[Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 0 [ 41]: INVITE sip:71034577@sip.xxx.com SIP/2.0 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:2060;branch=z9hG4bK-qvtmdxzrbp1t;rport [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 2 [ 47]: From: ;tag=vhz53f0mx1 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 3 [ 43]: To: "Fliegl, Deti" <71034567@sip.xxx.com> [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 4 [ 55]: Call-ID: 3c26706b5f37-sglubclrsm2p@snom360-000413230FD7 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 5 [ 14]: CSeq: 2 INVITE [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 7 [ 69]: Contact: ;flow-id=1 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 8 [ 56]: Replaces: 770539e860a2cc1d594b578a03afc2f1@sip.xxx.com [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 9 [ 41]: P-Key-Flags: resolution="31x13", keys="4" [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 10 [ 25]: User-Agent: snom360/6.5.3 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 11 [ 23]: Accept: application/sdp [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 12 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 13 [ 31]: Allow-Events: talk, hold, refer [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 14 [ 44]: Supported: timer, 100rel, replaces, callerid [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 15 [ 35]: Session-Expires: 3600;refresher=uas [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 16 [ 10]: Min-SE: 90 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 17 [173]: Authorization: Digest username="71034554-B",realm="sip.xxx.com",nonce="5b083e51",uri="sip:71034577@sip.xxx.com",response="f96aafece45e17af16e1f9ba06dc2f1a",algorithm=md5 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 18 [ 29]: Content-Type: application/sdp [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 19 [ 19]: Content-Length: 477 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Header 20 [ 0]: [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 0 [ 3]: v=0 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 1 [ 48]: o=root 359605864 359605864 IN IP4 xxx.xxx.xxx.xxx [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 2 [ 6]: s=call [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 3 [ 23]: c=IN IP4 xxx.xxx.xxx.xxx [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 4 [ 5]: t=0 0 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 5 [ 40]: m=audio 51144 RTP/AVP 0 8 9 2 3 18 4 101 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FhE9df1GxRXJnzWaM2s0RBS7iXhr0nVj3y9CHc2O [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 8 [ 20]: a=rtpmap:8 pcma/8000 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 9 [ 20]: a=rtpmap:9 g722/8000 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 10 [ 23]: a=rtpmap:2 g726-32/8000 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 11 [ 19]: a=rtpmap:3 gsm/8000 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 12 [ 21]: a=rtpmap:18 g729/8000 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 13 [ 20]: a=rtpmap:4 g723/8000 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 15 [ 15]: a=fmtp:101 0-16 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 16 [ 10]: a=ptime:20 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 17 [ 21]: a=encryption:optional [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4654 parse_request: Body 18 [ 10]: a=sendrecv --- (20 headers 19 lines) --- [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:14692 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13303 handle_request_invite: INVITE part of call transfer. Replaces [770539e860a2cc1d594b578a03afc2f1@sip.xxx.com] [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13338 handle_request_invite: Invite/replaces: Will use Replace-Call-ID : 770539e860a2cc1d594b578a03afc2f1@sip.xxx.com Fromtag: Totag: Sending to xxx.xxx.xxx.xxx : 2060 (NAT) [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13403 handle_request_invite: Initializing initreq for method INVITE - callid 3c26706b5f37-sglubclrsm2p@snom360-000413230FD7 Using INVITE request as basis request - 3c26706b5f37-sglubclrsm2p@snom360-000413230FD7 Found user '71034554-B' for '71034554-B' [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:2678 do_setnat: Setting NAT on RTP to On [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:2688 do_setnat: Setting NAT on UDPTL to On Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:4943 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port xxx.xxx.xxx.xxx:51144 Got unsupported a:crypto in SDP offer Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g722 for ID 9 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g729 for ID 18 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:5173 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port xxx.xxx.xxx.xxx:51144 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:5250 process_sdp: We're settling with these formats: 0x10e (gsm|ulaw|alaw|g729) [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13480 handle_request_invite: Checking SIP call limits for device 71034554-B [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3101 update_call_counter: Updating call counter for incoming call [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3171 update_call_counter: Call from peer '71034554-B' is 1 out of 2 [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:387 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/71034554-B Looking for 71034577 in fromsip (domain sip.xxx.com) [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:245 ast_device_state: No provider found, checking channel drivers for SIP - 71034554 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:15318 sip_devicestate: Checking device state for peer 71034554 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3880 sip_new: *** Our native formats are 0x8 (alaw) [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3881 sip_new: *** Joint capabilities are 0x10e (gsm|ulaw|alaw|g729) [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3882 sip_new: *** Our capabilities are 0x50e (gsm|ulaw|alaw|g729|ilbc) [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3883 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3906 sip_new: This channel will not be able to handle video. [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:8000 build_route: build_route: Contact hop: ;flow-id=1 list_route: hop: [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13544 handle_request_invite: Sending this call to the invite/replcaes handler 3c26706b5f37-sglubclrsm2p@snom360-000413230FD7 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13090 handle_invite_replaces: Attended transfer attempted to replace call with no bridge (maybe ringing). Channel SIP/71034567-U-08229b98! [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13101 handle_invite_replaces: SIP transfer: Invite Replace incoming channel should replace and hang up channel SIP/71034567-U-08229b98 (one call leg) <--- Transmitting (NAT) to xxx.xxx.xxx.xxx:2060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:2060;branch=z9hG4bK-qvtmdxzrbp1t;received=xxx.xxx.xxx.xxx;rport=2060 From: ;tag=vhz53f0mx1 To: "Fliegl, Deti" <71034567@sip.xxx.com> Call-ID: 3c26706b5f37-sglubclrsm2p@snom360-000413230FD7 CSeq: 2 INVITE User-Agent: Sipfel Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:387 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/71034554-B-08269740 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6472 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6247 add_sdp: ** Our capability: 0x10e (gsm|ulaw|alaw|g729) Video flag: True [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6248 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at yyy.yyy.yyy.yyy port 50084 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6373 add_sdp: -- Done with adding codecs to SDP [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6417 add_sdp: Done building SDP. Settling with this capability: 0x10e (gsm|ulaw|alaw|g729) <--- Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:2060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:2060;branch=z9hG4bK-qvtmdxzrbp1t;received=xxx.xxx.xxx.xxx;rport=2060 From: ;tag=vhz53f0mx1 To: "Fliegl, Deti" <71034567@sip.xxx.com>;tag=as4a1a32d5 Call-ID: 3c26706b5f37-sglubclrsm2p@snom360-000413230FD7 CSeq: 2 INVITE User-Agent: Sipfel Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 338 v=0 o=root 7887 7887 IN IP4 yyy.yyy.yyy.yyy s=session c=IN IP4 yyy.yyy.yyy.yyy t=0 0 m=audio 50084 RTP/AVP 8 0 18 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:2004 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #361 [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:387 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/71034554-B-08269740 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13150 handle_invite_replaces: Invite/Replaces: preparing to masquerade SIP/71034554-B-08269740 into SIP/71034567-U-08229b98 [Jan 3 17:28:24] DEBUG[7890]: channel.c:3113 ast_channel_masquerade: Planning to masquerade channel SIP/71034554-B-08269740 into the structure of SIP/71034567-U-08229b98 [Jan 3 17:28:24] DEBUG[7890]: channel.c:3127 ast_channel_masquerade: Done planning to masquerade channel SIP/71034554-B-08269740 into the structure of SIP/71034567-U-08229b98 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13164 handle_invite_replaces: Invite/Replaces: Going to masquerade SIP/71034554-B-08269740 into SIP/71034567-U-08229b98 [Jan 3 17:28:24] DEBUG[7890]: channel.c:3244 ast_do_masquerade: Actually Masquerading SIP/71034554-B-08269740(6) into the structure of SIP/71034567-U-08229b98(5) [Jan 3 17:28:24] DEBUG[7890]: channel.c:3259 ast_do_masquerade: Got clone lock for masquerade on 'SIP/71034554-B-08269740' at 0x82227c8 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3666 sip_fixup: SIP Fixup: New owner for dialogue 770539e860a2cc1d594b578a03afc2f1@sip.xxx.com: SIP/71034554-B-08269740 (Old parent: SIP/71034554-B-08269740) [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3384 sip_hangup: SIP Transfer: Not hanging up right now... Rescheduling hangup for 770539e860a2cc1d594b578a03afc2f1@sip.xxx.com. Scheduling destruction of SIP dialog '770539e860a2cc1d594b578a03afc2f1@sip.xxx.com' in 34880 ms (Method: INVITE) [Jan 3 17:28:24] WARNING[7890]: channel.c:2695 set_format: Unable to find a codec translation path from alaw to g729 [Jan 3 17:28:24] WARNING[7890]: channel.c:2695 set_format: Unable to find a codec translation path from alaw to g729 [Jan 3 17:28:24] DEBUG[7890]: channel.c:3454 ast_do_masquerade: Putting channel SIP/71034554-B-08269740 in 256/256 formats [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3666 sip_fixup: SIP Fixup: New owner for dialogue 3c26706b5f37-sglubclrsm2p@snom360-000413230FD7: SIP/71034554-B-08269740 (Old parent: SIP/71034567-U-08229b98) [Jan 3 17:28:24] DEBUG[7890]: channel.c:3490 ast_do_masquerade: Released clone lock on 'SIP/71034567-U-08229b98' [Jan 3 17:28:24] DEBUG[7890]: channel.c:3500 ast_do_masquerade: Done Masquerading SIP/71034554-B-08269740 (6) [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13178 handle_invite_replaces: Invite/Replace: Could successfully read frame from RING channel! [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:387 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/71034567-U-08229b98 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13201 handle_invite_replaces: After transfer:---------------------------- [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13202 handle_invite_replaces: -- C: SIP/71034567-U-08229b98 State Down [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13204 handle_invite_replaces: -- replacecall: SIP/71034554-B-08269740 State Up [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13206 handle_invite_replaces: -- P->owner: SIP/71034554-B-08269740 State Up [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13211 handle_invite_replaces: -- No call bridged to C->owner [New Thread 1103027120 (LWP 7973)] [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:13214 handle_invite_replaces: End After transfer:---------------------------- [Jan 3 17:28:24] DEBUG[7890]: channel.c:1597 ast_hangup: Hanging up zombie 'SIP/71034567-U-08229b98' [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:387 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/71034567-U-08229b98 -- SIP/71034554-B-08269740 answered SIP/71034567-F-081ce8a0 [Jan 3 17:28:24] DEBUG[7890]: channel.c:2724 set_format: Set channel SIP/71034554-B-08269740 to write format alaw [Jan 3 17:28:24] DEBUG[7890]: channel.c:2724 set_format: Set channel SIP/71034554-B-08269740 to read format alaw [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:387 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/71034567-F-081ce8a0 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:3546 sip_answer: SIP answering channel: SIP/71034567-F-081ce8a0 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6472 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6247 add_sdp: ** Our capability: 0x40c (ulaw|alaw|ilbc) Video flag: True [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6248 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at yyy.yyy.yyy.yyy port 50030 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6373 add_sdp: -- Done with adding codecs to SDP [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:6417 add_sdp: Done building SDP. Settling with this capability: 0x40c (ulaw|alaw|ilbc) <--- Reliably Transmitting (NAT) to 172.16.2.180:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.2.180:5060;branch=z9hG4bK784DCA208184A555;received=172.16.2.180 From: ;tag=9229F351329C03F9 To: ;tag=as463a8807 Call-ID: 813DE7A8F7EA01AB@172.16.2.180 CSeq: 303 INVITE User-Agent: Sipfel Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 313 v=0 o=root 7887 7887 IN IP4 yyy.yyy.yyy.yyy s=session c=IN IP4 yyy.yyy.yyy.yyy t=0 0 m=audio 50030 RTP/AVP 8 0 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:2004 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #363 -- Packet2Packet bridging SIP/71034567-F-081ce8a0 and SIP/71034554-B-08269740 [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:371 do_state_change: Changing state for SIP/71034554 - state 4 (Invalid) <--- SIP read from 172.16.2.180:5060 ---> ACK sip:71034577@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP 172.16.2.180:5060;branch=z9hG4bK01A2C40E6D465BCE From: ;tag=9229F351329C03F9 To: ;tag=as463a8807 Call-ID: 813DE7A8F7EA01AB@172.16.2.180 CSeq: 303 ACK Contact: Max-Forwards: 70 User-Agent: AVM FRITZ!Box Fon WLAN 7050 (UI) 14.04.25 (Sep 6 2006) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:14692 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:2103 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #363 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:2114 __sip_ack: Stopping retransmission on '813DE7A8F7EA01AB@172.16.2.180' of Response 303: Match Found [Jan 3 17:28:24] NOTICE[7890]: rtp.c:3088 bridge_p2p_loop: p2p-rtp-bridge: Ooh, empty read... [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:245 ast_device_state: No provider found, checking channel drivers for SIP - 71034554-B [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:15318 sip_devicestate: Checking device state for peer 71034554-B [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:371 do_state_change: Changing state for SIP/71034554-B - state 2 (In use) [New Thread 1096371120 (LWP 7975)] [Jan 3 17:28:24] NOTICE[7890]: rtp.c:3088 bridge_p2p_loop: p2p-rtp-bridge: Ooh, empty read... [Jan 3 17:28:24] DEBUG[7890]: devicestate.c:371 do_state_change: Changing state for SIP/71034554-B - state 2 (In use) [Jan 3 17:28:24] DEBUG[7890]: app_queue.c:568 changethread: Device 'SIP/71034554-B' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jan 3 17:28:24] DEBUG[7890]: app_queue.c:568 changethread: Device 'SIP/71034554' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [New Thread 1096629168 (LWP 7976)] [Thread 1096371120 (LWP 7975) exited] <--- SIP read from xxx.xxx.xxx.xxx:2060 ---> ACK sip:71034577@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:2060;branch=z9hG4bK-we0hn1sbm7xj;rport From: ;tag=vhz53f0mx1 To: "Fliegl, Deti" <71034567@sip.xxx.com>;tag=as4a1a32d5 Call-ID: 3c26706b5f37-sglubclrsm2p@snom360-000413230FD7 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:14692 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:2103 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #361 [Jan 3 17:28:24] DEBUG[7890]: chan_sip.c:2114 __sip_ack: Stopping retransmission on '3c26706b5f37-sglubclrsm2p@snom360-000413230FD7' of Response 2: Match Found [Jan 3 17:28:24] NOTICE[7890]: rtp.c:3088 bridge_p2p_loop: p2p-rtp-bridge: Ooh, empty read... [Jan 3 17:28:24] DEBUG[7890]: rtp.c:912 ast_rtcp_read: Got RTCP report of 52 bytes [Jan 3 17:28:24] NOTICE[7890]: rtp.c:3088 bridge_p2p_loop: p2p-rtp-bridge: Ooh, empty read... [Jan 3 17:28:24] NOTICE[7890]: rtp.c:3088 bridge_p2p_loop: p2p-rtp-bridge: Ooh, empty read... [Jan 3 17:28:24] NOTICE[7890]: rtp.c:3088 bridge_p2p_loop: p2p-rtp-bridge: Ooh, empty read... [Thread 1096629168 (LWP 7976) exited] [Jan 3 17:28:24] NOTICE[7890]: rtp.c:3088 bridge_p2p_loop: p2p-rtp-bridge: Ooh, empty read...