<-- SIP read from 192.168.0.231:5060: INVITE sip:30@domain.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK630d15244a71258d From: "Martin" ;tag=2aaf013b2b6d7fed To: Contact: Supported: replaces, timer Call-ID: e18bfa7218927f87@192.168.0.231 CSeq: 28142 INVITE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 154 v=0 o=231 8000 8000 IN IP4 192.168.0.231 s=SIP Call c=IN IP4 192.168.0.231 t=0 0 m=audio 5004 RTP/AVP 8 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 0: INVITE sip:30@domain.de SIP/2.0 (47) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK630d15244a71258d (65) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 2: From: "Martin" ;tag=2aaf013b2b6d7fed (71) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 3: To: (38) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 4: Contact: (36) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 5: Supported: replaces, timer (26) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 6: Call-ID: e18bfa7218927f87@192.168.0.231 (38) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 7: CSeq: 28142 INVITE (18) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.0.13 (40) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 9: Max-Forwards: 70 (16) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK (77) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 11: Content-Type: application/sdp (29) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 12: Content-Length: 154 (19) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 13: (0) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: v=0 (3) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: o=231 8000 8000 IN IP4 192.168.0.231 (35) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: s=SIP Call (10) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: c=IN IP4 192.168.0.231 (21) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: t=0 0 (5) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: m=audio 5004 RTP/AVP 8 (22) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: a=sendrecv (10) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: a=ptime:20 (10) --- (13 headers 9 lines)--- Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3150 sip_alloc: Allocating new SIP dialog for e18bfa7218927f87@192.168.0.231 - INVITE (With RTP) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:11212 handle_request: **** Received INVITE (5) - Command in SIP INVITE Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1007 parse_sip_options: Begin: parsing SIP "Supported: replaces, timer" Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1019 parse_sip_options: Found SIP option: -replaces- Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1025 parse_sip_options: Matched SIP option: replaces Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1019 parse_sip_options: Found SIP option: -timer- Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1025 parse_sip_options: Matched SIP option: timer Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1036 parse_sip_options: * SIP extension value: 5 for call e18bfa7218927f87@192.168.0.231 Using INVITE request as basis request - e18bfa7218927f87@192.168.0.231 Sending to 192.168.0.231 : 5060 (non-NAT) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:7188 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 192.168.0.231:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK630d15244a71258d;received=192.168.0.231 From: "Martin" ;tag=2aaf013b2b6d7fed To: ;tag=as1e4cddb4 Call-ID: e18bfa7218927f87@192.168.0.231 CSeq: 28142 INVITE User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="domain.de", nonce="413d7f49" Content-Length: 0 --- Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1296 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #50 Scheduling destruction of call 'e18bfa7218927f87@192.168.0.231' in 15000 ms Found user '231' voipserver4*CLI> <-- SIP read from 192.168.0.231:5060: ACK sip:30@domain.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK630d15244a71258d From: "Martin" ;tag=2aaf013b2b6d7fed To: ;tag=as1e4cddb4 Contact: Call-ID: e18bfa7218927f87@192.168.0.231 CSeq: 28142 ACK User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 0: ACK sip:30@domain.de SIP/2.0 (44) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK630d15244a71258d (65) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 2: From: "Martin" ;tag=2aaf013b2b6d7fed (71) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 3: To: ;tag=as1e4cddb4 (53) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 4: Contact: (36) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 5: Call-ID: e18bfa7218927f87@192.168.0.231 (38) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 6: CSeq: 28142 ACK (15) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 7: User-Agent: Grandstream GXP2000 1.1.0.13 (40) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 8: Max-Forwards: 70 (16) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK (77) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 10: Content-Length: 0 (21) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:11212 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1393 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #50 Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1404 __sip_ack: Stopping retransmission on 'e18bfa7218927f87@192.168.0.231' of Response 28142: Match Found voipserver4*CLI> <-- SIP read from 192.168.0.231:5060: INVITE sip:30@domain.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK150dd63499f12e90 From: "Martin" ;tag=2aaf013b2b6d7fed To: Contact: Supported: replaces, timer Proxy-Authorization: Digest username="231", realm="domain.de", algorithm=MD5, uri="sip:30@domain.de", nonce="413d7f49", response="2bd9b711a4521e5c42ac3040eb390060" Call-ID: e18bfa7218927f87@192.168.0.231 CSeq: 28143 INVITE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 154 v=0 o=231 8000 8001 IN IP4 192.168.0.231 s=SIP Call c=IN IP4 192.168.0.231 t=0 0 m=audio 5004 RTP/AVP 8 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 0: INVITE sip:30@domain.de SIP/2.0 (47) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK150dd63499f12e90 (65) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 2: From: "Martin" ;tag=2aaf013b2b6d7fed (71) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 3: To: (38) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 4: Contact: (36) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 5: Supported: replaces, timer (26) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 6: Proxy-Authorization: Digest username="231", realm="domain.de", algorithm=MD5, uri="sip:30@domain.de", nonce="413d7f49", response="2bd9b711a4521e5c42ac3040eb390060" (195) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 7: Call-ID: e18bfa7218927f87@192.168.0.231 (38) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 8: CSeq: 28143 INVITE (18) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 9: User-Agent: Grandstream GXP2000 1.1.0.13 (40) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 10: Max-Forwards: 70 (16) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 11: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK (77) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 12: Content-Type: application/sdp (29) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 13: Content-Length: 154 (19) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 14: (0) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: v=0 (3) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: o=231 8000 8001 IN IP4 192.168.0.231 (35) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: s=SIP Call (10) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: c=IN IP4 192.168.0.231 (21) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: t=0 0 (5) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: m=audio 5004 RTP/AVP 8 (22) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: a=sendrecv (10) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3412 parse_request: Line: a=ptime:20 (10) --- (14 headers 9 lines)--- Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:11212 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - e18bfa7218927f87@192.168.0.231 Sending to 192.168.0.231 : 5060 (non-NAT) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:7188 check_user_full: Setting NAT on RTP to 0 Found user '231' Found RTP audio format 8 Peer audio RTP is at port 192.168.0.231:5004 Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3621 process_sdp: Peer audio RTP is at port 192.168.0.231:5004 Found description format PCMA Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:10572 handle_request_invite: Checking SIP call limits for device 231 Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:2212 update_call_counter: Updating call counter for incoming call Looking for 30 in internal-sip-peers (domain domain.de) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:6170 build_route: build_route: Contact hop: list_route: hop: Transmitting (no NAT) to 192.168.0.231:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK150dd63499f12e90;received=192.168.0.231 From: "Martin" ;tag=2aaf013b2b6d7fed To: Call-ID: e18bfa7218927f87@192.168.0.231 CSeq: 28143 INVITE User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 21 16:48:03 DEBUG[16795]: chan_sip.c:11789 sip_devicestate: Checking device state for peer 231 Jul 21 16:48:03 DEBUG[16795]: channel.c:797 channel_find_locked: Avoiding initial deadlock for 'SIP/231-1a8b' Jul 21 16:48:03 DEBUG[16795]: devicestate.c:189 do_state_change: Changing state for SIP/231 - state 2 (In use) Jul 21 16:48:03 DEBUG[16795]: chan_sip.c:11789 sip_devicestate: Checking device state for peer 231 Jul 21 16:48:03 DEBUG[16795]: chan_sip.c:11789 sip_devicestate: Checking device state for peer 231 Jul 21 16:48:03 DEBUG[16876]: pbx.c:1678 pbx_extension_helper: Launching 'Dial' -- Executing Dial("SIP/231-1a8b", "SIP/230|3000|") in new stack Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3150 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:1877 create_addr_from_peer: Setting NAT on RTP to 0 Jul 21 16:48:03 DEBUG[16876]: channel.c:2878 ast_channel_inherit_variables: Not copying variable STACK-internal-sip-peers-30-1. Jul 21 16:48:03 DEBUG[16876]: channel.c:2878 ast_channel_inherit_variables: Not copying variable SIPCALLID. Jul 21 16:48:03 DEBUG[16876]: channel.c:2878 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. Jul 21 16:48:03 DEBUG[16876]: channel.c:2878 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. Jul 21 16:48:03 DEBUG[16876]: channel.c:2878 ast_channel_inherit_variables: Not copying variable SIPURI. Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:2071 sip_call: Outgoing Call for 230 Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:2212 update_call_counter: Updating call counter for outgoing call We're at 192.168.0.244 port 6722 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 0: INVITE sip:230@192.168.0.230 SIP/2.0 (35) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK29e48e3b;rport (63) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 2: From: "231" ;tag=as17b0729e (48) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 3: To: (26) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 4: Contact: (30) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 5: Call-ID: 2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244 (54) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 7: User-Agent: asterisk (22) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 8: Max-Forwards: 70 (16) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 9: Remote-Party-ID: "231" ;privacy=off;screen=no (66) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 10: Date: Fri, 21 Jul 2006 14:48:03 GMT (35) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 12: Content-Type: application/sdp (29) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 13: Content-Length: 262 (19) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3380 parse_request: Header 14: (0) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: v=0 (3) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: o=root 16876 16876 IN IP4 192.168.0.244 (38) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: s=session (9) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: c=IN IP4 192.168.0.244 (21) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: t=0 0 (5) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: m=audio 6722 RTP/AVP 8 0 3 101 (30) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: a=fmtp:101 0-16 (15) Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:3412 parse_request: Line: a=silenceSupp:off - - - - (25) 14 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.0.230:5060: INVITE sip:230@192.168.0.230 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK29e48e3b;rport From: "231" ;tag=as17b0729e To: Contact: Call-ID: 2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244 CSeq: 102 INVITE User-Agent: asterisk Max-Forwards: 70 Remote-Party-ID: "231" ;privacy=off;screen=no Date: Fri, 21 Jul 2006 14:48:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 262 v=0 o=root 16876 16876 IN IP4 192.168.0.244 s=session c=IN IP4 192.168.0.244 t=0 0 m=audio 6722 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 21 16:48:03 DEBUG[16876]: chan_sip.c:1296 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #52 -- Called 230 Jul 21 16:48:03 DEBUG[16876]: channel.c:2379 set_format: Set channel SIP/230-9fd6 to read format alaw Jul 21 16:48:03 DEBUG[16876]: channel.c:2379 set_format: Set channel SIP/231-1a8b to write format alaw Jul 21 16:48:03 DEBUG[16876]: channel.c:2379 set_format: Set channel SIP/231-1a8b to read format alaw Jul 21 16:48:03 DEBUG[16876]: channel.c:2379 set_format: Set channel SIP/230-9fd6 to write format alaw Jul 21 16:48:03 DEBUG[16877]: app_queue.c:523 changethread: Device 'SIP/231' changed to state '2' (In use) but we don't care because they're not a member of any queue. voipserver4*CLI> <-- SIP read from 192.168.0.230:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK29e48e3b;rport From: "231" ;tag=as17b0729e To: Call-ID: 2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.0.1.9 Content-Length: 0 Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 0: SIP/2.0 100 Trying (18) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK29e48e3b;rport (63) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 2: From: "231" ;tag=as17b0729e (48) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 3: To: (26) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 4: Call-ID: 2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244 (54) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.0.1.9 (39) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 7: Content-Length: 0 (17) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1448 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #52 - INVITE (got response) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1457 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244' Request 102: Found Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:9654 handle_response_invite: SIP response 100 to standard invite voipserver4*CLI> <-- SIP read from 192.168.0.230:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK29e48e3b;rport From: "231" ;tag=as17b0729e To: ;tag=43ccd12220a6d9be Call-ID: 2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.0.1.9 Content-Length: 0 Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 0: SIP/2.0 180 Ringing (19) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK29e48e3b;rport (63) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 2: From: "231" ;tag=as17b0729e (48) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 3: To: ;tag=43ccd12220a6d9be (47) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 4: Call-ID: 2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244 (54) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.0.1.9 (39) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 7: Content-Length: 0 (17) Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:1457 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244' Request 102: Found Jul 21 16:48:03 DEBUG[16802]: chan_sip.c:9654 handle_response_invite: SIP response 180 to standard invite -- SIP/230-9fd6 is ringing Transmitting (no NAT) to 192.168.0.231:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK150dd63499f12e90;received=192.168.0.231 From: "Martin" ;tag=2aaf013b2b6d7fed To: ;tag=as44c3f2b9 Call-ID: e18bfa7218927f87@192.168.0.231 CSeq: 28143 INVITE User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 21 16:48:03 DEBUG[16795]: chan_sip.c:11789 sip_devicestate: Checking device state for peer 230 Jul 21 16:48:03 DEBUG[16795]: devicestate.c:189 do_state_change: Changing state for SIP/230 - state 6 (Ringing) Jul 21 16:48:03 DEBUG[16795]: chan_sip.c:11789 sip_devicestate: Checking device state for peer 230 Jul 21 16:48:03 DEBUG[16795]: chan_sip.c:11789 sip_devicestate: Checking device state for peer 230 Jul 21 16:48:03 DEBUG[16795]: chan_sip.c:11789 sip_devicestate: Checking device state for peer 230 Jul 21 16:48:03 DEBUG[16795]: chan_sip.c:11789 sip_devicestate: Checking device state for peer 230 Jul 21 16:48:03 DEBUG[16878]: app_queue.c:523 changethread: Device 'SIP/230' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Jul 21 16:48:05 DEBUG[16802]: chan_sip.c:1187 retrans_pkt: SIP TIMER: Rescheduling retransmission #48 (5) NOTIFY - 4 Jul 21 16:48:05 DEBUG[16802]: chan_sip.c:1201 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #48)) Retransmitting #5 (no NAT) to 192.168.0.232:5060: NOTIFY sip:232@192.168.0.232:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK65f8a567;rport From: "asterisk" ;tag=as1d73cf8c To: Contact: Call-ID: 1edf5ca25d403c4f77bdf4256ce7a8a1@192.168.0.244 CSeq: 102 NOTIFY User-Agent: asterisk Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 88 Messages-Waiting: no Message-Account: sip:9999@192.168.0.244 Voice-Message: 0/0 (0/0) --- Jul 21 16:48:08 DEBUG[16802]: chan_sip.c:1326 __sip_autodestruct: Auto destroying call '1edf5ca25d403c4f77bdf4256ce7a8a1@192.168.0.244' Destroying call '1edf5ca25d403c4f77bdf4256ce7a8a1@192.168.0.244' voipserver4*CLI> <-- SIP read from 192.168.0.230:5060: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK29e48e3b;rport From: "231" ;tag=as17b0729e To: ;tag=fb8749425431eeee Call-ID: 2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.0.1.9 Content-Length: 0 Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 0: SIP/2.0 487 Request Cancelled (29) Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK29e48e3b;rport (63) Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 2: From: "231" ;tag=as17b0729e (48) Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 3: To: ;tag=fb8749425431eeee (47) Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 4: Call-ID: 2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244 (54) Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.0.1.9 (39) Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 7: Content-Length: 0 (17) Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:3380 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:1382 __sip_ack: Acked pending invite 102 Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:1404 __sip_ack: Stopping retransmission on '2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244' of Request 102: Match Found Jul 21 16:49:03 DEBUG[16802]: chan_sip.c:2212 update_call_counter: Updating call counter for outgoing call Transmitting (no NAT) to 192.168.0.230:5060: ACK sip:230@192.168.0.230 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.244:5060;branch=z9hG4bK29e48e3b;rport From: "231" ;tag=as17b0729e To: ;tag=fb8749425431eeee Contact: Call-ID: 2b14461a30e501d1332f6f357d2eb3c3@192.168.0.244 CSeq: 102 ACK User-Agent: asterisk Max-Forwards: 70 Remote-Party-ID: "231" ;privacy=off;screen=no Content-Length: 0 --- voipserver4*CLI> exit At this point nothing happends until I Hangup on the Calling phone