*CLI> sip debug ip xxx.yyy.89.135 SIP Debugging Enabled for IP: xxx.yyy.89.135 The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. *CLI> <--- SIP read from xxx.yyy.89.135:1024 ---> INVITE sip:5192922001@xxx.yyy.89.134 SIP/2.0 Via: SIP/2.0/UDP xxx.yyy.89.135:5060;branch=z9hG4bK632edb90;rport Max-Forwards: 70 From: "MR ME" ;tag=as7e9d839a To: Contact: Call-ID: 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r67164 Date: Fri, 22 Jun 2007 00:10:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 251 v=0 o=root 1577940051 1577940051 IN IP4 xxx.yyy.89.135 s=session c=IN IP4 xxx.yyy.89.135 t=0 0 m=audio 11670 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- Sending to xxx.yyy.89.135 : 1024 (NAT) Using INVITE request as basis request - 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135 Found no matching peer or user for 'xxx.yyy.89.135:1024' Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port xxx.yyy.89.135:11670 Found description format GSM for ID 3 Found description format telephone-event for ID 101 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port xxx.yyy.89.135:11670 Looking for 5192922001 in default (domain xxx.yyy.89.134) list_route: hop: <--- Transmitting (NAT) to xxx.yyy.89.135:1024 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.yyy.89.135:5060;branch=z9hG4bK632edb90;received=xxx.yyy.89.135;rport=1024 From: "MR ME" ;tag=as7e9d839a To: Call-ID: 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [5192922001@default:1] NoOp("SIP/xxx.yyy.89.135-082be860", "default context hit "5192922001"| this is wrong!") in new stack -- Executing [5192922001@default:2] Hangup("SIP/xxx.yyy.89.135-082be860", "") in new stack == Spawn extension (default, 5192922001, 2) exited non-zero on 'SIP/xxx.yyy.89.135-082be860' -- Executing [h@default:1] NoOp("SIP/xxx.yyy.89.135-082be860", "default context hit h exten| this is wrong!") in new stack Scheduling destruction of SIP dialog '50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to xxx.yyy.89.135:1024 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP xxx.yyy.89.135:5060;branch=z9hG4bK632edb90;received=xxx.yyy.89.135;rport=1024 From: "MR ME" ;tag=as7e9d839a To: ;tag=as04a4fedb Call-ID: 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from xxx.yyy.89.135:1024 ---> ACK sip:5192922001@xxx.yyy.89.134 SIP/2.0 Via: SIP/2.0/UDP xxx.yyy.89.135:5060;branch=z9hG4bK632edb90;rport Max-Forwards: 70 From: "MR ME" ;tag=as7e9d839a To: ;tag=as04a4fedb Contact: Call-ID: 50332f3d496972bd1434da9c0d99b7a3@xxx.yyy.89.135 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r67164 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- sip no denbugReliably Transmitting (no NAT) to xxx.yyy.89.135:5060: OPTIONS sip:xxx.yyy.89.135 SIP/2.0 Via: SIP/2.0/UDP xxx.yyy.89.134:5060;branch=z9hG4bK7936ee6e;rport From: "asterisk" ;tag=as166183ee To: Contact: Call-ID: 5360daef0022147d1a3008e660f78df7@xxx.yyy.89.134 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Jun 2007 00:04:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from xxx.yyy.89.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.yyy.89.134:5060;branch=z9hG4bK7936ee6e;rport From: "asterisk" ;tag=as166183ee To: ;tag=B77C89E0-228A4B1F CSeq: 102 OPTIONS Call-ID: 5360daef0022147d1a3008e660f78df7@xxx.yyy.89.134 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '5360daef0022147d1a3008e660f78df7@xxx.yyy.89.134' Method: OPTIONS Reliably Transmitting (NAT) to xxx.yyy.89.135:5060: OPTIONS sip:843@192.168.77.146 SIP/2.0 Via: SIP/2.0/UDP xxx.yyy.89.134:5060;branch=z9hG4bK199022ab;rport From: "asterisk" ;tag=as756087c7 To: Contact: Call-ID: 42a8d8b820b459b153fb4f4e72fcbc08@xxx.yyy.89.134 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Jun 2007 00:04:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from xxx.yyy.89.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.yyy.89.134:5060;branch=z9hG4bK199022ab;rport From: "asterisk" ;tag=as756087c7 To: ;tag=2FA17F4B-A0805606 CSeq: 102 OPTIONS Call-ID: 42a8d8b820b459b153fb4f4e72fcbc08@xxx.yyy.89.134 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '42a8d8b820b459b153fb4f4e72fcbc08@xxx.yyy.89.134' Method: OPTIONS sip no debug SIP Debugging Disabled The 'sip no debug' command is deprecated and will be removed in a future release. Please use 'sip set debug off' instead. *CLI> Machine A (the one placing the call) [machineB] type=peer host=xxx.yyy.89.134 deny=0.0.0.0/0 permit=xxx.yyy.89.134/32 dtmfmode=rfc2833 context=incoming disallow=all allow=gsm nat=no canreinvite=no qualify=1000 Machine B (the one receiving and putting the call in [default] instead of where it should be: [machineA] type=peer host=xxx.yyy.89.135 deny=0.0.0.0/0 permit=xxx.yyy.89.135/32 context=from_andrew qualify=500 qualifysmoothing=yes disallow=all allow=gsm dtmfmode=rfc2833 canreinvite=no nat=no