mayamini-lx*CLI> <--- SIP read from 10.2.110.195:5060 ---> INVITE sip:1234@192.168.9.1 SIP/2.0 Via: SIP/2.0/UDP 10.2.110.195:5060;branch=z9hG4bK4fe9cdf7;rport From: "foo" ;tag=as2c33c35f To: Contact: Call-ID: 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195 CSeq: 102 INVITE User-Agent: 3000 Max-Forwards: 70 Date: Tue, 13 Feb 2007 17:00:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 269 v=0 o=root 1845 1845 IN IP4 10.2.119.177 s=session c=IN IP4 10.2.119.177 t=0 0 m=audio 11388 RTP/AVP 8 111 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 13 lines) --- Sending to 10.2.110.195 : 5060 (NAT) Using INVITE request as basis request - 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195 Found user '86' Found RTP audio format 8 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 10.2.119.177:11388 Found description format PCMA for ID 8 Found description format G726-32 for ID 111 Found description format telephone-event for ID 101 Capabilities: us - 0x80c0e (gsm|ulaw|alaw|g726|ilbc|h263), peer - audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.119.177:11388 Looking for 1234 in default (domain 192.168.9.1) list_route: hop: <--- Transmitting (no NAT) to 10.2.110.195:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.110.195:5060;branch=z9hG4bK4fe9cdf7;received=10.2.110.195;rport=5060 From: "foo" ;tag=as2c33c35f To: Call-ID: 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [1234@default:1] DumpChan("SIP/86-081b3ee0", "") in new stack Dumping Info For Channel: SIP/86-081b3ee0: ================================================================================ Info: Name= SIP/86-081b3ee0 Type= SIP UniqueID= 1171386124.7 CallerID= 86 CallerIDName= foo DNIDDigits= 1234 RDNIS= (N/A) State= Ring (4) Rings= 0 NativeFormat= 0x800 (g726) WriteFormat= 0x800 (g726) ReadFormat= 0x800 (g726) 1stFileDescriptor= 20 Framesin= 0 Framesout= 0 TimetoHangup= 0 ElapsedTime= 0h0m0s Context= default Extension= 1234 Priority= 1 CallGroup= PickupGroup= Application= DumpChan Data= (Empty) Blocking_in= (Not Blocking) Variables: SIPCALLID=346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195 SIPUSERAGENT=Mediatrix 3000 SIPDOMAIN=192.168.9.1 SIPURI=sip:86@10.2.110.195 ================================================================================ -- Executing [1234@default:2] MeetMe("SIP/86-081b3ee0", "1234") in new stack [Feb 13 12:02:04] WARNING[3603]: app_meetme.c:2181 conf_exec: cant transcode chan(SIP/86-081b3ee0) since his nativeformats is 0x800 (g726) (2048) == Spawn extension (default, 1234, 2) exited non-zero on 'SIP/86-081b3ee0' Scheduling destruction of SIP dialog '346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 10.2.110.195:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 10.2.110.195:5060;branch=z9hG4bK4fe9cdf7;received=10.2.110.195;rport=5060 From: "foo" ;tag=as2c33c35f To: ;tag=as00440db7 Call-ID: 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> mayamini-lx*CLI> <--- SIP read from 10.2.110.195:5060 ---> ACK sip:1234@192.168.9.1 SIP/2.0 Via: SIP/2.0/UDP 10.2.110.195:5060;branch=z9hG4bK4fe9cdf7;rport From: "foo" ;tag=as2c33c35f To: ;tag=as00440db7 Contact: Call-ID: 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195 CSeq: 102 ACK User-Agent: 3000 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- mayamini-lx*CLI>