Ok,
So I have a cisco 7960 behind an ADSL rounter and Asterisk 13.7.2 on a public ip address on the internet.
180.216.8.69 is the IP of the router
In Asterisk, extension shows active as:
0100/0100 180.216.8.79 D
No No A 5061 OK (220 ms)
Cisco 7960 uses extension 0100
0100 - nat=no
sip show peer 0100
Name : 0100
Description :
Secret :
MD5Secret :
Remote Secret:
Context : from-internal
Record On feature : automon
Record Off feature : automon
Subscr.Cont. :
Language :
Tonezone :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : “Pablo Cisco” <0100>
MaxCallBR : 384 kbps
Expire : 3191
Insecure : no
Force rport : No
Symmetric RTP: No
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : No
Path support : No
Path : N/A
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 180.216.8.79:5061
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 0100
SIP Options : (none)
Codecs : (g729|g723|g726|ilbc|gsm|speex|alaw|g722|opus|adpcm|slin|lpc10|
Auto-Framing : No
Status : OK (193 ms)
Useragent : Cisco-CP7960G/8.0
Reg. Contact : sip:[email protected]:5061;transport=udp
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
**Settings on 7960 **
nat_enabled=1
nat_address=""
nat_received_processing=1
voip_control_port=5061
When receiving INVITE options the Cisco 7960 seems to not be able to
match the line as the INVITE contains the IP address of the firewall and
returns a 404 not found
[04:06:27:103525] SIPProcessUDPMessage: recv UDP message from
<163.53.228.113>:<50195>, length=<1603>, message=
[04:06:27:103526] INVITE sip:[email protected]:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
Max-Forwards: 70
From: “Pablo Android” ;tag=as6ca2516a
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76.2(13.6.0)
Date: Wed, 17 Feb 2016 10:49:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 986
v=0
o=root 649373155 649373155 IN IP4 163.53.228.113
s=Asterisk PBX 13.6.0
c=IN IP4 163.53.228.113
b=CT:384
t=0 0
m=audio 11406 RTP/AVP 97 18 4 111 3 110 8 9 107 5 10 7 117 112 118 102 115 116 119 0 101
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:117 speex/16000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:118 L16/16000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:119 speex/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv
m=video 16040 RTP/AVP 99 104
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=sendrecv
[04:06:27:103532] Unknown address in Request URI
[04:06:27:103533] sipSPICheckRequest: Request URI Not Found
[04:06:27:103533] SIPTaskProcessSIPMessage: Error: sipSPICheckRequest() returned error.
[04:06:27:103535] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=INVITE, to_tag=<>)
[04:06:27:103537]
sipTransportSendMessage: Opened a one-time UDP send channel to server
<163.53.228.113>:<5060>, handle = 8 local port= 5061
[04:06:27:103538]
sipTransportSendMessage:Sent SIP message to
<163.53.228.113>:<5060>, handle=<8>,
length=<298>, message=
[04:06:27:103538] SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
From: “Pablo Android” ;tag=as6ca2516a
To:
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Content-Length: 0
[04:06:27:103539] sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
[04:06:27:103545]
SIPProcessUDPMessage: recv UDP message from
<163.53.228.113>:<50195>, length=<1603>, message=
[04:06:27:103546] INVITE sip:[email protected]:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
Max-Forwards: 70
From: “Pablo Android” ;tag=as6ca2516a
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76.2(13.6.0)
Date: Wed, 17 Feb 2016 10:49:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 986
v=0
o=root 649373155 649373155 IN IP4 163.53.228.113
s=Asterisk PBX 13.6.0
c=IN IP4 163.53.228.113
b=CT:384
t=0 0
m=audio 11406 RTP/AVP 97 18 4 111 3 110 8 9 107 5 10 7 117 112 118 102 115 116 119 0 101
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:117 speex/16000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:118 L16/16000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:119 speex/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv
m=video 16040 RTP/AVP 99 104
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=sendrecv
[04:06:27:103552] Unknown address in Request URI
[04:06:27:103553] sipSPICheckRequest: Request URI Not Found
[04:06:27:103553] SIPTaskProcessSIPMessage: Error: sipSPICheckRequest() returned error.
[04:06:27:103555] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=INVITE, to_tag=<>)
[04:06:27:103557]
sipTransportSendMessage: Opened a one-time UDP send channel to server
<163.53.228.113>:<5060>, handle = 8 local port= 5061
[04:06:27:103558]
sipTransportSendMessage:Sent SIP message to
<163.53.228.113>:<5060>, handle=<8>,
length=<298>, message=
[04:06:27:103558] SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
From: “Pablo Android” ;tag=as6ca2516a
To:
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Content-Length: 0
[04:06:27:103559] sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
[04:06:27:103560] SIPProcessUDPMessage: recv UDP message from <163.53.228.113>:<50195>, length=<417>, message=
[04:06:27:103561] ACK sip:[email protected]:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
Max-Forwards: 70
From: “Pablo Android” ;tag=as6ca2516a
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.2(13.6.0)
Content-Length: 0
[04:06:27:103566] SIPProcessUDPMessage: recv UDP message from <163.53.228.113>:<50195>, length=<417>, message=
[04:06:27:103566] ACK sip:[email protected]:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
Max-Forwards: 70
From: “Pablo Android” ;tag=as6ca2516a
To:
Contact:
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.2(13.6.0)
Content-Length: 0
[04:06:40:104843] SIPProcessUDPMessage: recv UDP message from <163.53.228.113>:<50195>, length=<582>, message=
[04:06:40:104844]
OPTIONS sip:[email protected]:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP
163.53.228.113:5060;branch=z9hG4bK43ad6428
Max-Forwards: 70
From: “Unknown” ;tag=as29d04985
To:
Contact:
Call-ID:
[email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.2(13.6.0)
Date: Wed, 17 Feb 2016 10:49:31 GMT
Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported:
replaces, timer
Content-Length: 0
[04:06:40:104849] sippmh_parse_supported_require: Invalid tag in
Require/Supported timer
[04:06:40:104854]
sipRelDevCoupledMessageStore: Storing for reTx (cseq=102,
method=OPTIONS, to_tag=<00131a6ed26c00184e2c6fe8-20662660>)
[04:06:40:104856]
sipTransportSendMessage: Opened a one-time UDP send channel to server
<163.53.228.113>:<5060>, handle = 8 local port= 5061
[04:06:40:104857]
sipTransportSendMessage:Sent SIP message to
<163.53.228.113>:<5060>, handle=<8>,
length=<883>, message=
[04:06:40:104858] SIP/2.0 200 OK
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK43ad6428
From: “Unknown” ;tag=as29d04985
To: ;tag=00131a6ed26c00184e2c6fe8-20662660
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: Cisco-CP7960G/8.0
Allow: ACK,BYE,CANCE