[root@pbx1 ~]# asterisk -r Asterisk SVN-branch-1.2-r24911, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk SVN-branch-1.2-r24911 currently running on pbx1 (pid = 2626) ion1*CLI> Verbosity is at least 3 pbx1*CLI> show dialplan *99@features [ Context 'features' created by 'pbx_config' ] '*99' => 1. Answer() [pbx_config] 2. Wait(0.5) [pbx_config] 3. NoOp( ${SIPCHANINFO(peername)} connecting from ${SIPCHANINFO(peerip)} or ${CALLERID(name)}) [pbx_config] 4. Set(PEER=${CHANNEL:4:9}) [pbx_config] 5. NoOp( ${PEER}) [pbx_config] 6. GotoIF($[${SIPPEER(${PEER}:mailbox)} > ""]?VM:NOVM) [pbx_config] [VM] 7. Set(DB(users/${USERID}/VM)=1) [pbx_config] 8. Playback(self-destruct) [pbx_config] 9. Hangup() [pbx_config] [NOVM] 10. Playback(error) [pbx_config] 11. Hangup() [pbx_config] -= 1 extension (11 priorities) in 1 context. =- pbx1*CLI> sip debug SIP Debugging enabled pbx1*CLI> <-- SIP read from 192.168.1.10:5060: INVITE sip:*99@192.168.0.231:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK98673fb0C9F3B58D From: "John Studebaker" ;tag=FE3C41EA-92AB2869 To: CSeq: 1 INVITE Call-ID: c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 213 v=0 o=- 1148571169 1148571169 IN IP4 192.168.1.10 s=Polycom IP Phone c=IN IP4 192.168.1.10 t=0 0 m=audio 2236 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 9 lines)--- Using INVITE request as basis request - c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 Sending to 192.168.1.10 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.1.10:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK98673fb0C9F3B58D;received=192.168.1.10 From: "John Studebaker" ;tag=FE3C41EA-92AB2869 To: ;tag=as084edfb2 Call-ID: c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="617adf28" Content-Length: 0 --- Scheduling destruction of call 'c6b0369e-da9574dc-b0eb0eb3@192.168.1.10' in 15000 ms Found user 'p501_eed5' pbx1*CLI> <-- SIP read from 192.168.1.10:5060: ACK sip:*99@192.168.0.231:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK98673fb0C9F3B58D From: "John Studebaker" ;tag=FE3C41EA-92AB2869 To: ;tag=as084edfb2 CSeq: 1 ACK Call-ID: c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- pbx1*CLI> <-- SIP read from 192.168.1.10:5060: INVITE sip:*99@192.168.0.231:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bKd501082fFFBAD948 From: "John Studebaker" ;tag=FE3C41EA-92AB2869 To: CSeq: 2 INVITE Call-ID: c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="p501_eed5", realm="asterisk", nonce="617adf28", uri="sip:*99@192.168.0.231:5060", response="4b9cf6e4c0cfae78e8e45a4e6ccf2c30", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 213 v=0 o=- 1148571169 1148571169 IN IP4 192.168.1.10 s=Polycom IP Phone c=IN IP4 192.168.1.10 t=0 0 m=audio 2236 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 9 lines)--- Using INVITE request as basis request - c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 Sending to 192.168.1.10 : 5060 (non-NAT) Found user 'p501_eed5' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.10:2236 Found description format G729 Found description format PCMU Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for *99 in longdistance (domain 192.168.0.231) list_route: hop: Transmitting (no NAT) to 192.168.1.10:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bKd501082fFFBAD948;received=192.168.1.10 From: "John Studebaker" ;tag=FE3C41EA-92AB2869 To: Call-ID: c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Answer("SIP/p501_eed5-06c3", "") in new stack We're at 192.168.0.231 port 19478 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bKd501082fFFBAD948;received=192.168.1.10 From: "John Studebaker" ;tag=FE3C41EA-92AB2869 To: ;tag=as24caa2e2 Call-ID: c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 2626 2626 IN IP4 192.168.0.231 s=session c=IN IP4 192.168.0.231 t=0 0 m=audio 19478 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait("SIP/p501_eed5-06c3", "0.5") in new stack pbx1*CLI> <-- SIP read from 192.168.1.10:5060: ACK sip:*99@192.168.0.231 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3e9f58f41DEB3EE1 From: "John Studebaker" ;tag=FE3C41EA-92AB2869 To: ;tag=as24caa2e2 CSeq: 2 ACK Call-ID: c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- -- Executing NoOp("SIP/p501_eed5-06c3", " connecting from 192.168.1.10 or Bruce Reeves") in new stack -- Executing Set("SIP/p501_eed5-06c3", "PEER=p501_eed5") in new stack -- Executing NoOp("SIP/p501_eed5-06c3", " p501_eed5") in new stack -- Executing GotoIf("SIP/p501_eed5-06c3", "1?VM:NOVM") in new stack -- Goto (longdistance,*99,7) -- Executing Set("SIP/p501_eed5-06c3", "DB(users/breeves/VM)=1") in new stack -- Executing Playback("SIP/p501_eed5-06c3", "self-destruct") in new stack -- Playing 'self-destruct' (language 'en') -- Executing Hangup("SIP/p501_eed5-06c3", "") in new stack == Spawn extension (longdistance, *99, 9) exited non-zero on 'SIP/p501_eed5-06c3' set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.10, port 5060 Reliably Transmitting (no NAT) to 192.168.1.10:5060: BYE sip:p501_eed5@192.168.1.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK686d6ff1 From: ;tag=as24caa2e2 To: "John Studebaker" ;tag=FE3C41EA-92AB2869 Contact: Call-ID: c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx1*CLI> <-- SIP read from 192.168.1.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK686d6ff1 From: ;tag=as24caa2e2 To: "John Studebaker" ;tag=FE3C41EA-92AB2869 CSeq: 102 BYE Call-ID: c6b0369e-da9574dc-b0eb0eb3@192.168.1.10 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067 Content-Length: 0 --- (9 headers 0 lines)--- Destroying call 'c6b0369e-da9574dc-b0eb0eb3@192.168.1.10' pbx1*CLI> sip no debug SIP Debugging Disabled pbx1*CLI>