Audio is at 200.51.41.133 port 17454 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 200.51.200.245:5060: INVITE sip:02994437843@serroca.ip-tel.com.ar SIP/2.0 Via: SIP/2.0/UDP 200.51.41.133:5060;branch=z9hG4bK305f089d;rport From: "Cristian Farias" ;tag=as0713175d To: Contact: Call-ID: 14aba77933f06c21301682592c131ac0@200.51.41.133 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 07 Nov 2007 17:40:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 309 v=0 o=root 4013 4013 IN IP4 200.51.41.133 s=session c=IN IP4 200.51.41.133 t=0 0 m=audio 17454 RTP/AVP 0 97 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 2941605113/02994437843 hsipbx*CLI> <--- SIP read from 200.51.200.245:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 200.51.41.133:5060;branch=z9hG4bK305f089d;rport=5060;received=10.10.0.5 From: "Cristian Farias" ;tag=as0713175d To: ;tag=4baa1850736c757568bdba88a3c615b0.8f33 Call-ID: 14aba77933f06c21301682592c131ac0@200.51.41.133 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sipserverzte.ip-tel.com.ar", nonce="473217b61ddb3432470d239d96fca5ac06531e37" Server: Sip EXpress router (0.9.3 (i386/freebsd)) Content-Length: 0 Warning: 392 200.51.200.245:5060 "Noisy feedback tells: pid=25103 req_src_ip=10.10.0.5 req_src_port=5060 in_uri=sip:02994437843@serroca.ip-tel.com.ar out_uri=sip:02994437843@serroca.ip-tel.com.ar via_cnt==1" <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 200.51.200.245:5060: ACK sip:02994437843@serroca.ip-tel.com.ar SIP/2.0 Via: SIP/2.0/UDP 200.51.41.133:5060;branch=z9hG4bK305f089d;rport From: "Cristian Farias" ;tag=as0713175d To: ;tag=4baa1850736c757568bdba88a3c615b0.8f33 Contact: Call-ID: 14aba77933f06c21301682592c131ac0@200.51.41.133 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at 200.51.41.133 port 17454 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 200.51.200.245:5060: INVITE sip:02994437843@serroca.ip-tel.com.ar SIP/2.0 Via: SIP/2.0/UDP 200.51.41.133:5060;branch=z9hG4bK13b2d710;rport From: "Cristian Farias" ;tag=as0713175d To: Contact: Call-ID: 14aba77933f06c21301682592c131ac0@200.51.41.133 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="2941605113", realm="sipserverzte.ip-tel.com.ar", algorithm=MD5, uri="sip:02994437843@serroca.ip-tel.com.ar", nonce="473217b61ddb3432470d239d96fca5ac06531e37", response="c467e624808c1420d81fba5c7466ff62", opaque="" Date: Wed, 07 Nov 2007 17:40:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 309 v=0 o=root 4013 4014 IN IP4 200.51.41.133 s=session c=IN IP4 200.51.41.133 t=0 0 m=audio 17454 RTP/AVP 0 97 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- hsipbx*CLI> <--- SIP read from 200.51.200.245:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 200.51.41.133:5060;branch=z9hG4bK13b2d710;rport=5060;received=10.10.0.5 From: "Cristian Farias" ;tag=as0713175d To: Call-ID: 14aba77933f06c21301682592c131ac0@200.51.41.133 CSeq: 103 INVITE Server: Sip EXpress router (0.9.3 (i386/freebsd)) Content-Length: 0 Warning: 392 200.51.200.245:5060 "Noisy feedback tells: pid=25107 req_src_ip=10.10.0.5 req_src_port=5060 in_uri=sip:02994437843@serroca.ip-tel.com.ar out_uri=sip:02994437843@200.51.200.244:5060 via_cnt==1" <-------------> --- (9 headers 0 lines) --- hsipbx*CLI> <--- SIP read from 200.51.200.245:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 200.51.41.133:5060;received=10.10.0.5;branch=z9hG4bK13b2d710;rport=5060 From: "Cristian Farias" ;tag=as0713175d To: ;tag=as025ae418 Call-ID: 14aba77933f06c21301682592c131ac0@200.51.41.133 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 268 v=0 o=root 29580 29580 IN IP4 200.51.200.244 s=session c=IN IP4 200.51.200.244 t=0 0 m=audio 12262 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (12 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 200.51.200.244:12262 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x40c (ulaw|alaw|ilbc), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 200.51.200.244:12262 -- SIP/2941605113-08260ee8 is making progress passing it to SIP/342-08248170