<--- SIP read from 172.16.1.253:5060 ---> INVITE sip:9999@172.16.1.43;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.1.253;branch=z9hG4bKac501626729 Max-Forwards: 70 From: ;tag=1c501623461 To: Call-ID: 501623120312000194830@172.16.1.253 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.00A.024 Content-Type: application/sdp Content-Disposition: session Content-Length: 521 v=0 o=AudiocodesGW 501611097 501610983 IN IP4 172.16.1.253 s=Phone-Call c=IN IP4 172.16.1.253 t=0 0 m=audio 6050 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:6051 IN IP4 172.16.1.253 m=image 6052 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:122 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (14 headers 22 lines) --- Sending to 172.16.1.253 : 5060 (no NAT) Using INVITE request as basis request - 501623120312000194830@172.16.1.253 Found peer 'acodes1' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Got T.38 offer in SDP in dialog 501623120312000194830@172.16.1.253 Peer audio RTP is at port 172.16.1.253:6050 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.1.253:6050 Looking for 9999 in entrantes (domain 172.16.1.43) list_route: hop: <--- Transmitting (no NAT) to 172.16.1.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.1.253;branch=z9hG4bKac501626729;received=172.16.1.253 From: ;tag=1c501623461 To: Call-ID: 501623120312000194830@172.16.1.253 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [9999@entrantes:1] Goto("SIP/acodes1-b7bfde18", "inicial|s|1") in new stack -- Goto (inicial,s,1) -- Executing [s@inicial:1] Wait("SIP/acodes1-b7bfde18", "1") in new stack -- Executing [s@inicial:2] Answer("SIP/acodes1-b7bfde18", "") in new stack pbx*CLI> <--- Reliably Transmitting (no NAT) to 172.16.1.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.1.253;branch=z9hG4bKac501626729;received=172.16.1.253 From: ;tag=1c501623461 To: ;tag=as700dcc3d Call-ID: 501623120312000194830@172.16.1.253 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 346 v=0 o=root 7787 7787 IN IP4 172.16.1.43 s=session c=IN IP4 172.16.1.43 t=0 0 m=image 4800 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:122 a=T38FaxMaxDatagram:122 a=T38FaxUdpEC:t38UDPRedundancy <------------> -- Executing [s@inicial:3] Wait("SIP/acodes1-b7bfde18", "1") in new stack pbx-upcn*CLI> <--- SIP read from 172.16.1.253:5060 ---> ACK sip:9999@172.16.1.43 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.253;branch=z9hG4bKac504212125 Max-Forwards: 70 From: ;tag=1c501623461 To: ;tag=as700dcc3d Call-ID: 501623120312000194830@172.16.1.253 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.00A.024 Content-Length: 0 <-------------> --- (12 headers 0 lines) ---