<--- SIP read from 11.222.33.3:46922 ---> INVITE sip:069911160036@11.222.32.165 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.51:46922;branch=z9hG4bK-d8754z-8a7f3865651c476f-1---d8754z-;rport Max-Forwards: 70 Contact: To: "fobar" From: "fobar samuel";tag=e742596b Call-ID: M2JiNDUzODEwNTM4YjkyZDBmZWRjZDhiYzNiZTA0NmU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1101l stamp 49847 Content-Length: 285 v=0 o=- 5 2 IN IP4 11.222.33.3 s=CounterPath eyeBeam 1.5 c=IN IP4 11.222.33.3 t=0 0 m=audio 19558 RTP/AVP 0 8 18 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:877240B526EA49B6AD2DF121BD776B6F <-------------> --- (13 headers 12 lines) --- Sending to 11.222.33.3 : 46922 (NAT) Using INVITE request as basis request - M2JiNDUzODEwNTM4YjkyZDBmZWRjZDhiYzNiZTA0NmU. Found user 'fobar' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 11.222.33.3:19558 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x3f1ffe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 11.222.33.3:19558 Looking for 069911160036 in direct (domain 11.222.32.165) find_pidflo wurde aufgerufen. kein pidf-lo vorhanden PIDF-LO Document: list_route: hop: <--- Transmitting (NAT) to 11.222.33.3:46922 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.0.51:46922;branch=z9hG4bK-d8754z-8a7f3865651c476f-1---d8754z-;received=11.222.33.3;rport=46922 From: "fobar samuel";tag=e742596b To: "fobar" Call-ID: M2JiNDUzODEwNTM4YjkyZDBmZWRjZDhiYzNiZTA0NmU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [069911160036@direct:1] Dial("SIP/fobar-09734180", "Dahdi/g2/069911160036") in new stack -- Making new call for cr 32773 -- Requested transfer capability: 0x00 - SPEECH > Protocol Discriminator: Q.931 (8) len=45 > Call Ref: len= 2 (reference 5/0x5) (Originator) > Message type: SETUP (5) > [04 03 80 90 a3] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) > User information layer 1: A-Law (35) > [18 03 a9 83 81] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 > ChanSel: As indicated in following octets > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 1 ] > [6c 0d 21 80 36 39 39 31 31 31 36 30 30 33 36] > Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) > Presentation: Presentation permitted, user number not screened (0) '69911160036' ] > [70 0d 80 30 36 39 39 31 31 31 36 30 30 33 36] > Called Number (len=15) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '069911160036' ] q931.c:3092 q931_setup: call 32773 on channel 1 enters state 1 (Call Initiated) -- Called g2/069911160036 Audio is at 11.222.32.165 port 10140 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 11.222.33.3:46922 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.0.51:46922;branch=z9hG4bK-d8754z-8a7f3865651c476f-1---d8754z-;received=11.222.33.3;rport=46922 From: "fobar samuel";tag=e742596b To: "fobar";tag=as5f95f34e Call-ID: M2JiNDUzODEwNTM4YjkyZDBmZWRjZDhiYzNiZTA0NmU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 313 v=0 o=root 17191 17191 IN IP4 11.222.32.165 s=session c=IN IP4 11.222.32.165 t=0 0 m=audio 10140 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 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 <------------> < Protocol Discriminator: Q.931 (8) len=10 < Call Ref: len= 2 (reference 5/0x5) (Terminator) < Message type: SETUP ACKNOWLEDGE (13) < [18 03 a9 83 81] < Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 < ChanSel: As indicated in following octets < Ext: 1 Coding: 0 Number Specified Channel Type: 3 < Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3849 q931_receive: call 32773 on channel 1 enters state 2 (Overlap sending) < Protocol Discriminator: Q.931 (8) len=9 < Call Ref: len= 2 (reference 5/0x5) (Terminator) < Message type: CALL PROCEEDING (2) < [1e 02 84 88] < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) < Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 30 (cs0, Progress Indicator) q931.c:3641 q931_receive: call 32773 on channel 1 enters state 3 (Outgoing call Proceeding) Klaus: received AST_CONTROL_PROCEEDING on channel DAHDI/1-1 -- DAHDI/1-1 is proceeding passing it to SIP/fobar-09734180 Klaus: received AST_CONTROL_PROGRESS on channel DAHDI/1-1 -- DAHDI/1-1 is making progress passing it to SIP/fobar-09734180 < Protocol Discriminator: Q.931 (8) len=5 < Call Ref: len= 2 (reference 5/0x5) (Terminator) < Message type: ALERTING (1) q931.c:3554 q931_receive: call 32773 on channel 1 enters state 4 (Call Delivered) Klaus: received AST_CONTROL_RINGING on channel DAHDI/1-1 -- DAHDI/1-1 is ringing <--- Transmitting (NAT) to 11.222.33.3:46922 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.0.51:46922;branch=z9hG4bK-d8754z-8a7f3865651c476f-1---d8754z-;received=11.222.33.3;rport=46922 From: "fobar samuel";tag=e742596b To: "fobar";tag=as5f95f34e Call-ID: M2JiNDUzODEwNTM4YjkyZDBmZWRjZDhiYzNiZTA0NmU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0