Cisco 7960 behind NAT - No incoming calls, 7960 returning 404

Ok,

So I have a cisco 7960 behind an ADSL rounter and Asterisk 13.6 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:0100@180.216.8.79:5061;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:0100@180.216.8.79:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
Max-Forwards: 70
From: “Pablo Android” sip:0103@163.53.228.113;tag=as6ca2516a
To: sip:0100@180.216.8.79:5061;transport=udp
Contact: sip:0103@163.53.228.113:5060
Call-ID: 3a6fb38413883f3a05fed4c53e2f04c4@163.53.228.113:5060
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” sip:0103@163.53.228.113;tag=as6ca2516a
To: sip:0100@180.216.8.79:5061;transport=udp
Call-ID: 3a6fb38413883f3a05fed4c53e2f04c4@163.53.228.113:5060
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:0100@180.216.8.79:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
Max-Forwards: 70
From: “Pablo Android” sip:0103@163.53.228.113;tag=as6ca2516a
To: sip:0100@180.216.8.79:5061;transport=udp
Contact: sip:0103@163.53.228.113:5060
Call-ID: 3a6fb38413883f3a05fed4c53e2f04c4@163.53.228.113:5060
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” sip:0103@163.53.228.113;tag=as6ca2516a
To: sip:0100@180.216.8.79:5061;transport=udp
Call-ID: 3a6fb38413883f3a05fed4c53e2f04c4@163.53.228.113:5060
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:0100@10.1.1.107:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
Max-Forwards: 70
From: “Pablo Android” sip:0103@163.53.228.113;tag=as6ca2516a
To: sip:0100@10.1.1.107:5061;transport=udp
Contact: sip:0103@163.53.228.113:5060
Call-ID: 3a6fb38413883f3a05fed4c53e2f04c4@163.53.228.113:5060
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:0100@10.1.1.107:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK3c389a6e
Max-Forwards: 70
From: “Pablo Android” sip:0103@163.53.228.113;tag=as6ca2516a
To: sip:0100@10.1.1.107:5061;transport=udp
Contact: sip:0103@163.53.228.113:5060
Call-ID: 3a6fb38413883f3a05fed4c53e2f04c4@163.53.228.113:5060
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:0100@10.1.1.107:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 163.53.228.113:5060;branch=z9hG4bK43ad6428
Max-Forwards: 70
From: “Unknown” sip:Unknown@163.53.228.113;tag=as29d04985
To: sip:0100@10.1.1.107:5061;transport=udp
Contact: sip:Unknown@163.53.228.113:5060
Call-ID: 0a1560ef115730501924599055da3c97@163.53.228.113:5060
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” sip:Unknown@163.53.228.113;tag=as29d04985
To: sip:0100@10.1.1.107:5061;transport=udp;tag=00131a6ed26c00184e2c6fe8-20662660
Call-ID: 0a1560ef115730501924599055da3c97@163.53.228.113:5060
CSeq: 102 OPTIONS
Server: Cisco-CP7960G/8.0
Allow: ACK,BYE,CANCE

Ok, so Cisco registers, can make calls but not receive

Maybe I missed something - but shouldn’t you change the 7960-setting
nat_address=""
to match the external IP, then it should see that?

I’ve tried that, the 7960 is in a local network (my house) which is behind an adsl router with a dynamic ip address. I’ve tried setting the dyndns host name as nat_address and no go. If i throw todays public ip address there, it’ll stop working tomorrow. I am sure people have them working behind dynamic ip addresses. In fact, it was working fine until it stopped working and I still don’t know the reason. I forgot to say, I upgraded the firmware on the 7960 to the latest version 8.12.00

Cisco phones were designed for corporate use, where there would be stable, intranet addresses, not NAT, and especially not addresses that are unstable. Some later SIP firmware may allow NAT, but I believe the original Cisco SIP firmware had no support for NAT.

Ok, it has a thre seetings called: ‘nat_enable’ ‘nat_address’ and ‘nat_receive_processing’, it was made by Cisco, which I am sure understands what NAT means and it has the latest firmware installed. Moreover it was working fine a week ago. I am sure there is an explanation.

if it works when you set the current public_ip - then well, then it works - thats your issue - the incoming traffic now matches the host.
(My Cable provider lets me have my IP for weeks and months as long as my lease is active so its not an issues for me to adjust NAT settings locally)

Why it worked a week ago is obviously some change… but I am willing to bet the denied invite is due to the mismatched ip

Ok, just tried setting public ip and same behaviour

Oh I forgot to say. It is running the latest firmware 8-12-00, Have also tried 8-8-00, 8-4-00 and I think also 9 and 11