pbx*CLI> sip debug peer 2351 SIP Debugging Enabled for IP: 84.42.163.115:5060 We're at 86.49.60.237 port 10062 Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (NAT) to 84.42.163.115:5060: INVITE sip:2351@84.42.163.115:5060 SIP/2.0 Via: SIP/2.0/UDP 86.49.60.237:5060;branch=z9hG4bK0bde2351;rport From: "Dvorak Martin" ;tag=as107d0632 To: Contact: Call-ID: 0ad40ea36fdbdc6a7af40b5e5d6fc844@pbx.crs-net.cz CSeq: 102 INVITE User-Agent: CRS-PBX Max-Forwards: 70 Date: Thu, 13 Apr 2006 14:05:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 260 v=0 o=root 9405 9405 IN IP4 86.49.60.237 s=session c=IN IP4 86.49.60.237 t=0 0 m=audio 10062 RTP/AVP 18 3 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 84.42.163.115:5060: SIP/2.0 100 Trying To: From: "Dvorak Martin" ;tag=as107d0632 Call-ID: 0ad40ea36fdbdc6a7af40b5e5d6fc844@pbx.crs-net.cz CSeq: 102 INVITE Via: SIP/2.0/UDP 86.49.60.237:5060;branch=z9hG4bK0bde2351 Server: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (8 headers 0 lines)--- pbx*CLI> <-- SIP read from 84.42.163.115:5060: SIP/2.0 180 Ringing To: ;tag=b3ed04ecbfa7db84i0 From: "Dvorak Martin" ;tag=as107d0632 Call-ID: 0ad40ea36fdbdc6a7af40b5e5d6fc844@pbx.crs-net.cz CSeq: 102 INVITE Via: SIP/2.0/UDP 86.49.60.237:5060;branch=z9hG4bK0bde2351 Server: Sipura/SPA2100-3.2.5(d) Remote-Party-ID: 2351 ;screen=yes;party=called Content-Length: 0 --- (9 headers 0 lines)--- pbx*CLI> <-- SIP read from 84.42.163.115:5060: SUBSCRIBE sip:pbx.crs-net.cz SIP/2.0 Via: SIP/2.0/UDP 84.42.163.115:5060;branch=z9hG4bK-f4ea65df From: 2351 ;tag=4958d15b85b56450 To: 2351 Call-ID: 7c054277-37d045e4@84.42.163.115 CSeq: 61492 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="2351",realm="pbx.crs-net.cz",nonce="11e03456",uri="sip:pbx.crs-net.cz",algorithm=MD5,response="dd47eaf40ad9e778ba76c59080fec158" Contact: 2351 Expires: 2147483647 Event: message-summary User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (13 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 84.42.163.115 : 5060 (non-NAT) Transmitting (NAT) to 84.42.163.115:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 84.42.163.115:5060;branch=z9hG4bK-f4ea65df;received=84.42.163.115 From: 2351 ;tag=4958d15b85b56450 To: 2351 ;tag=as02cd6532 Call-ID: 7c054277-37d045e4@84.42.163.115 CSeq: 61492 SUBSCRIBE User-Agent: CRS-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="pbx.crs-net.cz", nonce="616bd043" Content-Length: 0 --- Scheduling destruction of call '7c054277-37d045e4@84.42.163.115' in 15000 ms Found user '2351' pbx*CLI> <-- SIP read from 84.42.163.115:5060: SIP/2.0 200 OK To: ;tag=b3ed04ecbfa7db84i0 From: "Dvorak Martin" ;tag=as107d0632 Call-ID: 0ad40ea36fdbdc6a7af40b5e5d6fc844@pbx.crs-net.cz CSeq: 102 INVITE Via: SIP/2.0/UDP 86.49.60.237:5060;branch=z9hG4bK0bde2351 Contact: 2351 Server: Sipura/SPA2100-3.2.5(d) Remote-Party-ID: 2351 ;screen=yes;party=called Content-Length: 264 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 110872833 110872833 IN IP4 84.42.163.115 s=- c=IN IP4 84.42.163.115 t=0 0 m=audio 16482 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (13 headers 13 lines)--- Found RTP audio format 18 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 84.42.163.115:16482 Found description format G729a Found description format NSE Found description format telephone-event Capabilities: us - 0x102 (gsm|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 84.42.163.115, port 5060 Transmitting (NAT) to 84.42.163.115:5060: ACK sip:2351@84.42.163.115:5060 SIP/2.0 Via: SIP/2.0/UDP 86.49.60.237:5060;branch=z9hG4bK38e4795e;rport From: "Dvorak Martin" ;tag=as107d0632 To: ;tag=b3ed04ecbfa7db84i0 Contact: Call-ID: 0ad40ea36fdbdc6a7af40b5e5d6fc844@pbx.crs-net.cz CSeq: 102 ACK User-Agent: CRS-PBX Max-Forwards: 70 Content-Length: 0 --- Apr 13 16:05:22 WARNING[9506]: chan_sip.c:12842 sip_getheader: SIPGetHeader is deprecated, use the SIP_HEADER function instead. pbx*CLI> Disconnected from Asterisk server [root@pbx ~]#