<--- Received SIP request (1269 bytes) from UDP:2002:x:x::123:5060 ---> INVITE sip:4989xxxxxxx5@[2a00:x:x:x::1]:5060;line=utnvpdj SIP/2.0 Max-Forwards: 70 Record-Route: Record-Route: Via: SIP/2.0/UDP [2002:x:x:0:0:0:0:123];branch=z9hG4bK27cd.c22b55190ea4b8b66d5e1f4e17ce2db4.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKm6uC~arz;rport=5080 From: ;tag=6E703E9B-589A5AFC000EFC70-F9FE6700 To: CSeq: 10 INVITE Call-ID: 0a6531ae0f004dd4-Acc1800-B2b12@10.233.181.36_b2b-1 Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS Supported: resource-priority Content-Type: application/sdp Content-Length: 204 Contact: v=0 o=- 3013973546 1 IN IP6 2002:x:x::123 s=- c=IN IP6 2002:x:x::123 t=0 0 m=audio 30780 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=sendrecv a=rtcp:30781 <--- Transmitting SIP response (766 bytes) to UDP:2002:x:x::123:5060 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP [2002:x:x:0:0:0:0:123];rport=5060;received=2002:x:x::123;branch=z9hG4bK27cd.c22b55190ea4b8b66d5e1f4e17ce2db4.0 v: SIP/2.0/UDP 127.0.0.1:5080;rport=5080;received=127.0.0.1;branch=z9hG4bKm6uC~arz Record-Route: Record-Route: i: 0a6531ae0f004dd4-Acc1800-B2b12@10.233.181.36_b2b-1 f: ;tag=6E703E9B-589A5AFC000EFC70-F9FE6700 t: CSeq: 10 INVITE Server: Asterisk l: 0 <--- Transmitting SIP request (955 bytes) to UDP:88.x.x.38:50060 ---> INVITE sip:553@88.x.x.38:50060;uniq=C465CA211E8EEE061CD248D47C840 SIP/2.0 v: SIP/2.0/UDP 78.x.x.40:5060;rport;branch=z9hG4bKPjf632173f-587b-4ae1-b2c4-7ad926ef7b56 f: "4989xxxxxxx7" ;tag=0bd02b3a-78b3-42b8-a93a-190af651dc9c t: m: i: 4b4bbaee-12df-487f-93cb-83207ede60e4 CSeq: 30506 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER k: 100rel, timer, replaces, norefersub x: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk c: application/sdp l: 259 v=0 o=- 778216150 778216150 IN IP4 192.168.1.100 s=Asterisk c=IN IP4 78.x.x.40 t=0 0 m=audio 18672 RTP/AVP 9 8 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP response (430 bytes) from UDP:88.x.x.38:50060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100:5060;rport=5060;branch=z9hG4bKPjf632173f-587b-4ae1-b2c4-7ad926ef7b56 From: "4989xxxxxxx7" ;tag=0bd02b3a-78b3-42b8-a93a-190af651dc9c To: Call-ID: 4b4bbaee-12df-487f-93cb-83207ede60e4 CSeq: 30506 INVITE User-Agent: AVM FRITZ!Box 7490 113.06.80 (Jan 19 2017) Content-Length: 0 <--- Received SIP response (530 bytes) from UDP:88.x.x.38:50060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.100:5060;rport=5060;branch=z9hG4bKPjf632173f-587b-4ae1-b2c4-7ad926ef7b56 From: "4989xxxxxxx7" ;tag=0bd02b3a-78b3-42b8-a93a-190af651dc9c To: ;tag=27A17AE953704D05 Call-ID: 4b4bbaee-12df-487f-93cb-83207ede60e4 CSeq: 30506 INVITE Contact: User-Agent: AVM FRITZ!Box 7490 113.06.80 (Jan 19 2017) Content-Length: 0 <--- Transmitting SIP response (957 bytes) to UDP:2002:x:x::123:5060 ---> SIP/2.0 180 Ringing v: SIP/2.0/UDP [2002:x:x:0:0:0:0:123];rport=5060;received=2002:x:x::123;branch=z9hG4bK27cd.c22b55190ea4b8b66d5e1f4e17ce2db4.0 v: SIP/2.0/UDP 127.0.0.1:5080;rport=5080;received=127.0.0.1;branch=z9hG4bKm6uC~arz Record-Route: Record-Route: i: 0a6531ae0f004dd4-Acc1800-B2b12@10.233.181.36_b2b-1 f: ;tag=6E703E9B-589A5AFC000EFC70-F9FE6700 t: ;tag=4810e6e4-e2b2-454c-9e57-e40fee58f284 CSeq: 10 INVITE Server: Asterisk m: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER l: 0 <--- Received SIP response (1045 bytes) from UDP:88.x.x.38:50060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5060;rport=5060;branch=z9hG4bKPjf632173f-587b-4ae1-b2c4-7ad926ef7b56 From: "4989xxxxxxx7" ;tag=0bd02b3a-78b3-42b8-a93a-190af651dc9c To: ;tag=27A17AE953704D05 Call-ID: 4b4bbaee-12df-487f-93cb-83207ede60e4 CSeq: 30506 INVITE Contact: User-Agent: AVM FRITZ!Box 7490 113.06.80 (Jan 19 2017) Supported: 100rel,replaces Allow-Events: telephone-event,refer Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH Content-Type: application/sdp Accept: application/sdp, multipart/mixed Accept-Encoding: identity Content-Length: 256 v=0 o=user 10392804 10392804 IN IP4 88.x.x.38 s=Asterisk c=IN IP4 88.x.x.38 t=0 0 m=audio 7078 RTP/AVP 9 8 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:7079 a=ptime:30 <--- Transmitting SIP request (479 bytes) to UDP:88.x.x.38:50060 ---> ACK sip:553@88.x.x.38:50060;uniq=C465CA211E8EEE061CD248D47C840 SIP/2.0 v: SIP/2.0/UDP 78.x.x.40:5060;rport;branch=z9hG4bKPj14ed6bf1-6f12-4e92-aa66-4833d37bd93f f: "4989xxxxxxx7" ;tag=0bd02b3a-78b3-42b8-a93a-190af651dc9c t: ;tag=27A17AE953704D05 i: 4b4bbaee-12df-487f-93cb-83207ede60e4 CSeq: 30506 ACK Max-Forwards: 70 User-Agent: Asterisk l: 0 <--- Transmitting SIP response (1259 bytes) to UDP:2002:x:x::123:5060 ---> SIP/2.0 200 OK v: SIP/2.0/UDP [2002:x:x:0:0:0:0:123];rport=5060;received=2002:x:x::123;branch=z9hG4bK27cd.c22b55190ea4b8b66d5e1f4e17ce2db4.0 v: SIP/2.0/UDP 127.0.0.1:5080;rport=5080;received=127.0.0.1;branch=z9hG4bKm6uC~arz Record-Route: Record-Route: i: 0a6531ae0f004dd4-Acc1800-B2b12@10.233.181.36_b2b-1 f: ;tag=6E703E9B-589A5AFC000EFC70-F9FE6700 t: ;tag=4810e6e4-e2b2-454c-9e57-e40fee58f284 CSeq: 10 INVITE Server: Asterisk Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER m: k: 100rel, timer, replaces, norefersub c: application/sdp l: 244 v=0 o=- 3013973546 3 IN IP6 2a00:x:x:x::1 s=Asterisk c=IN IP6 2a00:x:x:x::1 t=0 0 m=audio 12946 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=ptime:20 a=maxptime:150 a=sendrecv [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge_native_rtp.c:332 native_rtp_bridge_compatible: Bridge '34b1a247-8e1e-4c1b-a7ec-9ea15089523a' can not use native RTP bridge as two channels are required [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:506 find_best_technology: Bridge technology native_rtp is not compatible with properties of existing bridge. [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:515 find_best_technology: Chose bridge technology simple_bridge [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:794 bridge_base_init: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: calling simple_bridge technology constructor [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:802 bridge_base_init: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: calling simple_bridge technology start [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge_channel.c:2683 bridge_channel_internal_join: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: 0x7fcffc592a18(PJSIP/553-00000032) is joining [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge_channel.c:2118 bridge_channel_internal_push_full: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: pushing 0x7fcffc592a18(PJSIP/553-00000032) -- Channel PJSIP/553-00000032 joined 'simple_bridge' basic-bridge <34b1a247-8e1e-4c1b-a7ec-9ea15089523a> [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge_native_rtp.c:332 native_rtp_bridge_compatible: Bridge '34b1a247-8e1e-4c1b-a7ec-9ea15089523a' can not use native RTP bridge as two channels are required [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:506 find_best_technology: Bridge technology native_rtp is not compatible with properties of existing bridge. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:515 find_best_technology: Chose bridge technology simple_bridge [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:1048 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a is already using the new technology. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:432 bridge_channel_complete_join: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: 0x7fcffc592a18(PJSIP/553-00000032) is joining simple_bridge technology [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge_channel.c:2683 bridge_channel_internal_join: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: 0x7fcffc269bc8(PJSIP/sip.myprov.com-0000002f) is joining [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge_channel.c:2118 bridge_channel_internal_push_full: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: pushing 0x7fcffc269bc8(PJSIP/sip.myprov.com-0000002f) -- Channel PJSIP/sip.myprov.com-0000002f joined 'simple_bridge' basic-bridge <34b1a247-8e1e-4c1b-a7ec-9ea15089523a> [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:515 find_best_technology: Chose bridge technology native_rtp > Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: switching from simple_bridge technology to native_rtp [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:1097 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: calling native_rtp technology constructor [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:1115 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: moving 0x7fcffc592a18(PJSIP/553-00000032) to dummy bridge temporarily [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:1132 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: 0x7fcffc592a18(PJSIP/553-00000032) is leaving simple_bridge technology (dummy) [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:1149 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: calling simple_bridge technology stop [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:432 bridge_channel_complete_join: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: 0x7fcffc269bc8(PJSIP/sip.myprov.com-0000002f) is joining native_rtp technology > Remotely bridged 'PJSIP/sip.myprov.com-0000002f' and 'PJSIP/553-00000032' - media will flow directly between them [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:432 bridge_channel_complete_join: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: 0x7fcffc592a18(PJSIP/553-00000032) is joining native_rtp technology > Remotely bridged 'PJSIP/sip.myprov.com-0000002f' and 'PJSIP/553-00000032' - media will flow directly between them [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:1160 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: calling native_rtp technology start [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:1177 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a: calling simple_bridge technology destructor [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:515 find_best_technology: Chose bridge technology native_rtp [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: bridge.c:1048 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a is already using the new technology. [Feb 8 00:40:48] DEBUG[24896]: cdr.c:1296 cdr_object_finalize: Finalized CDR for PJSIP/553-00000032 - start 1486510845.210109 answer 1486510848.953659 end 1486510848.972132 dispo ANSWERED [Feb 8 00:40:48] DEBUG[1822]: chan_pjsip.c:371 send_direct_media_request: RTP changed on PJSIP/sip.myprov.com-0000002f; initiating direct media update [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:911 ast_sip_session_refresh: Delay sending reinvite to sip.myprov.com because of outstanding transaction... [Feb 8 00:40:48] DEBUG[1822]: chan_pjsip.c:371 send_direct_media_request: RTP changed on PJSIP/553-00000032; initiating direct media update [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_t38.c:734 create_outgoing_sdp_stream: Not creating outgoing SDP stream: T.38 not enabled [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:971 ast_sip_session_refresh: Sending session refresh SDP via re-INVITE to 553 [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:2479 handle_outgoing_request: Method is INVITE [Feb 8 00:40:48] DEBUG[1822]: res_pjsip/pjsip_resolver.c:477 sip_resolve: Performing SIP DNS resolution of target '88.x.x.38' [Feb 8 00:40:48] DEBUG[1822]: res_pjsip/pjsip_resolver.c:504 sip_resolve: Transport type for target '88.x.x.38' is 'UDP' [Feb 8 00:40:48] DEBUG[1822]: res_pjsip/pjsip_resolver.c:525 sip_resolve: Target '88.x.x.38' is an IP address, skipping resolution [Feb 8 00:40:48] DEBUG[1822]: res_pjsip/pjsip_message_ip_updater.c:215 multihomed_on_tx_message: Re-wrote Contact URI host/port to 192.168.1.100:5060 [Feb 8 00:40:48] DEBUG[1822]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '88.x.x.38' into... [Feb 8 00:40:48] DEBUG[1822]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '88.x.x.38' and port ''. [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_nat.c:280 nat_on_tx_message: Re-wrote Contact URI port to 5060 [Feb 8 00:40:48] DEBUG[1822]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '2002:x:x::123' into... [Feb 8 00:40:48] DEBUG[1822]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '2002:x:x::123' and port ''. [Feb 8 00:40:48] DEBUG[1822]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.100:5060' into... [Feb 8 00:40:48] DEBUG[1822]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.100' and port '5060'. [Feb 8 00:40:48] DEBUG[1822]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '88.x.x.38:50060' into... [Feb 8 00:40:48] DEBUG[1822]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '88.x.x.38' and port '50060'. <--- Transmitting SIP request (959 bytes) to UDP:88.x.x.38:50060 ---> INVITE sip:553@88.x.x.38:50060;uniq=C465CA211E8EEE061CD248D47C840 SIP/2.0 v: SIP/2.0/UDP 78.x.x.40:5060;rport;branch=z9hG4bKPj2a03aad7-ea0a-4e6d-bc58-baa39d5807b2 f: "4989xxxxxxx7" ;tag=0bd02b3a-78b3-42b8-a93a-190af651dc9c t: ;tag=27A17AE953704D05 m: i: 4b4bbaee-12df-487f-93cb-83207ede60e4 CSeq: 30507 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER k: 100rel, timer, replaces, norefersub x: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk c: application/sdp l: 242 v=0 o=- 778216150 778216151 IN IP4 192.168.1.100 s=Asterisk c=IN IP4 2002:x:x::123 t=0 0 m=audio 30780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:2384 __print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:2398 __print_debug_details: The state change pertains to the endpoint '553(PJSIP/553-00000032)' [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:2403 __print_debug_details: The inv session still has an invite_tsx (0x7fcffc2bf168) [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:2409 __print_debug_details: The UAC INVITE transaction involved in this state change is 0x7fcffc2bf168 [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:2413 __print_debug_details: The current transaction state is Calling [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:2415 __print_debug_details: The transaction state change event is TX_MSG [Feb 8 00:40:48] DEBUG[1822]: res_pjsip_session.c:2420 __print_debug_details: The current inv state is CONFIRMED [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:515 find_best_technology: Chose bridge technology native_rtp [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:1048 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a is already using the new technology. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: res_rtp_asterisk.c:4518 ast_rtp_read: 0x7fcfec01e310 -- Probation learning mode pass with source address 88.x.x.38:7078 > 0x7fcfec01e310 -- Probation passed - setting RTP source address to 88.x.x.38:7078 Got RTP packet from 88.x.x.38:7078 (type 09, seq 000001, ts 000240, len 000240) [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: chan_pjsip.c:739 chan_pjsip_read: Oooh, got a frame with format of g722 on channel 'PJSIP/553-00000032' when we're sending 'alaw', switching to match [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: channel.c:5657 set_format: Channel PJSIP/553-00000032 setting write format path: alaw -> alaw [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: channel.c:5457 ast_set_read_format_path: Channel PJSIP/553-00000032 setting read format path: g722 -> g722 [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: channel.c:5277 ast_write: Channel PJSIP/sip.myprov.com-0000002f changing write format from alaw to g722, native formats (alaw) [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: channel.c:5657 set_format: Channel PJSIP/sip.myprov.com-0000002f setting write format path: g722 -> alaw [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: translate.c:600 ast_translate: Sample size different 480 vs 240 [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: res_rtp_asterisk.c:3599 ast_rtp_write: Ooh, format changed from none to alaw Sent RTP packet to [2002:x:x::123]:30780 (type 08, seq 029448, ts 000136, len 000240) [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:515 find_best_technology: Chose bridge technology native_rtp [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:1048 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a is already using the new technology. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: chan_pjsip.c:739 chan_pjsip_read: Oooh, got a frame with format of g722 on channel 'PJSIP/553-00000032' when we're sending 'alaw', switching to match [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: channel.c:5657 set_format: Channel PJSIP/553-00000032 setting write format path: alaw -> alaw [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:515 find_best_technology: Chose bridge technology native_rtp [Feb 8 00:40:48] DEBUG[2474][C-0000000c]: bridge.c:1048 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a is already using the new technology. [Feb 8 00:40:48] DEBUG[2472][C-0000000c]: translate.c:600 ast_translate: Sample size different 480 vs 240 Sent RTP packet to [2002:x:x::123]:30780 (type 08, seq 029449, ts 000376, len 000240) Got RTP packet from 88.x.x.38:7078 (type 09, seq 000003, ts 000720, len 000240) [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: chan_pjsip.c:739 chan_pjsip_read: Oooh, got a frame with format of g722 on channel 'PJSIP/553-00000032' when we're sending 'alaw', switching to match [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: channel.c:5657 set_format: Channel PJSIP/553-00000032 setting write format path: alaw -> alaw [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: bridge.c:515 find_best_technology: Chose bridge technology native_rtp [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: bridge.c:1048 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a is already using the new technology. [Feb 8 00:40:49] DEBUG[2472][C-0000000c]: translate.c:600 ast_translate: Sample size different 480 vs 240 Sent RTP packet to [2002:x:x::123]:30780 (type 08, seq 029450, ts 000616, len 000240) Got RTP packet from 88.x.x.38:7078 (type 09, seq 000004, ts 000960, len 000240) [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: chan_pjsip.c:739 chan_pjsip_read: Oooh, got a frame with format of g722 on channel 'PJSIP/553-00000032' when we're sending 'alaw', switching to match [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: channel.c:5657 set_format: Channel PJSIP/553-00000032 setting write format path: alaw -> alaw [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: bridge.c:515 find_best_technology: Chose bridge technology native_rtp [Feb 8 00:40:49] DEBUG[2474][C-0000000c]: bridge.c:1048 smart_bridge_operation: Bridge 34b1a247-8e1e-4c1b-a7ec-9ea15089523a is already using the new technology. [Feb 8 00:40:49] DEBUG[2472][C-0000000c]: translate.c:600 ast_translate: Sample size different 480 vs 240 Sent RTP packet to [2002:x:x::123]:30780 (type 08, seq 029451, ts 000856, len 000240) <--- Received SIP request (906 bytes) from UDP:2002:x:x::123:5060 ---> ACK sip:[2a00:x:x:x::1]:5060 SIP/2.0 Max-Forwards: 70 Record-Route: Record-Route: Via: SIP/2.0/UDP [2002:x:x:0:0:0:0:123];branch=z9hG4bK27cd.bdcedfdd0fb7273abce2c33427ff2182.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bK6hF7waSN;rport=5080 From: ;tag=6E703E9B-589A5AFC000EFC70-F9FE6700 To: ;tag=4810e6e4-e2b2-454c-9e57-e40fee58f284 CSeq: 10 ACK Call-ID: 0a6531ae0f004dd4-Acc1800-B2b12@10.233.181.36_b2b-1 Content-Length: 0 Contact: [Feb 8 00:40:49] DEBUG[24905]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '[2002:x:x::123]:5060' into... [Feb 8 00:40:49] DEBUG[24905]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '2002:x:x::123' and port '5060'. [Feb 8 00:40:49] DEBUG[24905]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '[2a00:x:x:x::1]:5060' into... [Feb 8 00:40:49] DEBUG[24905]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '2a00:x:x:x::1' and port '5060'. [Feb 8 00:40:49] DEBUG[24905]: res_pjsip/pjsip_distributor.c:390 distributor: Searching for serializer on dialog dlg0x7fd000048498 for Request msg ACK/cseq=10 (rdata0x104a108) [Feb 8 00:40:49] DEBUG[24905]: res_pjsip/pjsip_distributor.c:396 distributor: Found serializer pjsip/distributor-00000057 on dialog dlg0x7fd000048498 [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2384 __print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2398 __print_debug_details: The state change pertains to the endpoint 'sip.myprov.com(PJSIP/sip.myprov.com-0000002f)' [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2406 __print_debug_details: The inv session does NOT have an invite_tsx [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2409 __print_debug_details: The UAS INVITE transaction involved in this state change is 0x7fcfec08a578 [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2413 __print_debug_details: The current transaction state is Terminated [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2415 __print_debug_details: The transaction state change event is USER [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2420 __print_debug_details: The current inv state is CONNECTING [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2885 session_inv_on_tsx_state_changed: Endpoint 'sip.myprov.com(PJSIP/sip.myprov.com-0000002f)' INVITE delay check. tsx-state:Terminated [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2384 __print_debug_details: Function session_inv_on_state_changed called on event RX_MSG [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2398 __print_debug_details: The state change pertains to the endpoint 'sip.myprov.com(PJSIP/sip.myprov.com-0000002f)' [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2406 __print_debug_details: The inv session does NOT have an invite_tsx [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2418 __print_debug_details: There is no transaction involved in this state change [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2420 __print_debug_details: The current inv state is CONFIRMED [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2463 handle_incoming: Received request [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2430 handle_incoming_request: Method is ACK [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:601 send_delayed_request: Endpoint 'sip.myprov.com(PJSIP/sip.myprov.com-0000002f)' sending delayed INVITE request. [Feb 8 00:40:49] DEBUG[733]: res_pjsip_t38.c:738 create_outgoing_sdp_stream: Not creating outgoing SDP stream: T.38 not enabled [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:971 ast_sip_session_refresh: Sending session refresh SDP via re-INVITE to sip.myprov.com [Feb 8 00:40:49] DEBUG[733]: res_pjsip_session.c:2479 handle_outgoing_request: Method is INVITE [Feb 8 00:40:49] DEBUG[733]: res_pjsip/pjsip_resolver.c:477 sip_resolve: Performing SIP DNS resolution of target '2002:x:x:0:0:0:0:123' [Feb 8 00:40:49] DEBUG[733]: res_pjsip/pjsip_resolver.c:504 sip_resolve: Transport type for target '2002:x:x:0:0:0:0:123' is 'UDP' [Feb 8 00:40:49] DEBUG[733]: res_pjsip/pjsip_resolver.c:525 sip_resolve: Target '2002:x:x:0:0:0:0:123' is an IP address, skipping resolution [Feb 8 00:40:49] DEBUG[733]: res_pjsip/pjsip_message_ip_updater.c:215 multihomed_on_tx_message: Re-wrote Contact URI host/port to 2a00:x:x:x::1:5060 [Feb 8 00:40:49] DEBUG[733]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '[2a00:x:x:x::1]:5060' into... [Feb 8 00:40:49] DEBUG[733]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '2a00:x:x:x::1' and port '5060'. [Feb 8 00:40:49] DEBUG[733]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '[2002:x:x::123]:5060' into... [Feb 8 00:40:49] DEBUG[733]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '2002:x:x::123' and port '5060'. <--- Transmitting SIP request (1248 bytes) to UDP:2002:x:x::123:5060 ---> INVITE sip:ngcp-lb@[2002:x:x:0:0:0:0:123]:5060;ngcpct=7369703a3132372e302e302e313a35303830 SIP/2.0 v: SIP/2.0/UDP [2a00:x:x:x::1]:5060;rport;branch=z9hG4bKPj24810b48-42ad-4f99-9d48-98be31491cd6 f: ;tag=4810e6e4-e2b2-454c-9e57-e40fee58f284 t: ;tag=6E703E9B-589A5AFC000EFC70-F9FE6700 m: i: 0a6531ae0f004dd4-Acc1800-B2b12@10.233.181.36_b2b-1 CSeq: 5257 INVITE Route: Route: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER k: 100rel, timer, replaces, norefersub x: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk c: application/sdp l: 227 v=0 o=- 3013973546 4 IN IP4 192.168.1.100 s=Asterisk c=IN IP4 88.x.x.38 t=0 0 m=audio 7078 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=ptime:20 a=maxptime:150 a=sendrecv