rendered paste bodyprxyafr*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" <sip:Unknown@190.24.227.62>;tag=as4e0e2d11
To: <sip:106@200.93.220.7:5060;user=phone;transport=udp>
Contact: <sip:Unknown@190.24.227.62>
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" <sip:Unknown@190.24.227.62>;tag=as4e0e2d11
To: <sip:106@200.93.220.7:5060;user=phone;transport=udp>;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 <sip:106@190.24.227.62;user=phone>;tag=2481365825
To: <sip:101@190.24.227.62;user=phone>
Call-ID: 3442052975@200.93.220.7
CSeq: 1 INVITE
Contact: Guillermo Salas M <sip:106@200.93.220.7:5060;user=phone;transport=udp>
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 <sip:106@190.24.227.62;user=phone>;tag=2481365825
To: <sip:101@190.24.227.62;user=phone>;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 <sip:106@190.24.227.62;user=phone>;tag=2481365825
To: <sip:101@190.24.227.62;user=phone>;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 <sip:106@190.24.227.62;user=phone>;tag=2481365825
To: <sip:101@190.24.227.62;user=phone>
Call-ID: 3442052975@200.93.220.7
CSeq: 2 INVITE
Contact: Guillermo Salas M <sip:106@200.93.220.7:5060;user=phone;transport=udp>
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: <sip:106@200.93.220.7:5060;user=phone;transport=udp>
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 <sip:106@190.24.227.62;user=phone>;tag=2481365825
To: <sip:101@190.24.227.62;user=phone>
Call-ID: 3442052975@200.93.220.7
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:101@190.24.227.62>
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 <sip:106@190.24.227.62;user=phone>;tag=2481365825
To: <sip:101@190.24.227.62;user=phone>;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: <sip:101@190.24.227.62>
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 <sip:106@190.24.227.62;user=phone>;tag=2481365825
To: <sip:101@190.24.227.62;user=phone>;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 <sip:106@190.24.227.62;user=phone>;tag=2481365825
To: <sip:101@190.24.227.62;user=phone>;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 <sip:106@190.24.227.62;user=phone>;tag=2481365825
To: <sip:101@190.24.227.62;user=phone>;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: <sip:101@190.24.227.62>
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'