pbx*CLI> dial 13806 -- Executing NoOp("OSS/dsp", "call_agi") in new stack -- Executing Set("OSS/dsp", "__ORIG_CHAN=OSS/dsp") in new stack -- Executing AGI("OSS/dsp", "agi://localhost:5555/") in new stack -- AGI Script Executing Application: (DIAL) Options: (SIP/21.0.20.11/5004||ht) ------------------------------------- Total wait time 36 seconds before anything else occurs ------------------------------------- Mar 13 12:46:06 DEBUG[3671]: chan_sip.c:2085 sip_call: Outgoing Call for 5004 We're at 21.0.20.10 port 10030 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 21.0.20.11:5060: INVITE sip:5004@21.0.20.11 SIP/2.0 Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK78858200;rport From: "asterisk" ;tag=as1e6f9d84 To: Contact: Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 13 Mar 2008 17:46:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 210 v=0 o=root 5435 5435 IN IP4 21.0.20.10 s=session c=IN IP4 21.0.20.10 t=0 0 m=audio 10030 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 21.0.20.11/5004 pbx*CLI> <-- SIP read from 21.0.20.11:1243: SIP/2.0 100 Trying To: ;tag=29655 From: "asterisk" ;tag=as1e6f9d84 Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK78858200;rport CSeq: 102 INVITE Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10 Server: ININ-CUMAICSRVR-36357178 Content-Length: 0 --- (8 headers 0 lines) --- Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1e1adfae607f8881463e5aa461b1d33e@21.0.20.10' Request 102: Found pbx*CLI> <-- SIP read from 21.0.20.11:1243: SIP/2.0 180 Ringing To: ;tag=29655 From: "asterisk" ;tag=as1e6f9d84 Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK78858200;rport CSeq: 102 INVITE Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10 Server: ININ-CUMAICSRVR-36357178 Content-Length: 0 --- (8 headers 0 lines) --- Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1e1adfae607f8881463e5aa461b1d33e@21.0.20.10' Request 102: Found -- SIP/21.0.20.11-081d6678 is ringing Mar 13 12:46:06 WARNING[6069]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory pbx*CLI> <-- SIP read from 21.0.20.11:1243: SIP/2.0 200 OK To: ;tag=29655 From: "asterisk" ;tag=as1e6f9d84 Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK78858200;rport CSeq: 102 INVITE Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10 Contact: Content-Type: application/sdp Server: ININ-CUMAICSRVR-36357178 Content-Length: 194 v=0 o=ININ 44362390 44362390 IN IP4 21.0.20.11 s=Interaction c=IN IP4 21.0.20.12 t=0 0 m=audio 4770 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (10 headers 9 lines) --- Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '1e1adfae607f8881463e5aa461b1d33e@21.0.20.10' of Request 102: Match Found Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 21.0.20.12:4770 Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:3689 process_sdp: Peer audio RTP is at port 21.0.20.12:4770 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:6282 build_route: build_route: Contact hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 21.0.20.11, port 5060 Transmitting (no NAT) to 21.0.20.11:5060: ACK sip:5004@21.0.20.11 SIP/2.0 Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK2c09f15b;rport From: "asterisk" ;tag=as1e6f9d84 To: ;tag=29655 Contact: Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/21.0.20.11-081d6678 answered OSS/dsp << Console call has been answered >>