prxyafr*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 115 (Unspecified) D N 0 UNKNOWN 114 (Unspecified) D N 0 UNKNOWN 113 (Unspecified) D N 0 UNKNOWN 112 (Unspecified) D N 0 UNKNOWN 111 (Unspecified) D N 0 UNKNOWN 110 (Unspecified) D 0 UNKNOWN 109 (Unspecified) D N 0 UNKNOWN 108 (Unspecified) D N 0 UNKNOWN 107 (Unspecified) D N 0 UNKNOWN 106/106 200.93.220.7 D N 5060 OK (280 ms) 105/105 200.58.211.225 D N 60928 OK (120 ms) 104/104 (Unspecified) D N 0 UNKNOWN 103/103 (Unspecified) D N 0 UNKNOWN 102 (Unspecified) D N 0 UNKNOWN 101/101 192.168.1.70 D N 5061 OK (17 ms) 100/100 192.168.1.70 D N 5060 OK (16 ms) 16 sip peers [4 online , 12 offline] When try to call to ext 101 I get a voice regarding the extension is not avaliable: prxyafr*CLI> sip debug peer 106 SIP Debugging Enabled for IP: 200.93.220.7:5060 12 headers, 0 lines Reliably Transmitting (NAT) to 200.93.220.7:5060: OPTIONS sip:106@200.93.220.7:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 190.24.227.62:5060;branch=z9hG4bK71d9c91a;rport From: "Unknown" ;tag=as4e0e2d11 To: Contact: Call-ID: 0eaa9aaf040d3de27cb8ee0f4f4d6895@190.24.227.62 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 14:07:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- prxyafr*CLI> <-- SIP read from 200.93.220.7:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 190.24.227.62:5060;branch=z9hG4bK71d9c91a;rport From: "Unknown" ;tag=as4e0e2d11 To: ;tag=2481365825 Call-ID: 0eaa9aaf040d3de27cb8ee0f4f4d6895@190.24.227.62 CSeq: 102 OPTIONS Server: Cisco-CP7905/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATESupported: replaces Content-Length: 275 Content-Type: application/sdp v=0 o=106 4062 4062 IN IP4 200.93.220.7 s=Cisco 7905 SIP Call c=IN IP4 200.93.220.7 t=0 0 m=audio 16384 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (11 headers 12 lines) --- Destroying call '0eaa9aaf040d3de27cb8ee0f4f4d6895@190.24.227.62' prxyafr*CLI> <-- SIP read from 200.93.220.7:5060: INVITE sip:101@190.24.227.62;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.93.220.7:5060;branch=z9hG4bK7f876c7f180d03e5 From: Guillermo Salas M ;tag=2481365825 To: Call-ID: 3442052975@200.93.220.7 CSeq: 1 INVITE Contact: Guillermo Salas M User-Agent: Cisco-CP7905/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATESupported: replaces, 100rel Expires: 300 Content-Length: 275 Content-Type: application/sdp v=0 o=106 4688 4688 IN IP4 200.93.220.7 s=Cisco 7905 SIP Call c=IN IP4 200.93.220.7 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (13 headers 12 lines) --- Using INVITE request as basis request - 3442052975@200.93.220.7 Sending to 200.93.220.7 : 5060 (NAT) Reliably Transmitting (NAT) to 200.93.220.7:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 200.93.220.7:5060;branch=z9hG4bK7f876c7f180d03e5;received=200.93.220.7 From: Guillermo Salas M ;tag=2481365825 To: ;tag=as4f18d332 Call-ID: 3442052975@200.93.220.7 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25be3e5d" Content-Length: 0 --- Scheduling destruction of call '3442052975@200.93.220.7' in 15000 ms Found user '106' prxyafr*CLI> <-- SIP read from 200.93.220.7:5060: ACK sip:101@190.24.227.62;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.93.220.7:5060;branch=z9hG4bK7f876c7f180d03e5 From: Guillermo Salas M ;tag=2481365825 To: ;tag=as4f18d332 Call-ID: 3442052975@200.93.220.7 CSeq: 1 ACK User-Agent: Cisco-CP7905/8.0.0-060111A Content-Length: 0 --- (8 headers 0 lines) --- prxyafr*CLI> <-- SIP read from 200.93.220.7:5060: INVITE sip:101@190.24.227.62;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.93.220.7:5060;branch=z9hG4bK2408a4215f66df6 From: Guillermo Salas M ;tag=2481365825 To: Call-ID: 3442052975@200.93.220.7 CSeq: 2 INVITE Contact: Guillermo Salas M User-Agent: Cisco-CP7905/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATESupported: replaces, 100rel Proxy-Authorization: Digest username="106",realm="asterisk",nonce="25be3e5d",uri="sip:101@190.24.227.62",response="c5362180149a0c19f954fb43a7126483" Expires: 300 Content-Length: 275 Content-Type: application/sdp v=0 o=106 4717 4717 IN IP4 200.93.220.7 s=Cisco 7905 SIP Call c=IN IP4 200.93.220.7 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (14 headers 12 lines) --- Using INVITE request as basis request - 3442052975@200.93.220.7 Sending to 200.93.220.7 : 5060 (NAT) Found user '106' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 200.93.220.7:16384 Found description format G729 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0x50f (g723|gsm|ulaw|alaw|g729|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 101 in from-internal (domain 190.24.227.62;user=phone) list_route: hop: Transmitting (NAT) to 200.93.220.7:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.93.220.7:5060;branch=z9hG4bK2408a4215f66df6;received=200.93.220.7 From: Guillermo Salas M ;tag=2481365825 To: Call-ID: 3442052975@200.93.220.7 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/106-0816a430", "exten-vm|101|101") in new stack -- Executing Macro("SIP/106-0816a430", "user-callerid") in new stack -- Executing NoOp("SIP/106-0816a430", "user-callerid: device 106") in new stack -- Executing GotoIf("SIP/106-0816a430", "0?report") in new stack -- Executing GotoIf("SIP/106-0816a430", "0?start") in new stack -- Executing Set("SIP/106-0816a430", "REALCALLERIDNUM=106") in new stack -- Executing NoOp("SIP/106-0816a430", "REALCALLERIDNUM is 106") in new stack -- Executing Set("SIP/106-0816a430", "AMPUSER=106") in new stack -- Executing Set("SIP/106-0816a430", "AMPUSERCIDNAME=Extension 106") in new stack -- Executing GotoIf("SIP/106-0816a430", "0?report") in new stack -- Executing Set("SIP/106-0816a430", "CALLERID(all)=Extension 106 <106>") in new stack -- Executing Set("SIP/106-0816a430", "REALCALLERIDNUM=106") in new stack -- Executing NoOp("SIP/106-0816a430", "TTL: ARG1: 101") in new stack -- Executing GotoIf("SIP/106-0816a430", "0?continue") in new stack -- Executing Set("SIP/106-0816a430", "_TTL=64") in new stack -- Executing GotoIf("SIP/106-0816a430", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/106-0816a430", "Using CallerID "Extension 106" <106>") in new stack -- Executing Set("SIP/106-0816a430", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/106-0816a430", "VMBOX=101") in new stack -- Executing Set("SIP/106-0816a430", "EXTTOCALL=101") in new stack -- Executing Set("SIP/106-0816a430", "CFUEXT=") in new stack -- Executing Set("SIP/106-0816a430", "RT=25") in new stack -- Executing Macro("SIP/106-0816a430", "record-enable|101|IN") in new stack -- Executing GotoIf("SIP/106-0816a430", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/106-0816a430", "recordingcheck|20070208-090729|1170943649.15") in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/106-0816a430", "No recording needed") in new stack -- Executing Macro("SIP/106-0816a430", "dial|25|trTw|101") in new stack -- Executing DeadAGI("SIP/106-0816a430", "dialparties.agi") in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/dialparties.agi -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp("SIP/106-0816a430", "Returned from dialparties with no extensions to call") in new stack -- Executing NoOp("SIP/106-0816a430", "DIALSTATUS is ") in new stack -- Executing GosubIf("SIP/106-0816a430", "0?docfu|1") in new stack -- Executing NoOp("SIP/106-0816a430", "Voicemail is 101") in new stack -- Executing GotoIf("SIP/106-0816a430", "0?s-|1") in new stack -- Executing NoOp("SIP/106-0816a430", "Sending to Voicemail box 101") in new stack -- Executing Macro("SIP/106-0816a430", "vm|101|") in new stack -- Executing Macro("SIP/106-0816a430", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/106-0816a430", "user-callerid: Extension 106 106") in new stack -- Executing GotoIf("SIP/106-0816a430", "0?report") in new stack -- Executing GotoIf("SIP/106-0816a430", "1?start") in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp("SIP/106-0816a430", "REALCALLERIDNUM is 106") in new stack -- Executing Set("SIP/106-0816a430", "AMPUSER=106") in new stack -- Executing Set("SIP/106-0816a430", "AMPUSERCIDNAME=Extension 106") in new stack -- Executing GotoIf("SIP/106-0816a430", "0?report") in new stack -- Executing Set("SIP/106-0816a430", "CALLERID(all)=Extension 106 <106>") in new stack -- Executing Set("SIP/106-0816a430", "REALCALLERIDNUM=106") in new stack -- Executing NoOp("SIP/106-0816a430", "TTL: 64 ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/106-0816a430", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/106-0816a430", "Using CallerID "Extension 106" <106>") in new stack -- Executing Set("SIP/106-0816a430", "VMGAIN=") in new stack -- Executing GotoIf("SIP/106-0816a430", "1?s-|1") in new stack -- Goto (macro-vm,s-,1) -- Executing Macro("SIP/106-0816a430", "get-vmcontext|101") in new stack -- Executing Set("SIP/106-0816a430", "VMCONTEXT=default") in new stack -- Executing GotoIf("SIP/106-0816a430", "0?200:300") in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing NoOp("SIP/106-0816a430", "") in new stack -- Executing VoiceMail("SIP/106-0816a430", "101@default|u") in new stack We're at 190.24.227.62 port 10896 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 200.93.220.7:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.93.220.7:5060;branch=z9hG4bK2408a4215f66df6;received=200.93.220.7 From: Guillermo Salas M ;tag=2481365825 To: ;tag=as697c9b3f Call-ID: 3442052975@200.93.220.7 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 2108 2108 IN IP4 190.24.227.62 s=session c=IN IP4 190.24.227.62 t=0 0 m=audio 10896 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'vm-theperson' (language 'es') prxyafr*CLI> <-- SIP read from 200.93.220.7:5060: ACK sip:101@190.24.227.62 SIP/2.0 Via: SIP/2.0/UDP 200.93.220.7:5060;branch=z9hG4bK906c28384f2dc1d From: Guillermo Salas M ;tag=2481365825 To: ;tag=as697c9b3f Call-ID: 3442052975@200.93.220.7 CSeq: 2 ACK User-Agent: Cisco-CP7905/8.0.0-060111A Proxy-Authorization: Digest username="106",realm="asterisk",nonce="25be3e5d",uri="sip:101@190.24.227.62",response="c5362180149a0c19f954fb43a7126483" Content-Length: 0 --- (9 headers 0 lines) --- -- Playing 'digits/1' (language 'es') prxyafr*CLI> <-- SIP read from 200.93.220.7:5060: BYE sip:101@190.24.227.62 SIP/2.0 Via: SIP/2.0/UDP 200.93.220.7:5060;branch=z9hG4bK738af25110e27d33 From: Guillermo Salas M ;tag=2481365825 To: ;tag=as697c9b3f Call-ID: 3442052975@200.93.220.7 CSeq: 3 BYE User-Agent: Cisco-CP7905/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATESupported: replaces, 100rel Proxy-Authorization: Digest username="106",realm="asterisk",nonce="25be3e5d",uri="sip:101@190.24.227.62",response="cfa909e6d531d7650d6a400dd7cb8f42" Content-Length: 0 --- (11 headers 0 lines) --- Sending to 200.93.220.7 : 5060 (NAT) Transmitting (NAT) to 200.93.220.7:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.93.220.7:5060;branch=z9hG4bK738af25110e27d33;received=200.93.220.7 From: Guillermo Salas M ;tag=2481365825 To: ;tag=as697c9b3f Call-ID: 3442052975@200.93.220.7 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- == Spawn extension (macro-vm, s-, 2) exited non-zero on 'SIP/106-0816a430' in macro 'vm' == Spawn extension (macro-vm, s-, 2) exited non-zero on 'SIP/106-0816a430' in macro 'exten-vm' == Spawn extension (macro-vm, s-, 2) exited non-zero on 'SIP/106-0816a430' Destroying call '3442052975@200.93.220.7'