[Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: <--- SIP read from UDP://10.0.0.163:5061 ---> INVITE sip:+222222222@10.0.0.33:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.163:5061;branch=+222222222-+111111111-73176 From: Anonymous ;tag=8794645 To: +222222222 Call-ID: 1-22953@10.0.0.163 CSeq: 1 INVITE Max-Forwards: 70 Supported: timer Session-Expires: 1800 Min-SE: 1800 Contact: Anonymous Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE P-Asserted-Identity: Anonymous Privacy: id Content-Type: application/sdp Content-Length: 493 v=0 o=- 456789456 0 IN IP4 10.0.0.163 s=Cisco SDP 0 c=IN IP4 10.0.0.163 t=0 0 m=audio 6000 RTP/AVP 8 18 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=sqn:0 a=cdsc: 1 audio RTP/AVP 8 18 101 100 a=cdsc: 5 image udptl t38 a=cpar: a=T38FaxVersion:0 a=cpar: a=T38FaxRateManagement:transferredTCF a=cpar: a=T38FaxMaxDatagram:160 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=X-sqn:0 a=X-cap: 1 image udptl t38 <-------------> [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 0 [ 55]: INVITE sip:+222222222@10.0.0.33:5060;user=phone SIP/2.0 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 10.0.0.163:5061;branch=+222222222-+111111111-73176 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 2 [ 71]: From: Anonymous ;tag=8794645 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 3 [ 57]: To: +222222222 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 4 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 7 [ 16]: Supported: timer [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 8 [ 21]: Session-Expires: 1800 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 9 [ 12]: Min-SE: 1800 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 10 [ 51]: Contact: Anonymous [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 11 [ 69]: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 12 [ 69]: P-Asserted-Identity: Anonymous [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 13 [ 11]: Privacy: id [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 15 [ 19]: Content-Length: 493 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 16 [ 0]: [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 1 [ 33]: o=- 456789456 0 IN IP4 10.0.0.163 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 2 [ 13]: s=Cisco SDP 0 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.163 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 5 [ 33]: m=audio 6000 RTP/AVP 8 18 101 100 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 7 [ 15]: a=fmtp:101 0-15 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 8 [ 23]: a=rtpmap:100 X-NSE/8000 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 9 [ 26]: a=fmtp:100 192-194,200-202 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 10 [ 7]: a=sqn:0 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 11 [ 36]: a=cdsc: 1 audio RTP/AVP 8 18 101 100 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 12 [ 25]: a=cdsc: 5 image udptl t38 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 13 [ 25]: a=cpar: a=T38FaxVersion:0 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 14 [ 45]: a=cpar: a=T38FaxRateManagement:transferredTCF [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 15 [ 31]: a=cpar: a=T38FaxMaxDatagram:160 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 16 [ 38]: a=cpar: a=T38FaxUdpEC:t38UDPRedundancy [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 17 [ 9]: a=X-sqn:0 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Body 18 [ 26]: a=X-cap: 1 image udptl t38 [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: --- (16 headers 19 lines) --- [Dec 8 13:12:58] DEBUG[8928] acl.c: Found IP address for this socket [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Target address 10.0.0.163 is not local, substituting externip [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.33:5060 [Dec 8 13:12:58] VERBOSE[8928] netsock.c: == Using SIP RTP CoS mark 5 [Dec 8 13:12:58] VERBOSE[8928] netsock.c: == Using UDPTL CoS mark 5 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Setting NAT on RTP to On [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Setting NAT on UDPTL to On [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Allocating new SIP dialog for 1-22953@10.0.0.163 - INVITE (With RTP) [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Begin: parsing SIP "Supported: timer" [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Found SIP option: -timer- [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Matched SIP option: timer [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Sending to 10.0.0.163 : 5061 (NAT) [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Initializing initreq for method INVITE - callid 1-22953@10.0.0.163 [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Using INVITE request as basis request - 1-22953@10.0.0.163 [Dec 8 13:12:58] DEBUG[8928] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM peer WHERE name = ? AND host = ? [Dec 8 13:12:58] DEBUG[8928] res_config_odbc.c: Parameter 1 ('name') = '+111111111' [Dec 8 13:12:58] DEBUG[8928] res_config_odbc.c: Parameter 2 ('host') = 'dynamic' [Dec 8 13:12:58] DEBUG[8928] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM peer WHERE name = ? [Dec 8 13:12:58] DEBUG[8928] res_config_odbc.c: Parameter 1 ('name') = '+111111111' [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Found peer 'in_2' for '+111111111' from 10.0.0.163:5061 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Setting NAT on RTP to Off [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Setting NAT on UDPTL to Off [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing session-level SDP o=- 456789456 0 IN IP4 10.0.0.163... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing session-level SDP s=Cisco SDP 0... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.163... OK. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Found RTP audio format 8 [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Found RTP audio format 18 [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Found RTP audio format 101 [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Found RTP audio format 100 [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Found unknown media description format X-NSE for ID 100 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:100 X-NSE/8000... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=fmtp:100 192-194,200-202... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=sqn:0... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cdsc: 1 audio RTP/AVP 8 18 101 100... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cdsc: 5 image udptl t38... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cpar: a=T38FaxVersion:0... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cpar: a=T38FaxRateManagement:transferredTCF... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cpar: a=T38FaxMaxDatagram:160... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cpar: a=T38FaxUdpEC:t38UDPRedundancy... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=X-sqn:0... UNSUPPORTED. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=X-cap: 1 image udptl t38... UNSUPPORTED. [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729) [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Peer audio RTP is at port 10.0.0.163:6000 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Peer doesn't provide T.38 UDPTL [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: We're settling with these formats: 0x108 (alaw|g729) [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Checking SIP call limits for device [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Updating call counter for incoming call [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: Looking for +222222222 in incoming (domain 10.0.0.33) [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: *** Our native formats are 0x8 (alaw) [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: *** Joint capabilities are 0x108 (alaw|g729) [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: *** Our capabilities are 0x108 (alaw|g729) [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: This channel will not be able to handle video. [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: build_route: Contact hop: Anonymous [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: list_route: hop: [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Incoming INVITE with 'timer' option enabled [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Session-Expires: 1800 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Received Min-SE: 1800 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Session timer started: 5 - 1-22953@10.0.0.163 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: SIP/in_2-00000000: New call is still down.... Trying... [Dec 8 13:12:58] VERBOSE[8928] chan_sip.c: <--- Transmitting (no NAT) to 10.0.0.163:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.163:5061;branch=+222222222-+111111111-73176;received=10.0.0.163 From: Anonymous ;tag=8794645 To: +222222222 Call-ID: 1-22953@10.0.0.163 CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 10.0.0.163:5061;branch=+222222222-+111111111-73176;received=10.0.0.163 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 2 [ 71]: From: Anonymous ;tag=8794645 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 3 [ 57]: To: +222222222 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 4 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 6 [ 29]: Server: Asterisk PBX 1.6.1.11 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 9 [ 14]: Require: timer [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 11 [ 35]: Contact: [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Header 13 [ 0]: [Dec 8 13:12:58] DEBUG[8928] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.0.163:5061 [Dec 8 13:12:58] DEBUG[8912] devicestate.c: No provider found, checking channel drivers for SIP - in_2 [Dec 8 13:12:58] DEBUG[8912] chan_sip.c: Checking device state for peer in_2 [Dec 8 13:12:58] DEBUG[8912] devicestate.c: Changing state for SIP/in_2 - state 1 (Not in use) [Dec 8 13:12:58] DEBUG[8912] devicestate.c: device 'SIP/in_2' state '1' [Dec 8 13:12:58] DEBUG[8940] pbx.c: Launching 'Gosub' [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Executing [+222222222@incoming:1] Gosub("SIP/in_2-00000000", "sub_incoming,s,1") in new stack [Dec 8 13:12:58] DEBUG[8940] app_stack.c: Channel SIP/in_2-00000000 has no datastore, so we're allocating one. [Dec 8 13:12:58] DEBUG[8940] pbx.c: Launching 'Set' [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Executing [s@sub_incoming:1] Set("SIP/in_2-00000000", "Trust_rdnis=1") in new stack [Dec 8 13:12:58] DEBUG[8940] pbx.c: Launching 'Set' [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Executing [s@sub_incoming:2] Set("SIP/in_2-00000000", "Trust_p_asserted_identity=1") in new stack [Dec 8 13:12:58] DEBUG[8940] pbx.c: Launching 'Return' [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Executing [s@sub_incoming:3] Return("SIP/in_2-00000000", "") in new stack [Dec 8 13:12:58] DEBUG[8940] pbx.c: Launching 'Goto' [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Executing [+222222222@incoming:2] Goto("SIP/in_2-00000000", "setup_call,+222222222,1") in new stack [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Goto (setup_call,+222222222,1) [Dec 8 13:12:58] DEBUG[8940] pbx.c: Function result is '' [Dec 8 13:12:58] DEBUG[8940] pbx.c: Function result is '0' [Dec 8 13:12:58] DEBUG[8940] pbx.c: Expression result is '0' [Dec 8 13:12:58] DEBUG[8940] pbx.c: Function result is '' [Dec 8 13:12:58] DEBUG[8940] pbx.c: Function result is '' [Dec 8 13:12:58] DEBUG[8940] pbx.c: Launching 'Set' [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Executing [+222222222@setup_call:1] Set("SIP/in_2-00000000", "CDR(rdnis)=") in new stack [Dec 8 13:12:58] DEBUG[8940] pbx.c: Launching 'Goto' [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Executing [+222222222@setup_call:2] Goto("SIP/in_2-00000000", "outgoing,+222222222,1") in new stack [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Goto (outgoing,+222222222,1) [Dec 8 13:12:58] DEBUG[8940] pbx.c: Launching 'Dial' [Dec 8 13:12:58] VERBOSE[8940] pbx.c: -- Executing [+222222222@outgoing:1] Dial("SIP/in_2-00000000", "SIP/out/+222222222") in new stack [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Dec 8 13:12:58] VERBOSE[8940] netsock.c: == Using SIP RTP CoS mark 5 [Dec 8 13:12:58] VERBOSE[8940] netsock.c: == Using UDPTL CoS mark 5 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Allocating new SIP dialog for 72d6f4331f2fa0414e801ca755e3d9d9@127.0.1.1 - INVITE (With RTP) [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Setting NAT on RTP to Off [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Setting NAT on UDPTL to Off [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 8 13:12:58] DEBUG[8940] acl.c: Found IP address for this socket [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Target address 10.0.0.172 is not local, substituting externip [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.0.33:5060 [Dec 8 13:12:58] DEBUG[8924] app_queue.c: Device 'SIP/in_2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: *** Our native formats are 0x8 (alaw) [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: *** Our capabilities are 0x108 (alaw|g729) [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: This channel will not be able to handle video. [Dec 8 13:12:58] DEBUG[8940] rtp.c: Seeded SDP of 'SIP/out-00000001' with that of 'SIP/in_2-00000000' [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable DIALEDTIME. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable ANSWEREDTIME. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable DIALEDPEERNAME. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable DIALEDPEERNUMBER. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable DIALSTATUS. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable GOSUB_RETVAL. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable Trust_p_asserted_identity. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable Trust_rdnis. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable SIPCALLID. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable SIPDOMAIN. [Dec 8 13:12:58] DEBUG[8940] channel.c: Not copying variable SIPURI. [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Outgoing Call for +222222222 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Updating call counter for outgoing call [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: False Text flag: False [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Dec 8 13:12:58] VERBOSE[8940] chan_sip.c: Audio is at 10.0.0.33 port 12832 [Dec 8 13:12:58] VERBOSE[8940] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Dec 8 13:12:58] VERBOSE[8940] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: -- Done with adding codecs to SDP [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Initializing initreq for method INVITE - callid 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 0 [ 56]: INVITE sip:+222222222@10.0.0.172:5060;user=phone SIP/2.0 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK0d6024d0;rport [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 3 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 4 [ 47]: To: [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 6 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 9 [ 35]: Date: Tue, 08 Dec 2009 12:12:58 GMT [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 11 [ 12]: Min-SE: 1800 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 15 [ 19]: Content-Length: 258 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 16 [ 0]: [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 1 [ 45]: o=root 1018458217 1018458217 IN IP4 10.0.0.33 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 2 [ 23]: s=Asterisk PBX 1.6.1.11 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 3 [ 18]: c=IN IP4 10.0.0.33 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 5 [ 27]: m=audio 12832 RTP/AVP 8 101 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 11 [ 10]: a=sendrecv [Dec 8 13:12:58] VERBOSE[8940] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.0.172:5060: INVITE sip:+222222222@10.0.0.172:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK0d6024d0;rport Max-Forwards: 70 From: "Anonymous" ;tag=as344f3130 To: Contact: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.11 Date: Tue, 08 Dec 2009 12:12:58 GMT Session-Expires: 1800 Min-SE: 1800 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 258 v=0 o=root 1018458217 1018458217 IN IP4 10.0.0.33 s=Asterisk PBX 1.6.1.11 c=IN IP4 10.0.0.33 t=0 0 m=audio 12832 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 0 [ 56]: INVITE sip:+222222222@10.0.0.172:5060;user=phone SIP/2.0 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK0d6024d0;rport [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 3 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 4 [ 47]: To: [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 6 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 9 [ 35]: Date: Tue, 08 Dec 2009 12:12:58 GMT [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 11 [ 12]: Min-SE: 1800 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 15 [ 19]: Content-Length: 258 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Header 16 [ 0]: [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 1 [ 45]: o=root 1018458217 1018458217 IN IP4 10.0.0.33 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 2 [ 23]: s=Asterisk PBX 1.6.1.11 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 3 [ 18]: c=IN IP4 10.0.0.33 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 5 [ 27]: m=audio 12832 RTP/AVP 8 101 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Body 11 [ 10]: a=sendrecv [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Dec 8 13:12:58] DEBUG[8940] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.0.172:5060 [Dec 8 13:12:58] VERBOSE[8940] app_dial.c: -- Called out/+222222222 [Dec 8 13:12:59] VERBOSE[8928] chan_sip.c: <--- SIP read from UDP://10.0.0.172:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK0d6024d0;rport From: "Anonymous" ;tag=as344f3130 To: ;tag=1 Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 102 INVITE Content-Length: 0 <-------------> [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK0d6024d0;rport [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: Header 2 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: Header 3 [ 53]: To: ;tag=1 [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: Header 4 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: Header 7 [ 0]: [Dec 8 13:12:59] VERBOSE[8928] chan_sip.c: --- (7 headers 0 lines) --- [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: *** SIP TIMER: Cancelling retransmission #7 - INVITE (got response) [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '53f4863d3133b028339f9b2c55a8fac2@10.0.0.33' Request 102: Found [Dec 8 13:12:59] DEBUG[8928] chan_sip.c: SIP response 100 to standard invite [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: <--- SIP read from UDP://10.0.0.172:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK0d6024d0;rport From: "Anonymous" ;tag=as344f3130 To: ;tag=1 Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 102 INVITE Contact: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Require: timer Supported: timer Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Length: 498 v=0 o=- 45612357 0 IN IP4 10.0.0.180 s=Cisco SDP 0 c=IN IP4 10.0.0.180 t=0 0 m=audio 6000 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=ptime:20 a=sqn:0 a=cdsc: 1 audio RTP/AVP 8 101 100 a=cdsc: 4 image udptl t38 a=cpar: a=T38FaxVersion:0 a=cpar: a=T38FaxRateManagement:transferredTCF a=cpar: a=T38FaxMaxDatagram:160 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=X-sqn:0 a=X-cap: 1 image udptl t38 <-------------> [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK0d6024d0;rport [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 2 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 3 [ 53]: To: ;tag=1 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 4 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 6 [ 41]: Contact: [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 7 [ 69]: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 8 [ 14]: Require: timer [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 9 [ 16]: Supported: timer [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uac [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 12 [ 19]: Content-Length: 498 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 13 [ 0]: [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 1 [ 32]: o=- 45612357 0 IN IP4 10.0.0.180 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 2 [ 13]: s=Cisco SDP 0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.180 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 5 [ 30]: m=audio 6000 RTP/AVP 8 101 100 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 7 [ 15]: a=fmtp:101 0-15 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 8 [ 23]: a=rtpmap:100 X-NSE/8000 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 9 [ 26]: a=fmtp:100 192-194,200-202 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 11 [ 7]: a=sqn:0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 12 [ 33]: a=cdsc: 1 audio RTP/AVP 8 101 100 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 13 [ 25]: a=cdsc: 4 image udptl t38 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 14 [ 25]: a=cpar: a=T38FaxVersion:0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 15 [ 45]: a=cpar: a=T38FaxRateManagement:transferredTCF [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 16 [ 31]: a=cpar: a=T38FaxMaxDatagram:160 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 17 [ 38]: a=cpar: a=T38FaxUdpEC:t38UDPRedundancy [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 18 [ 9]: a=X-sqn:0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 19 [ 26]: a=X-cap: 1 image udptl t38 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: --- (13 headers 20 lines) --- [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Acked pending invite 102 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Stopping retransmission on '53f4863d3133b028339f9b2c55a8fac2@10.0.0.33' of Request 102: Match Found [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: SIP response 200 to standard invite [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing session-level SDP o=- 45612357 0 IN IP4 10.0.0.180... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing session-level SDP s=Cisco SDP 0... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.180... OK. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Found RTP audio format 8 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Found RTP audio format 101 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Found RTP audio format 100 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Found unknown media description format X-NSE for ID 100 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:100 X-NSE/8000... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=fmtp:100 192-194,200-202... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=sqn:0... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cdsc: 1 audio RTP/AVP 8 101 100... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cdsc: 4 image udptl t38... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cpar: a=T38FaxVersion:0... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cpar: a=T38FaxRateManagement:transferredTCF... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cpar: a=T38FaxMaxDatagram:160... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=cpar: a=T38FaxUdpEC:t38UDPRedundancy... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=X-sqn:0... UNSUPPORTED. [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Processing media-level (audio) SDP a=X-cap: 1 image udptl t38... UNSUPPORTED. [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Peer audio RTP is at port 10.0.0.180:6000 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Peer doesn't provide T.38 UDPTL [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: We have an owner, now see if we need to change this call [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Updating call counter for outgoing call [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: build_route: Contact hop: [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: list_route: hop: [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Session-Expires: 1800 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Refresher: UAC [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Session timer started: 9 - 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Strict routing enforced for session 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: set_destination: set destination to 10.0.0.172, port 5060 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Transmitting (no NAT) to 10.0.0.172:5060: ACK sip:+333333333@10.0.0.172:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK022d6cac;rport Max-Forwards: 70 From: "Anonymous" ;tag=as344f3130 To: ;tag=1 Contact: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.11 Content-Length: 0 --- [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 0 [ 42]: ACK sip:+333333333@10.0.0.172:5060 SIP/2.0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK022d6cac;rport [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 3 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 4 [ 53]: To: ;tag=1 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 6 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 10 [ 0]: [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Trying to put 'ACK sip:+33' onto UDP socket destined for 10.0.0.172:5060 [Dec 8 13:13:00] DEBUG[8912] devicestate.c: No provider found, checking channel drivers for SIP - out [Dec 8 13:13:00] VERBOSE[8940] app_dial.c: -- SIP/out-00000001 answered SIP/in_2-00000000 [Dec 8 13:13:00] DEBUG[8912] chan_sip.c: Checking device state for peer out [Dec 8 13:13:00] DEBUG[8912] devicestate.c: Changing state for SIP/out - state 1 (Not in use) [Dec 8 13:13:00] DEBUG[8940] rtp.c: Setting early bridge SDP of 'SIP/in_2-00000000' with that of 'SIP/out-00000001' [Dec 8 13:13:00] DEBUG[8912] devicestate.c: device 'SIP/out' state '1' [Dec 8 13:13:00] DEBUG[8912] devicestate.c: No provider found, checking channel drivers for SIP - in_2 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: SIP answering channel: SIP/in_2-00000000 [Dec 8 13:13:00] DEBUG[8912] chan_sip.c: Checking device state for peer in_2 [Dec 8 13:13:00] DEBUG[8924] app_queue.c: Device 'SIP/out' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 8 13:13:00] DEBUG[8912] devicestate.c: Changing state for SIP/in_2 - state 1 (Not in use) [Dec 8 13:13:00] DEBUG[8912] devicestate.c: device 'SIP/in_2' state '1' [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Setting framing from config on incoming call [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: ** Our capability: 0x108 (alaw|g729) Video flag: True Text flag: True [Dec 8 13:13:00] DEBUG[8924] app_queue.c: Device 'SIP/in_2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: Audio is at 10.0.0.33 port 14784 [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: Adding codec 0x100 (g729) to SDP [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: -- Done with adding codecs to SDP [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Done building SDP. Settling with this capability: 0x108 (alaw|g729) [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.0.0.163:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.163:5061;branch=+222222222-+111111111-73176;received=10.0.0.163 From: Anonymous ;tag=8794645 To: +222222222 ;tag=as5a3c4f41 Call-ID: 1-22953@10.0.0.163 CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 303 v=0 o=root 368489782 368489782 IN IP4 10.0.0.33 s=Asterisk PBX 1.6.1.11 c=IN IP4 10.0.0.33 t=0 0 m=audio 14784 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 10.0.0.163:5061;branch=+222222222-+111111111-73176;received=10.0.0.163 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 2 [ 71]: From: Anonymous ;tag=8794645 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 3 [ 72]: To: +222222222 ;tag=as5a3c4f41 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 4 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 6 [ 29]: Server: Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 9 [ 14]: Require: timer [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 11 [ 35]: Contact: [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 13 [ 19]: Content-Length: 303 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 14 [ 0]: [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 1 [ 43]: o=root 368489782 368489782 IN IP4 10.0.0.33 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 2 [ 23]: s=Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 3 [ 18]: c=IN IP4 10.0.0.33 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 5 [ 30]: m=audio 14784 RTP/AVP 8 18 101 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 8 [ 19]: a=fmtp:18 annexb=no [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 13 [ 10]: a=sendrecv [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #10 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.0.163:5061 [Dec 8 13:13:00] DEBUG[8940] features.c: bridge answer set, chan answer set [Dec 8 13:13:00] VERBOSE[8940] rtp.c: -- Native bridging SIP/in_2-00000000 and SIP/out-00000001 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Deferring reinvite on SIP '1-22953@10.0.0.163' - It's audio will be redirected to IP 10.0.0.180 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Sending reinvite on SIP '53f4863d3133b028339f9b2c55a8fac2@10.0.0.33' - It's audio soon redirected to IP 10.0.0.163 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Strict routing enforced for session 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: set_destination: set destination to 10.0.0.172, port 5060 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: Audio is at 10.0.0.33 port 12832 [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: -- Done with adding codecs to SDP [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Initializing already initialized SIP dialog 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 (presumably reinvite) [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 0 [ 45]: INVITE sip:+333333333@10.0.0.172:5060 SIP/2.0 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK6c06b748;rport [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 3 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 4 [ 53]: To: ;tag=1 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 6 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 9 [ 14]: Require: timer [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uac [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 11 [ 12]: Min-SE: 1800 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 16 [ 19]: Content-Length: 259 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 17 [ 0]: [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 1 [ 46]: o=root 1018458217 1018458218 IN IP4 10.0.0.163 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 2 [ 23]: s=Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.163 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 5 [ 26]: m=audio 6000 RTP/AVP 8 101 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 11 [ 10]: a=sendrecv [Dec 8 13:13:00] VERBOSE[8940] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.0.172:5060: INVITE sip:+333333333@10.0.0.172:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK6c06b748;rport Max-Forwards: 70 From: "Anonymous" ;tag=as344f3130 To: ;tag=1 Contact: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.1.11 Require: timer Session-Expires: 1800;refresher=uac Min-SE: 1800 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 259 v=0 o=root 1018458217 1018458218 IN IP4 10.0.0.163 s=Asterisk PBX 1.6.1.11 c=IN IP4 10.0.0.163 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 0 [ 45]: INVITE sip:+333333333@10.0.0.172:5060 SIP/2.0 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK6c06b748;rport [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 3 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 4 [ 53]: To: ;tag=1 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 6 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 9 [ 14]: Require: timer [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uac [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 11 [ 12]: Min-SE: 1800 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 16 [ 19]: Content-Length: 259 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Header 17 [ 0]: [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 1 [ 46]: o=root 1018458217 1018458218 IN IP4 10.0.0.163 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 2 [ 23]: s=Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.163 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 5 [ 26]: m=audio 6000 RTP/AVP 8 101 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Body 11 [ 10]: a=sendrecv [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Dec 8 13:13:00] DEBUG[8940] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.0.172:5060 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: <--- SIP read from UDP://10.0.0.163:5061 ---> ACK sip:+999999999@10.0.0.33:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.163:5061;branch=z9hG4bK-22953-1-4 From: sipp ;tag=1 To: sut ;tag=as5a3c4f41 Call-ID: 1-22953@10.0.0.163 CSeq: 1 ACK Contact: sip:sipp@10.0.0.163:5061 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 <-------------> [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 0 [ 41]: ACK sip:+999999999@10.0.0.33:5060 SIP/2.0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.0.163:5061;branch=z9hG4bK-22953-1-4 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 2 [ 43]: From: sipp ;tag=1 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 3 [ 54]: To: sut ;tag=as5a3c4f41 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 4 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 6 [ 33]: Contact: sip:sipp@10.0.0.163:5061 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 8 [ 25]: Subject: Performance Test [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 10 [ 0]: [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: --- (10 headers 0 lines) --- [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Stopping retransmission on '1-22953@10.0.0.163' of Response 1: Match Found [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Sending pending reinvite on '1-22953@10.0.0.163' [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Strict routing enforced for session 1-22953@10.0.0.163 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: set_destination: set destination to 10.0.0.163, port 5061 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: ** Our capability: 0x108 (alaw|g729) Video flag: True Text flag: True [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Audio is at 10.0.0.33 port 14784 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Adding codec 0x100 (g729) to SDP [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: -- Done with adding codecs to SDP [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Done building SDP. Settling with this capability: 0x108 (alaw|g729) [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Initializing already initialized SIP dialog 1-22953@10.0.0.163 (presumably reinvite) [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 0 [ 45]: INVITE sip:+111111111@10.0.0.163:5061 SIP/2.0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK715f86bc;rport [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 3 [ 74]: From: +222222222 ;tag=as5a3c4f41 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 4 [ 69]: To: Anonymous ;tag=8794645 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 6 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 9 [ 14]: Require: timer [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 11 [ 12]: Min-SE: 1800 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 16 [ 19]: Content-Length: 304 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 17 [ 0]: [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 1 [ 44]: o=root 368489782 368489783 IN IP4 10.0.0.180 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 2 [ 23]: s=Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.180 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 5 [ 29]: m=audio 6000 RTP/AVP 8 18 101 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 8 [ 19]: a=fmtp:18 annexb=no [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 13 [ 10]: a=sendrecv [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.0.163:5061: INVITE sip:+111111111@10.0.0.163:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK715f86bc;rport Max-Forwards: 70 From: +222222222 ;tag=as5a3c4f41 To: Anonymous ;tag=8794645 Contact: Call-ID: 1-22953@10.0.0.163 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.11 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 1800 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 304 v=0 o=root 368489782 368489783 IN IP4 10.0.0.180 s=Asterisk PBX 1.6.1.11 c=IN IP4 10.0.0.180 t=0 0 m=audio 6000 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 0 [ 45]: INVITE sip:+111111111@10.0.0.163:5061 SIP/2.0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK715f86bc;rport [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 3 [ 74]: From: +222222222 ;tag=as5a3c4f41 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 4 [ 69]: To: Anonymous ;tag=8794645 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 6 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 9 [ 14]: Require: timer [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 11 [ 12]: Min-SE: 1800 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 16 [ 19]: Content-Length: 304 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 17 [ 0]: [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 1 [ 44]: o=root 368489782 368489783 IN IP4 10.0.0.180 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 2 [ 23]: s=Asterisk PBX 1.6.1.11 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.180 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 5 [ 29]: m=audio 6000 RTP/AVP 8 18 101 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 8 [ 19]: a=fmtp:18 annexb=no [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Body 13 [ 10]: a=sendrecv [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #12 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.0.163:5061 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: <--- SIP read from UDP://10.0.0.163:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK715f86bc;rport From: +222222222 ;tag=as5a3c4f41 To: Anonymous ;tag=8794645;tag=1 Call-ID: 1-22953@10.0.0.163 CSeq: 102 INVITE Content-Length: 0 <-------------> [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK715f86bc;rport [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 2 [ 74]: From: +222222222 ;tag=as5a3c4f41 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 3 [ 75]: To: Anonymous ;tag=8794645;tag=1 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 4 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 7 [ 0]: [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: --- (7 headers 0 lines) --- [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: *** SIP TIMER: Cancelling retransmission #12 - INVITE (got response) [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1-22953@10.0.0.163' Request 102: Found [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: SIP response 100 to RE-invite on outgoing call 1-22953@10.0.0.163 [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: <--- SIP read from UDP://10.0.0.172:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK6c06b748;rport From: "Anonymous" ;tag=as344f3130 To: ;tag=1;tag=1 Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 103 INVITE Content-Length: 0 <-------------> [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK6c06b748;rport [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 2 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 3 [ 59]: To: ;tag=1;tag=1 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 4 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: Header 7 [ 0]: [Dec 8 13:13:00] VERBOSE[8928] chan_sip.c: --- (7 headers 0 lines) --- [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: *** SIP TIMER: Cancelling retransmission #11 - INVITE (got response) [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '53f4863d3133b028339f9b2c55a8fac2@10.0.0.33' Request 103: Found [Dec 8 13:13:00] DEBUG[8928] chan_sip.c: SIP response 100 to RE-invite on outgoing call 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: <--- SIP read from UDP://10.0.0.163:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK715f86bc;rport From: +222222222 ;tag=as5a3c4f41 To: Anonymous ;tag=8794645;tag=1 Call-ID: 1-22953@10.0.0.163 CSeq: 102 INVITE Contact: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Require: timer Supported: timer Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Length: 498 v=0 o=- 45612357 0 IN IP4 10.0.0.163 s=Cisco SDP 0 c=IN IP4 10.0.0.163 t=0 0 m=audio 6000 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=ptime:20 a=sqn:0 a=cdsc: 1 audio RTP/AVP 8 101 100 a=cdsc: 4 image udptl t38 a=cpar: a=T38FaxVersion:0 a=cpar: a=T38FaxRateManagement:transferredTCF a=cpar: a=T38FaxMaxDatagram:160 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=X-sqn:0 a=X-cap: 1 image udptl t38 <-------------> [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK715f86bc;rport [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 2 [ 74]: From: +222222222 ;tag=as5a3c4f41 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 3 [ 75]: To: Anonymous ;tag=8794645;tag=1 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 4 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 6 [ 41]: Contact: [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 7 [ 69]: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 8 [ 14]: Require: timer [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 9 [ 16]: Supported: timer [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uac [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 12 [ 19]: Content-Length: 498 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 13 [ 0]: [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 1 [ 32]: o=- 45612357 0 IN IP4 10.0.0.163 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 2 [ 13]: s=Cisco SDP 0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.163 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 5 [ 30]: m=audio 6000 RTP/AVP 8 101 100 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 7 [ 15]: a=fmtp:101 0-15 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 8 [ 23]: a=rtpmap:100 X-NSE/8000 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 9 [ 26]: a=fmtp:100 192-194,200-202 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 11 [ 7]: a=sqn:0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 12 [ 33]: a=cdsc: 1 audio RTP/AVP 8 101 100 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 13 [ 25]: a=cdsc: 4 image udptl t38 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 14 [ 25]: a=cpar: a=T38FaxVersion:0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 15 [ 45]: a=cpar: a=T38FaxRateManagement:transferredTCF [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 16 [ 31]: a=cpar: a=T38FaxMaxDatagram:160 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 17 [ 38]: a=cpar: a=T38FaxUdpEC:t38UDPRedundancy [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 18 [ 9]: a=X-sqn:0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 19 [ 26]: a=X-cap: 1 image udptl t38 [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: --- (13 headers 20 lines) --- [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Acked pending invite 102 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Stopping retransmission on '1-22953@10.0.0.163' of Request 102: Match Found [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: SIP response 200 to RE-invite on outgoing call 1-22953@10.0.0.163 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Call 1-22953@10.0.0.163 responded to our reinvite without changing SDP version; ignoring SDP. [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Updating call counter for incoming call [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Strict routing enforced for session 1-22953@10.0.0.163 [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: set_destination: set destination to 10.0.0.163, port 5061 [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: Transmitting (no NAT) to 10.0.0.163:5061: ACK sip:+111111111@10.0.0.163:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK09c697ee;rport Max-Forwards: 70 From: +222222222 ;tag=as5a3c4f41 To: Anonymous ;tag=8794645 Contact: Call-ID: 1-22953@10.0.0.163 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.11 Content-Length: 0 --- [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 0 [ 42]: ACK sip:+111111111@10.0.0.163:5061 SIP/2.0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK09c697ee;rport [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 3 [ 74]: From: +222222222 ;tag=as5a3c4f41 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 4 [ 69]: To: Anonymous ;tag=8794645 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 6 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 10 [ 0]: [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Trying to put 'ACK sip:+11' onto UDP socket destined for 10.0.0.163:5061 [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: <--- SIP read from UDP://10.0.0.172:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK6c06b748;rport From: "Anonymous" ;tag=as344f3130 To: ;tag=1;tag=1 Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 103 INVITE Contact: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Require: timer Supported: timer Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Length: 498 v=0 o=- 45612357 0 IN IP4 10.0.0.180 s=Cisco SDP 0 c=IN IP4 10.0.0.180 t=0 0 m=audio 6000 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=ptime:20 a=sqn:0 a=cdsc: 1 audio RTP/AVP 8 101 100 a=cdsc: 4 image udptl t38 a=cpar: a=T38FaxVersion:0 a=cpar: a=T38FaxRateManagement:transferredTCF a=cpar: a=T38FaxMaxDatagram:160 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=X-sqn:0 a=X-cap: 1 image udptl t38 <-------------> [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK6c06b748;rport [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 2 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 3 [ 59]: To: ;tag=1;tag=1 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 4 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 6 [ 41]: Contact: [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 7 [ 69]: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 8 [ 14]: Require: timer [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 9 [ 16]: Supported: timer [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uac [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 12 [ 19]: Content-Length: 498 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 13 [ 0]: [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 1 [ 32]: o=- 45612357 0 IN IP4 10.0.0.180 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 2 [ 13]: s=Cisco SDP 0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.180 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 5 [ 30]: m=audio 6000 RTP/AVP 8 101 100 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 7 [ 15]: a=fmtp:101 0-15 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 8 [ 23]: a=rtpmap:100 X-NSE/8000 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 9 [ 26]: a=fmtp:100 192-194,200-202 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 11 [ 7]: a=sqn:0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 12 [ 33]: a=cdsc: 1 audio RTP/AVP 8 101 100 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 13 [ 25]: a=cdsc: 4 image udptl t38 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 14 [ 25]: a=cpar: a=T38FaxVersion:0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 15 [ 45]: a=cpar: a=T38FaxRateManagement:transferredTCF [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 16 [ 31]: a=cpar: a=T38FaxMaxDatagram:160 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 17 [ 38]: a=cpar: a=T38FaxUdpEC:t38UDPRedundancy [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 18 [ 9]: a=X-sqn:0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Body 19 [ 26]: a=X-cap: 1 image udptl t38 [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: --- (13 headers 20 lines) --- [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Acked pending invite 103 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Stopping retransmission on '53f4863d3133b028339f9b2c55a8fac2@10.0.0.33' of Request 103: Match Found [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: SIP response 200 to RE-invite on outgoing call 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Call 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 responded to our reinvite without changing SDP version; ignoring SDP. [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Updating call counter for outgoing call [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Strict routing enforced for session 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: set_destination: set destination to 10.0.0.172, port 5060 [Dec 8 13:13:01] VERBOSE[8928] chan_sip.c: Transmitting (no NAT) to 10.0.0.172:5060: ACK sip:+333333333@10.0.0.172:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK672e0f16;rport Max-Forwards: 70 From: "Anonymous" ;tag=as344f3130 To: ;tag=1 Contact: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.1.11 Content-Length: 0 --- [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 0 [ 42]: ACK sip:+333333333@10.0.0.172:5060 SIP/2.0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK672e0f16;rport [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 3 [ 59]: From: "Anonymous" ;tag=as344f3130 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 4 [ 53]: To: ;tag=1 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 6 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Header 10 [ 0]: [Dec 8 13:13:01] DEBUG[8928] chan_sip.c: Trying to put 'ACK sip:+33' onto UDP socket destined for 10.0.0.172:5060 [Dec 8 13:13:02] VERBOSE[8928] chan_sip.c: <--- SIP read from UDP://10.0.0.172:5060 ---> INVITE sip:+111111111@10.0.0.33 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.33:5060;branch=+111111111-+333333333-54623 From: ;tag=4532188654 To: "Anonymous" ;tag=jk868tj76 Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 1 INVITE Max-Forwards: 70 Supported: timer Session-Expires: 1800;refresher=uas Contact: Diversion: +222222222 ;reason=unconditional;counter=1;privacy=off Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Content-Type: application/sdp Content-Length: 488 v=0 o=- 45612357 1 IN IP4 10.0.0.190 s=Cisco SDP 0 c=IN IP4 10.0.0.190 t=0 0 m=image 38102 udptl t38 a=T38FaxVersion:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:160 a=T38FaxUdpEC:t38UDPRedundancy a=sqn:0 a=cdsc: 1 audio RTP/AVP 8 101 100 a=cdsc: 4 image udptl t38 a=cpar: a=T38FaxVersion:0 a=cpar: a=T38FaxRateManagement:transferredTCF a=cpar: a=T38FaxMaxDatagram:160 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=X-sqn:0 a=X-cap: 1 image udptl t38 <-------------> [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 0 [ 39]: INVITE sip:+111111111@10.0.0.33 SIP/2.0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=+111111111-+333333333-54623 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 2 [ 64]: From: ;tag=4532188654 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 3 [ 61]: To: "Anonymous" ;tag=jk868tj76 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 4 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 7 [ 16]: Supported: timer [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 8 [ 35]: Session-Expires: 1800;refresher=uas [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 9 [ 41]: Contact: [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 10 [ 96]: Diversion: +222222222 ;reason=unconditional;counter=1;privacy=off [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 11 [ 69]: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 13 [ 19]: Content-Length: 488 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 14 [ 0]: [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 1 [ 32]: o=- 45612357 1 IN IP4 10.0.0.190 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 2 [ 13]: s=Cisco SDP 0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.190 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 5 [ 23]: m=image 38102 udptl t38 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 6 [ 17]: a=T38FaxVersion:0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 7 [ 37]: a=T38FaxRateManagement:transferredTCF [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 8 [ 23]: a=T38FaxMaxDatagram:160 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 9 [ 30]: a=T38FaxUdpEC:t38UDPRedundancy [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 10 [ 7]: a=sqn:0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 11 [ 33]: a=cdsc: 1 audio RTP/AVP 8 101 100 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 12 [ 25]: a=cdsc: 4 image udptl t38 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 13 [ 25]: a=cpar: a=T38FaxVersion:0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 14 [ 45]: a=cpar: a=T38FaxRateManagement:transferredTCF [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 15 [ 31]: a=cpar: a=T38FaxMaxDatagram:160 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 16 [ 38]: a=cpar: a=T38FaxUdpEC:t38UDPRedundancy [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 17 [ 9]: a=X-sqn:0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Body 18 [ 26]: a=X-cap: 1 image udptl t38 [Dec 8 13:13:02] VERBOSE[8928] chan_sip.c: --- (14 headers 19 lines) --- [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Begin: parsing SIP "Supported: timer" [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Found SIP option: -timer- [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Matched SIP option: timer [Dec 8 13:13:02] VERBOSE[8928] chan_sip.c: Sending to 10.0.0.33 : 5060 (no NAT) [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Initializing initreq for method INVITE - callid 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing session-level SDP o=- 45612357 1 IN IP4 10.0.0.190... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing session-level SDP s=Cisco SDP 0... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.0.190... OK. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Dec 8 13:13:02] VERBOSE[8928] chan_sip.c: Got T.38 offer in SDP in dialog 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: FaxVersion: 0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=T38FaxVersion:0... OK. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: RateManagement: transferredTCF [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=T38FaxRateManagement:transferredTCF... OK. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: FaxMaxDatagram: 160 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=T38FaxMaxDatagram:160... OK. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: UDP EC: t38UDPRedundancy [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=T38FaxUdpEC:t38UDPRedundancy... OK. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=sqn:0... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=cdsc: 1 audio RTP/AVP 8 101 100... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=cdsc: 4 image udptl t38... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=cpar: a=T38FaxVersion:0... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=cpar: a=T38FaxRateManagement:transferredTCF... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=cpar: a=T38FaxMaxDatagram:160... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=cpar: a=T38FaxUdpEC:t38UDPRedundancy... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=X-sqn:0... UNSUPPORTED. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Processing media-level (image) SDP a=X-cap: 1 image udptl t38... UNSUPPORTED. [Dec 8 13:13:02] VERBOSE[8928] chan_sip.c: Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) [Dec 8 13:13:02] VERBOSE[8928] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Dec 8 13:13:02] VERBOSE[8928] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Peer T.38 UDPTL is at port 10.0.0.190:38102 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: T38 state changed to 2 on channel SIP/out-00000001 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Have T.38 but no audio, accepting offer anyway [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Got a SIP re-invite for call 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Incoming INVITE with 'timer' option enabled [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Session-Expires: 1800 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Refresher: UAS [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Restarting session-timers on a refresh - 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Session timer stopped: -1 - 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Session timer started: 13 - 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: SIP/out-00000001: This call is UP.... [Dec 8 13:13:02] VERBOSE[8928] chan_sip.c: <--- Transmitting (no NAT) to 10.0.0.33:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.33:5060;branch=+111111111-+333333333-54623;received=10.0.0.172 From: ;tag=4532188654 To: "Anonymous" ;tag=jk868tj76 Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=+111111111-+333333333-54623;received=10.0.0.172 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 2 [ 64]: From: ;tag=4532188654 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 3 [ 61]: To: "Anonymous" ;tag=jk868tj76 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 4 [ 51]: Call-ID: 53f4863d3133b028339f9b2c55a8fac2@10.0.0.33 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 6 [ 29]: Server: Asterisk PBX 1.6.1.11 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 9 [ 14]: Require: timer [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 11 [ 35]: Contact: [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Header 13 [ 0]: [Dec 8 13:13:02] DEBUG[8928] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.0.33:5060 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: T38 state changed to 1 on channel SIP/in_2-00000000 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Strict routing enforced for session 1-22953@10.0.0.163 [Dec 8 13:13:02] VERBOSE[8940] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 8 13:13:02] VERBOSE[8940] chan_sip.c: set_destination: set destination to 10.0.0.163, port 5061 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: T.38 UDPTL is at 10.0.0.33 port 38507 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Done building SDP. Settling with this capability: 0x0 (nothing) [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Initializing already initialized SIP dialog 1-22953@10.0.0.163 (presumably reinvite) [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 0 [ 45]: INVITE sip:+111111111@10.0.0.163:5061 SIP/2.0 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK180ab319;rport [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 3 [ 74]: From: +222222222 ;tag=as5a3c4f41 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 4 [ 69]: To: Anonymous ;tag=8794645 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 6 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 9 [ 14]: Require: timer [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 11 [ 12]: Min-SE: 1800 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 16 [ 19]: Content-Length: 258 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 17 [ 0]: [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 1 [ 44]: o=root 368489782 368489784 IN IP4 10.0.0.180 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 2 [ 23]: s=Asterisk PBX 1.6.1.11 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.180 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 5 [ 23]: m=image 38507 udptl t38 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 6 [ 17]: a=T38FaxVersion:0 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 7 [ 20]: a=T38MaxBitRate:2400 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 8 [ 37]: a=T38FaxRateManagement:transferredTCF [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 9 [ 22]: a=T38FaxMaxDatagram:82 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 10 [ 23]: a=T38FaxUdpEC:t38UDPFEC [Dec 8 13:13:02] VERBOSE[8940] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.0.163:5061: INVITE sip:+111111111@10.0.0.163:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK180ab319;rport Max-Forwards: 70 From: +222222222 ;tag=as5a3c4f41 To: Anonymous ;tag=8794645 Contact: Call-ID: 1-22953@10.0.0.163 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.1.11 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 1800 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 258 v=0 o=root 368489782 368489784 IN IP4 10.0.0.180 s=Asterisk PBX 1.6.1.11 c=IN IP4 10.0.0.180 t=0 0 m=image 38507 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:2400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:82 a=T38FaxUdpEC:t38UDPFEC --- [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 0 [ 45]: INVITE sip:+111111111@10.0.0.163:5061 SIP/2.0 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/UDP 10.0.0.33:5060;branch=z9hG4bK180ab319;rport [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 3 [ 74]: From: +222222222 ;tag=as5a3c4f41 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 4 [ 69]: To: Anonymous ;tag=8794645 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 5 [ 35]: Contact: [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 6 [ 27]: Call-ID: 1-22953@10.0.0.163 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.1.11 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 9 [ 14]: Require: timer [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 11 [ 12]: Min-SE: 1800 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 16 [ 19]: Content-Length: 258 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Header 17 [ 0]: [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 0 [ 3]: v=0 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 1 [ 44]: o=root 368489782 368489784 IN IP4 10.0.0.180 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 2 [ 23]: s=Asterisk PBX 1.6.1.11 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.0.180 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 5 [ 23]: m=image 38507 udptl t38 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 6 [ 17]: a=T38FaxVersion:0 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 7 [ 20]: a=T38MaxBitRate:2400 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 8 [ 37]: a=T38FaxRateManagement:transferredTCF [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 9 [ 22]: a=T38FaxMaxDatagram:82 [Dec 8 13:13:02] DEBUG[8940] chan_sip.c: Body 10 [ 23]: a=T38FaxUdpEC:t38UDPFEC