<-------------> --- (7 headers 0 lines) --- samuel*CLI> <--- SIP read from 11.222.33.3:46922 ---> INVITE sip:01234565107@11.222.32.165 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.51:46922;branch=z9hG4bK-d8754z-fd1a3d0df638233e-1---d8754z-;rport Max-Forwards: 70 Contact: To: "MIB +431234565107" From: "fobar samuel";tag=50394a7f Call-ID: YmVjNWIyM2Y0Njg0ODQ2MjU1MWVjYmY1ZGQyMzk0OWU. 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=- 6 2 IN IP4 11.222.33.3 s=CounterPath eyeBeam 1.5 c=IN IP4 11.222.33.3 t=0 0 m=audio 26638 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:B948317085BB44BB9EBA6ADD60C40677 <-------------> --- (13 headers 12 lines) --- Sending to 11.222.33.3 : 46922 (NAT) Using INVITE request as basis request - YmVjNWIyM2Y0Njg0ODQ2MjU1MWVjYmY1ZGQyMzk0OWU. 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 11.222.33.3:26638 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 11.222.33.3:26638 Looking for 01234565107 in direct (domain 11.222.32.165) find_pidflo wurde aufgerufen. kein pidf-lo vorhanden PIDF-LO Document: list_route: hop: samuel*CLI> <--- 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-fd1a3d0df638233e-1---d8754z-;received=11.222.33.3;rport=46922 From: "fobar samuel";tag=50394a7f To: "MIB +431234565107" Call-ID: YmVjNWIyM2Y0Njg0ODQ2MjU1MWVjYmY1ZGQyMzk0OWU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [01234565107@direct:1] Dial("SIP/fobar-09734180", "Dahdi/g2/01234565107") in new stack -- Making new call for cr 32774 -- Requested transfer capability: 0x00 - SPEECH > Protocol Discriminator: Q.931 (8) len=44 > Call Ref: len= 2 (reference 6/0x6) (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 0c 80 30 36 36 34 32 31 33 35 31 30 37] > Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '01234565107' ] q931.c:3092 q931_setup: call 32774 on channel 1 enters state 1 (Call Initiated) -- Called g2/01234565107 Audio is at 11.222.32.165 port 11904 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 samuel*CLI> <--- 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-fd1a3d0df638233e-1---d8754z-;received=11.222.33.3;rport=46922 From: "fobar samuel";tag=50394a7f To: "MIB +431234565107";tag=as3b58117b Call-ID: YmVjNWIyM2Y0Njg0ODQ2MjU1MWVjYmY1ZGQyMzk0OWU. 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 11904 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 6/0x6) (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 32774 on channel 1 enters state 2 (Overlap sending) < Protocol Discriminator: Q.931 (8) len=5 < Call Ref: len= 2 (reference 6/0x6) (Terminator) < Message type: ALERTING (1) q931.c:3554 q931_receive: call 32774 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-fd1a3d0df638233e-1---d8754z-;received=11.222.33.3;rport=46922 From: "fobar samuel";tag=50394a7f To: "MIB +431234565107";tag=as3b58117b Call-ID: YmVjNWIyM2Y0Njg0ODQ2MjU1MWVjYmY1ZGQyMzk0OWU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0