Sip read: INVITE sip:200@192.168.16.71;transport=UDP;user=phone SIP/2.0 From: "10" ;tag=4610a8c0-13c4-69b81-19ce5070-5930 To: "200" Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.16.70:5060;rport;branch=z9hG4bK-69b81-19ce5070-49a9 Max-Forwards: 70 Supported: replaces,100rel,timer Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS User-Agent: Swissvoice IP10S Contact: "10" Session-Expires: 3600 Content-Type: application/SDP Content-Length: 212 v=0 o=rtp/1 433025 433025 IN IP4 192.168.16.70 s=- c=IN IP4 192.168.16.70 t=0 0 m=audio 50000 RTP/AVP 18 0 8 a=ptime:40 a=SilenceSupp:on a=rtpmap:18 g729/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 14 headers, 11 lines Using latest request as basis request Sending to 192.168.16.70 : 5060 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.16.70:5060;branch=z9hG4bK-69b81-19ce5070-49a9 From: "10" ;tag=4610a8c0-13c4-69b81-19ce5070-5930 To: "200";tag=as7edd4a42 Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="7236741f" Content-Length: 0 to 192.168.16.70:5060 Scheduling destruction of call '1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70' in 15000 ms Found user '201' pbx*CLI> Sip read: ACK sip:200@192.168.16.71;transport=UDP SIP/2.0 From: "10" ;tag=4610a8c0-13c4-69b81-19ce5070-5930 To: "200";tag=as7edd4a42 Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70 CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.16.70:5060;rport;branch=z9hG4bK-69b81-19ce5070-49a9 Max-Forwards: 70 User-Agent: Swissvoice IP10S Contact: "10" Content-Length: 0 10 headers, 0 lines pbx*CLI> Sip read: INVITE sip:200@192.168.16.71;transport=UDP;user=phone SIP/2.0 From: "10" ;tag=4610a8c0-13c4-69b81-19ce5070-5930 To: "200" Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70 CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.16.70:5060;rport;branch=z9hG4bK-69b81-19ce50fe-76cf Max-Forwards: 70 Supported: replaces,100rel,timer Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS User-Agent: Swissvoice IP10S Contact: "10" Proxy-Authorization: Digest username="201", realm="asterisk", nonce="7236741f", uri="sip:200@192.168.16.71;transport=UDP", response="26226bb482414f37d233eb75fa2b4416", algorithm=MD5 Session-Expires: 3600 Content-Type: application/SDP Content-Length: 212 v=0 o=rtp/1 433025 433025 IN IP4 192.168.16.70 s=- c=IN IP4 192.168.16.70 t=0 0 m=audio 50000 RTP/AVP 18 0 8 a=ptime:40 a=SilenceSupp:on a=rtpmap:18 g729/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 15 headers, 11 lines Using latest request as basis request Sending to 192.168.16.70 : 5060 (non-NAT) Found user '201' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Peer audio RTP is at port 192.168.16.70:50000 Found description format g729 Found description format pcmu Found description format pcma Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 200 in sip-users list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.70:5060;branch=z9hG4bK-69b81-19ce50fe-76cf From: "10" ;tag=4610a8c0-13c4-69b81-19ce5070-5930 To: "200" Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.16.70:5060 Destroying call '5360421d5494a8ae6bce62e33abe5c60@192.168.16.71' Sep 27 21:36:23 NOTICE[56557]: app_dial.c:777 dial_exec: Unable to create channel of type 'SIP' pbx*CLI> Sip read: 0 headers, 0 lines Sep 27 21:36:33 WARNING[56557]: pbx.c:1952 ast_pbx_run: Timeout, but no rule 't' in context 'sip-users' Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.16.70:5060;branch=z9hG4bK-69b81-19ce50fe-76cf From: "10" ;tag=4610a8c0-13c4-69b81-19ce5070-5930 To: "200";tag=as5501d9cd Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.16.70:5060 pbx*CLI> Sip read: ACK sip:200@192.168.16.71;transport=UDP SIP/2.0 From: "10" ;tag=4610a8c0-13c4-69b81-19ce5070-5930 To: "200";tag=as5501d9cd Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70 CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.16.70:5060;rport;branch=z9hG4bK-69b81-19ce50fe-76cf Max-Forwards: 70 User-Agent: Swissvoice IP10S Contact: "10" Content-Length: 0 10 headers, 0 lines Destroying call '1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70' pbx*CLI> Sip read: 0 headers, 0 lines