peer BoxWares.51 siphub01*CLI> [Apr 21 14:40:29] <--- Transmitting (no NAT) to 10.1.1.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bK91886761bef7923579ccf4570826c2;received=10.1.1.41;rport=5060 From: "Siemens_Test_1" ;tag=585578558 To: "Siemens_Test_1" ;tag=as70b6a7c8 Call-ID: 326654607@10_1_1_41 CSeq: 875 REGISTER Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 180 Contact: ;expires=180 Date: Thu, 21 Apr 2011 12:40:29 GMT Content-Length: 0 <------------> siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.1.1.41:5060 [Apr 21 14:40:29] DEBUG[1012]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - BoxWares.51 [Apr 21 14:40:29] Scheduling destruction of SIP dialog '326654607@10_1_1_41' in 32000 ms (Method: REGISTER) siphub01*CLI> [Apr 21 14:40:29] DEBUG[1012]: chan_sip.c:24969 sip_devicestate: Checking device state for peer BoxWares.51 [Apr 21 14:40:29] DEBUG[1012]: devicestate.c:458 do_state_change: Changing state for SIP/BoxWares.51 - state 1 (Not in use) siphub01*CLI> [Apr 21 14:40:29] DEBUG[1012]: devicestate.c:438 devstate_event: device 'SIP/BoxWares.51' state '1' siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:29] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:29] DEBUG[1036]: app_queue.c:1330 handle_statechange: Device 'SIP/BoxWares.51' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. siphub01*CLI> [Apr 21 14:40:30] <--- SIP read from UDP:10.1.1.41:5060 ---> REGISTER sip:siphub01.boxwares.wan SIP/2.0 Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bKdac82eaa79fb58e8f92bf9805bb077b;rport From: "Siemens_Test_2" ;tag=1855426586 To: "Siemens_Test_2" Call-ID: 3651355513@10_1_1_41 CSeq: 855 REGISTER Contact: Max-Forwards: 70 User-Agent: A580 IP021920000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 42]: REGISTER sip:siphub01.boxwares.wan SIP/2.0 [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 83]: Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bKdac82eaa79fb58e8f92bf9805bb077b;rport [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 77]: From: "Siemens_Test_2" ;tag=1855426586 [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 60]: To: "Siemens_Test_2" [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 29]: Call-ID: 3651355513@10_1_1_41 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 18]: CSeq: 855 REGISTER [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 41]: Contact: [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 31]: User-Agent: A580 IP021920000000 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 12]: Expires: 180 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 17]: Content-Length: 0 siphub01*CLI> [Apr 21 14:40:30] --- (12 headers 0 lines) --- [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 3651355513@10_1_1_41 (Checking From) --From tag 1855426586 --To-tag siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: acl.c:715 ast_ouraddrfor: For destination '10.1.1.41', our source address is '172.20.20.110'. siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 3651355513@10_1_1_41 - REGISTER (No RTP) siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:23480 handle_request_register: Initializing initreq for method REGISTER - callid 3651355513@10_1_1_41 [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.41:5060' gives... [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.41' and port '5060'. siphub01*CLI> [Apr 21 14:40:30] Sending to 10.1.1.41:5060 (no NAT) siphub01*CLI> [Apr 21 14:40:30] <--- Transmitting (no NAT) to 10.1.1.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bKdac82eaa79fb58e8f92bf9805bb077b;received=10.1.1.41;rport=5060 From: "Siemens_Test_2" ;tag=1855426586 To: "Siemens_Test_2" ;tag=as2288a2ab Call-ID: 3651355513@10_1_1_41 CSeq: 855 REGISTER Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="telbox", nonce="655d3d66" Content-Length: 0 <------------> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.1.1.41:5060 [Apr 21 14:40:30] Scheduling destruction of SIP dialog '3651355513@10_1_1_41' in 32000 ms (Method: REGISTER) [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:30] <--- SIP read from UDP:10.1.1.41:5060 ---> REGISTER sip:siphub01.boxwares.wan SIP/2.0 Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bKb0ff917e529c277c87b66736591a0f6d;rport From: "Siemens_Test_2" ;tag=1855426586 To: "Siemens_Test_2" Call-ID: 3651355513@10_1_1_41 CSeq: 856 REGISTER Contact: Authorization: Digest username="BoxWares.52", realm="telbox", algorithm=MD5, uri="sip:siphub01.boxwares.wan", nonce="655d3d66", response="9bfe41e212cb63d64c67381eb40573bd" Max-Forwards: 70 User-Agent: A580 IP021920000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 42]: REGISTER sip:siphub01.boxwares.wan SIP/2.0 [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 84]: Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bKb0ff917e529c277c87b66736591a0f6d;rport [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 77]: From: "Siemens_Test_2" ;tag=1855426586 [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 60]: To: "Siemens_Test_2" siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 29]: Call-ID: 3651355513@10_1_1_41 [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 18]: CSeq: 856 REGISTER siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 41]: Contact: [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [171]: Authorization: Digest username="BoxWares.52", realm="telbox", algorithm=MD5, uri="sip:siphub01.boxwares.wan", nonce="655d3d66", response="9bfe41e212cb63d64c67381eb40573bd" siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 16]: Max-Forwards: 70 [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 31]: User-Agent: A580 IP021920000000 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 12]: Expires: 180 [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 72]: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 17]: Content-Length: 0 [Apr 21 14:40:30] --- (13 headers 0 lines) --- [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 3651355513@10_1_1_41 (Checking From) --From tag 1855426586 --To-tag siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting 'siphub01.boxwares.wan' gives... [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host 'siphub01.boxwares.wan' and port '(null)'. [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting 'siphub01.boxwares.wan' gives... [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host 'siphub01.boxwares.wan' and port '(null)'. siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:23480 handle_request_register: Initializing initreq for method REGISTER - callid 3651355513@10_1_1_41 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.41:5060' gives... siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.41' and port '5060'. [Apr 21 14:40:30] Sending to 10.1.1.41:5060 (no NAT) siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:13089 parse_register_contact: Store REGISTER's Contact header for call routing. siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.1.1.41:5060' gives... siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.1.1.41' and port '5060'. [Apr 21 14:40:30] > Saved useragent "A580 IP021920000000" for peer BoxWares.52 siphub01*CLI> [Apr 21 14:40:30] <--- Transmitting (no NAT) to 10.1.1.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.41:5060;branch=z9hG4bKb0ff917e529c277c87b66736591a0f6d;received=10.1.1.41;rport=5060 From: "Siemens_Test_2" ;tag=1855426586 To: "Siemens_Test_2" ;tag=as2288a2ab Call-ID: 3651355513@10_1_1_41 CSeq: 856 REGISTER Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 180 Contact: ;expires=180 Date: Thu, 21 Apr 2011 12:40:30 GMT Content-Length: 0 <------------> siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.1.1.41:5060 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1012]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for SIP - BoxWares.52 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1012]: chan_sip.c:24969 sip_devicestate: Checking device state for peer BoxWares.52 siphub01*CLI> [Apr 21 14:40:30] DEBUG[1012]: devicestate.c:458 do_state_change: Changing state for SIP/BoxWares.52 - state 1 (Not in use) siphub01*CLI> [Apr 21 14:40:30] DEBUG[1012]: devicestate.c:438 devstate_event: device 'SIP/BoxWares.52' state '1' [Apr 21 14:40:30] Scheduling destruction of SIP dialog '3651355513@10_1_1_41' in 32000 ms (Method: REGISTER) siphub01*CLI> [Apr 21 14:40:30] DEBUG[1036]: app_queue.c:1330 handle_statechange: Device 'SIP/BoxWares.52' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:30] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:31] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:31] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:31] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:31] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:32] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK [Apr 21 14:40:32] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:32] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE [Apr 21 14:40:32] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:33] <--- SIP read from UDP:10.51.1.10:5060 ---> INVITE sip:00888040220@172.20.20.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.51.1.10:5060;branch=z9hG4bK110517687;rport From: ;tag=1450788815 To: "00888040220" ;tag=as673b2bc9 Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 CSeq: 103 INVITE Contact: X-Grandstream-PBX: true Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2120 1.0.1.66 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 402 v=0 o=BoxWares.18 8001 8003 IN IP4 10.51.1.10 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 5008 RTP/AVP 0 8 4 18 9 97 2 101 a=sendonly a=rtpmap:0 PCMU/8000 a=ptime:40 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 49]: INVITE sip:00888040220@172.20.20.110:5060 SIP/2.0 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 62]: Via: SIP/2.0/UDP 10.51.1.10:5060;branch=z9hG4bK110517687;rport siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 54]: From: ;tag=1450788815 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 64]: To: "00888040220" ;tag=as673b2bc9 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 103 INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 42]: Contact: [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 23]: X-Grandstream-PBX: true [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 16]: Max-Forwards: 70 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 32]: Supported: replaces, path, timer [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 40]: User-Agent: Grandstream GXP2120 1.0.1.66 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 13 [ 47]: Accept: application/sdp, application/dtmf-relay [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 14 [ 19]: Content-Length: 402 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 15 [ 0]: [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 41]: o=BoxWares.18 8001 8003 IN IP4 10.51.1.10 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 10]: s=SIP Call [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 16]: c=IN IP4 0.0.0.0 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 40]: m=audio 5008 RTP/AVP 0 8 4 18 9 97 2 101 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 10]: a=sendonly [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 10]: a=ptime:40 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 20]: a=rtpmap:4 G723/8000 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 11 [ 21]: a=rtpmap:18 G729/8000 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 12 [ 19]: a=fmtp:18 annexb=no [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 13 [ 20]: a=rtpmap:9 G722/8000 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 14 [ 21]: a=rtpmap:97 iLBC/8000 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 15 [ 17]: a=fmtp:97 mode=30 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 16 [ 23]: a=rtpmap:2 G726-32/8000 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 17 [ 33]: a=rtpmap:101 telephone-event/8000 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 18 [ 15]: a=fmtp:101 0-15 [Apr 21 14:40:33] --- (15 headers 19 lines) --- siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 (Checking From) --From tag 1450788815 --To-tag as673b2bc9 [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Apr 21 14:40:33] DEBUG[1021]: sip/reqresp_parser.c:1584 parse_sip_options: Begin: parsing SIP "Supported: replaces, path, timer" [Apr 21 14:40:33] DEBUG[1021]: sip/reqresp_parser.c:1600 parse_sip_options: Found SIP option: -replaces- [Apr 21 14:40:33] DEBUG[1021]: sip/reqresp_parser.c:1608 parse_sip_options: Matched SIP option: replaces siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: sip/reqresp_parser.c:1600 parse_sip_options: Found SIP option: -path- siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: sip/reqresp_parser.c:1608 parse_sip_options: Matched SIP option: path siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: sip/reqresp_parser.c:1600 parse_sip_options: Found SIP option: -timer- siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: sip/reqresp_parser.c:1608 parse_sip_options: Matched SIP option: timer siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.51.1.10:5060' gives... siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.51.1.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:33] Sending to 10.51.1.10:5060 (no NAT) siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:21172 handle_request_invite: Initializing initreq for method INVITE - callid 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP o=BoxWares.18 8001 8003 IN IP4 10.51.1.10... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP s=SIP Call... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '0.0.0.0' gives... siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '0.0.0.0' and port '(null)'. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP c=IN IP4 0.0.0.0... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 0 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 8 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 4 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 4 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 18 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 9 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 97 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 97 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 2 siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 101 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=sendonly... OK. siphub01*CLI> [Apr 21 14:40:33] Found audio description format PCMU for ID 0 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=ptime:40... OK. siphub01*CLI> [Apr 21 14:40:33] Found audio description format PCMA for ID 8 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. siphub01*CLI> [Apr 21 14:40:33] Found audio description format G723 for ID 4 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:4 G723/8000... OK. siphub01*CLI> [Apr 21 14:40:33] Found audio description format G729 for ID 18 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] Found audio description format G722 for ID 9 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. siphub01*CLI> [Apr 21 14:40:33] Found audio description format iLBC for ID 97 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:97 iLBC/8000... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:97 mode=30... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] Found audio description format G726-32 for ID 2 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK. siphub01*CLI> [Apr 21 14:40:33] Found audio description format telephone-event for ID 101 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 2 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 4 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 9 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 97 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa2e8006560 siphub01*CLI> [Apr 21 14:40:33] Capabilities: us - 0x100 (g729), peer - audio=0x1d0d (g723|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) siphub01*CLI> [Apr 21 14:40:33] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xd9c278' siphub01*CLI> [Apr 21 14:40:33] Peer audio RTP is at port 0.0.0.0:5008 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x7fa2e8006560 to 0xd9c440 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 2 from 0x7fa2e8006560 to 0xd9c440 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 4 from 0x7fa2e8006560 to 0xd9c440 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0x7fa2e8006560 to 0xd9c440 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa2e8006560 to 0xd9c440 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0x7fa2e8006560 to 0xd9c440 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 97 from 0x7fa2e8006560 to 0xd9c440 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa2e8006560 to 0xd9c440 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8619 process_sdp: We're settling with these formats: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8624 process_sdp: We have an owner, now see if we need to change this call siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xd9c278' siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:21406 handle_request_invite: Got a SIP re-invite for call 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:21675 handle_request_invite: SIP/BoxWares.18-00000003: This call is UP.... siphub01*CLI> [Apr 21 14:40:33] <--- Transmitting (no NAT) to 10.51.1.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.51.1.10:5060;branch=z9hG4bK110517687;received=10.51.1.10;rport=5060 From: ;tag=1450788815 To: "00888040220" ;tag=as673b2bc9 Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 CSeq: 103 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.51.1.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:10924 transmit_response_with_sdp: Setting framing from config on incoming call siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:10570 add_sdp: ** Our capability: 0x100 (g729) Video flag: True Text flag: True siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:33] Audio is at 5060 siphub01*CLI> [Apr 21 14:40:33] Adding codec 0x100 (g729) to SDP siphub01*CLI> [Apr 21 14:40:33] Adding non-codec 0x1 (telephone-event) to SDP siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:10680 add_sdp: -- Done with adding codecs to SDP siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:10819 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:33] <--- Reliably Transmitting (no NAT) to 10.51.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.51.1.10:5060;branch=z9hG4bK110517687;received=10.51.1.10;rport=5060 From: ;tag=1450788815 To: "00888040220" ;tag=as673b2bc9 Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 CSeq: 103 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 468126143 468126146 IN IP4 172.20.54.10 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.54.10 t=0 0 m=audio 24984 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=recvonly <------------> siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #109 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.51.1.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:27692 sip_set_rtp_peer: Sending reinvite on SIP '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' - It's audio soon redirected to IP 172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:9580 reqprep: Strict routing enforced for session 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:33] set_destination: set destination to 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:10570 add_sdp: ** Our capability: 0x100 (g729) Video flag: True Text flag: True siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:10571 add_sdp: ** Our prefcodec: 0x0 (nothing) siphub01*CLI> [Apr 21 14:40:33] Audio is at 5060 siphub01*CLI> [Apr 21 14:40:33] Adding codec 0x100 (g729) to SDP siphub01*CLI> [Apr 21 14:40:33] Adding non-codec 0x1 (telephone-event) to SDP siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:10680 add_sdp: -- Done with adding codecs to SDP siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:10819 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:2821 initialize_initreq: Initializing already initialized SIP dialog 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 (presumably reinvite) siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 0 [ 48]: INVITE sip:00888040220@172.20.54.10:5060 SIP/2.0 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 1 [ 58]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7c345304 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 2 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 3 [ 58]: From: ;tag=as16ee21af siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 4 [ 63]: To: "00888040220" ;tag=as35b0b2c7 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 5 [ 46]: Contact: siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 6 [ 59]: Call-ID: 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 7 [ 16]: CSeq: 103 INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 8 [ 18]: User-Agent: TelBox siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 10 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:7816 parse_request: Header 12 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:33] Reliably Transmitting (no NAT) to 172.20.54.10:5060: INVITE sip:00888040220@172.20.54.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7c345304 Max-Forwards: 70 From: ;tag=as16ee21af To: "00888040220" ;tag=as35b0b2c7 Contact: Call-ID: 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 CSeq: 103 INVITE User-Agent: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 261 v=0 o=root 1207819696 1207819698 IN IP4 172.20.20.110 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.20.110 t=0 0 m=audio 22542 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=sendrecv --- siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #110 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: chan_sip.c:3089 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update siphub01*CLI> [Apr 21 14:40:33] -- Started music on hold, class 'default', on SIP/_SIP01-00000000 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: channel.c:3402 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:33] <--- SIP read from UDP:172.20.54.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7c345304;received=172.20.20.110 From: ;tag=as16ee21af To: "00888040220" ;tag=as35b0b2c7 Call-ID: 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 CSeq: 103 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <-------------> siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 81]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7c345304;received=172.20.20.110 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 58]: From: ;tag=as16ee21af siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 63]: To: "00888040220" ;tag=as35b0b2c7 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 59]: Call-ID: 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 103 INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 14]: Server: TelBox siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 44]: Contact: siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 17]: Content-Length: 0 siphub01*CLI> [Apr 21 14:40:33] --- (11 headers 0 lines) --- siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 (Checking To) --From tag as16ee21af --To-tag as35b0b2c7 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3814 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #110 - INVITE (got response) siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3821 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Request 103: Found siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:18792 handle_response_invite: SIP response 100 to RE-invite on outgoing call 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:33] <--- SIP read from UDP:172.20.54.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7c345304;received=172.20.20.110 From: ;tag=as16ee21af To: "00888040220" ;tag=as35b0b2c7 Call-ID: 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 CSeq: 103 INVITE Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 1886561017 1886561019 IN IP4 172.20.54.10 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.20.54.10 t=0 0 m=audio 24984 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:40 a=sendrecv <-------------> siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 14]: SIP/2.0 200 OK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 81]: Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK7c345304;received=172.20.20.110 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 58]: From: ;tag=as16ee21af siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 63]: To: "00888040220" ;tag=as35b0b2c7 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 59]: Call-ID: 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 16]: CSeq: 103 INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 14]: Server: TelBox siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 26]: Supported: replaces, timer siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 44]: Contact: siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 29]: Content-Type: application/sdp siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 19]: Content-Length: 259 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 0]: siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 0 [ 3]: v=0 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 1 [ 48]: o=root 1886561017 1886561019 IN IP4 172.20.54.10 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 2 [ 22]: s=Asterisk PBX 1.8.3.2 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 3 [ 21]: c=IN IP4 172.20.54.10 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 4 [ 5]: t=0 0 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 5 [ 28]: m=audio 24984 RTP/AVP 18 101 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 6 [ 21]: a=rtpmap:18 G729/8000 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 7 [ 19]: a=fmtp:18 annexb=no siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 9 [ 15]: a=fmtp:101 0-16 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Body 10 [ 10]: a=ptime:40 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7853 parse_request: Body 11 [ 10]: a=sendrecv siphub01*CLI> [Apr 21 14:40:33] --- (12 headers 12 lines) --- siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 (Checking To) --From tag as16ee21af --To-tag as35b0b2c7 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3742 __sip_ack: Acked pending invite 103 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' of Request 103: Match Found siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:18792 handle_response_invite: SIP response 200 to RE-invite on outgoing call 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP o=root 1886561017 1886561019 IN IP4 172.20.54.10... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP s=Asterisk PBX 1.8.3.2... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10' gives... siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '(null)'. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP c=IN IP4 172.20.54.10... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8200 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 18 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:33] Found RTP audio format 101 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:535 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:33] Found audio description format G729 for ID 18 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] Found audio description format telephone-event for ID 101 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=ptime:40... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8387 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:638 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x7fa2e80066a0 siphub01*CLI> [Apr 21 14:40:33] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) siphub01*CLI> [Apr 21 14:40:33] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xd892a8' siphub01*CLI> [Apr 21 14:40:33] Peer audio RTP is at port 172.20.54.10:24984 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0x7fa2e80066a0 to 0xd89470 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa2e80066a0 to 0xd89470 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8609 process_sdp: Peer doesn't provide T.38 UDPTL siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8619 process_sdp: We're settling with these formats: 0x100 (g729) siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:8624 process_sdp: We have an owner, now see if we need to change this call siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:5454 update_call_counter: Updating call counter for incoming call siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:9580 reqprep: Strict routing enforced for session 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:33] set_destination: set destination to 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] Transmitting (no NAT) to 172.20.54.10:5060: ACK sip:00888040220@172.20.54.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK38ebe9a6 Max-Forwards: 70 From: ;tag=as16ee21af To: "00888040220" ;tag=as35b0b2c7 Contact: Call-ID: 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 CSeq: 103 ACK User-Agent: TelBox Content-Length: 0 --- siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'ACK sip:008' onto UDP socket destined for 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: channel.c:5018 set_format: Set channel SIP/_SIP01-00000000 to write format slin siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_musiconhold.c:337 ast_moh_files_next: SIP/_SIP01-00000000 Opened file 0 '/var/lib/asterisk/moh/macroform-the_simplicity' siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1239 ast_rtp_write: Ooh, format changed from unknown to g729 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1270 ast_rtp_write: Created smoother: format: g729 ms: 40 len: 40 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1140 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0xd892a8' siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] <--- SIP read from UDP:10.51.1.10:5060 ---> ACK sip:00888040220@172.20.20.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.51.1.10:5060;branch=z9hG4bK1551434639;rport From: ;tag=1450788815 To: "00888040220" ;tag=as673b2bc9 Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 CSeq: 103 ACK Contact: X-Grandstream-PBX: true Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2120 1.0.1.66 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 46]: ACK sip:00888040220@172.20.20.110:5060 SIP/2.0 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 10.51.1.10:5060;branch=z9hG4bK1551434639;rport siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 54]: From: ;tag=1450788815 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 64]: To: "00888040220" ;tag=as673b2bc9 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 13]: CSeq: 103 ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 42]: Contact: siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 23]: X-Grandstream-PBX: true siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 32]: Supported: replaces, path, timer siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 40]: User-Agent: Grandstream GXP2120 1.0.1.66 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 17]: Content-Length: 0 siphub01*CLI> [Apr 21 14:40:33] --- (13 headers 0 lines) --- siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 (Checking From) --From tag 1450788815 --To-tag as673b2bc9 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received ACK (6) - Command in SIP ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3747 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #109 siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:3780 __sip_ack: Stopping retransmission on '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' of Response 103: Match Found siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:33] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:33] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:34] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:34] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:34] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:34] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:35] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:35] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:35] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:35] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 64 bytes siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060' Method: ACK siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog '4924f7ae03bb8cf61bc1dd737a690b82@172.20.20.110:5060' Method: INVITE siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:15937 dialog_needdestroy: Bridge still active. Delaying destroy of SIP dialog 'a8d24c89-98f0a9b8@10.1.1.20' Method: ACK siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: res_rtp_asterisk.c:1949 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped siphub01*CLI> [Apr 21 14:40:36] <--- SIP read from UDP:10.51.1.10:5060 ---> REFER sip:00888040220@172.20.20.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.51.1.10:5060;branch=z9hG4bK2011008319;rport From: ;tag=1450788815 To: "00888040220" ;tag=as673b2bc9 Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 CSeq: 104 REFER Contact: X-Grandstream-PBX: true Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2120 1.0.1.66 Refer-To: Referred-By: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 0 [ 48]: REFER sip:00888040220@172.20.20.110:5060 SIP/2.0 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 10.51.1.10:5060;branch=z9hG4bK2011008319;rport siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 2 [ 54]: From: ;tag=1450788815 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 3 [ 64]: To: "00888040220" ;tag=as673b2bc9 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 4 [ 60]: Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 5 [ 15]: CSeq: 104 REFER siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 6 [ 42]: Contact: siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 7 [ 23]: X-Grandstream-PBX: true siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 8 [ 16]: Max-Forwards: 70 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 9 [ 32]: Supported: replaces, path, timer siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 10 [ 40]: User-Agent: Grandstream GXP2120 1.0.1.66 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 11 [ 40]: Refer-To: siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 12 [ 52]: Referred-By: siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 13 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7816 parse_request: Header 14 [ 17]: Content-Length: 0 siphub01*CLI> [Apr 21 14:40:36] --- (15 headers 0 lines) --- siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:7414 find_call: = Looking for Call ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 (Checking From) --From tag 1450788815 --To-tag as673b2bc9 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:23649 handle_incoming: **** Received REFER (9) - Command in SIP REFER siphub01*CLI> [Apr 21 14:40:36] Call 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 got a SIP call transfer from caller: (REFER)! siphub01*CLI> [Apr 21 14:40:36] SIP transfer to extension 16@BoxWares by BoxWares.18@siphub01.boxwares.wan siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:22126 handle_request_refer: SIP blind transfer: Transferer channel SIP/BoxWares.18-00000003, transferee channel SIP/_SIP01-00000000 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:22142 handle_request_refer: Got SIP transfer, applying to bridged peer 'SIP/_SIP01-00000000' siphub01*CLI> [Apr 21 14:40:36] <--- Transmitting (no NAT) to 10.51.1.10:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.51.1.10:5060;branch=z9hG4bK2011008319;received=10.51.1.10;rport=5060 From: ;tag=1450788815 To: "00888040220" ;tag=as673b2bc9 Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 CSeq: 104 REFER Server: TelBox Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer siphub01*CLI> Contact: Content-Length: 0 <------------> siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 202' onto UDP socket destined for 10.51.1.10:5060 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:22207 handle_request_refer: chan1->name: SIP/BoxWares.18-00000003 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:9580 reqprep: Strict routing enforced for session 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 siphub01*CLI> [Apr 21 14:40:36] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '10.51.1.10:5060' gives... siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '10.51.1.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:36] set_destination: set destination to 10.51.1.10:5060 siphub01*CLI> [Apr 21 14:40:36] Reliably Transmitting (no NAT) to 10.51.1.10:5060: NOTIFY sip:BoxWares.18@10.51.1.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.110:5060;branch=z9hG4bK5784d7d8;rport Max-Forwards: 70 From: "00888040220" ;tag=as673b2bc9 To: ;tag=1450788815 Contact: Call-ID: 6dd6b58138662f486c73a9d13202ddc5@172.20.20.110:5060 CSeq: 105 NOTIFY User-Agent: TelBox Event: refer;id=104 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 21 SIP/2.0 183 Ringing --- siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:3544 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #112 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1021]: chan_sip.c:3089 __sip_xmit: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.51.1.10:5060 siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: chan_sip.c:27692 sip_set_rtp_peer: Sending reinvite on SIP '14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060' - It's audio soon redirected to IP (null) siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: chan_sip.c:9580 reqprep: Strict routing enforced for session 14faa1e10a0cb708128732397faa3f50@172.20.54.10:5060 siphub01*CLI> [Apr 21 14:40:36] set_destination: Parsing for address/port to send to siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: netsock2.c:125 ast_sockaddr_split_hostport: Splitting '172.20.54.10:5060' gives... siphub01*CLI> [Apr 21 14:40:36] DEBUG[1087]: netsock2.c:155 ast_sockaddr_split_hostport: ...host '172.20.54.10' and port '5060'. siphub01*CLI> [Apr 21 14:40:36] set_destination: set destination to 172.20.54.10:5060 siphub01*CLI> [Apr 21 14:41:47] -- Remote UNIX connection siphub01*CLI> [Apr 21 14:41:47] -- Remote UNIX connection disconnected siphub01*CLI>