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Incoming calls get a busy signal


Fisher Networks
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This is a new setup and not going too well. All incoming calls get are a busy signal. The log shows the attempt and displays a 404: Not Found error. Outgoing calls work fine. We are running pbxnsip 2.0.3.1715. The server has an external and internal IP. Ports 5060 and 5061 are open. Here is a snippet of my call (I masked some numbers):

 

 

[7] 2007/10/16 21:41:42: SIP Rx udp:4.xx.xxx.236:5060:

 

INVITE sip:+1206xxx4108@xx.xxx.173.140:5060;transport=udp SIP/2.0

 

Record-Route: <sip:4.xx.xxx.236;lr;ftag=VPSF506071629460>

 

Record-Route: <sip:4.xx.xxx.229;lr;ftag=VPSF506071629460>

 

Via: SIP/2.0/UDP 4.xx.xxx.236;branch=z9hG4bK347b.a1776e47.0

 

Via: SIP/2.0/UDP 4.xx.xxx.229;branch=z9hG4bK347b.127161b4.0

 

Via: SIP/2.0/UDP 4.xx.xxx.148:5060;branch=z9hG4bK506071629460-1192517067514

 

From: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148;isup-oli=62>;tag=VPSF506071629460

 

To: <sip:+1206xxx4108@4.xx.xxx.229:5060>

 

Call-ID: SEAMGC0120071017044305059728@xxx.xxx.63.11

 

CSeq: 1 INVITE

 

Contact: <sip:+1206xxx0931@4.xx.xxx.148:5060;transport=udp>

 

Max-Forwards: 67

 

Content-Type: application/sdp

 

Content-Length: 173

 

Remote-Party-ID: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148>;party=calling;screen=yes;privacy=off

 

 

 

v=0

 

o=- 1192596185 1192596186 IN IP4 xxx.xxx.31.53

 

s=-

 

c=IN IP4 xxx.xxx.31.53

 

t=0 0

 

m=audio 60724 RTP/AVP 0 18 101

 

a=rtpmap:101 telephone-event/8000

 

a=fmtp:101 0-15

 

 

 

[7] 2007/10/16 21:41:42: UDP: Opening socket on port 50952

 

[7] 2007/10/16 21:41:42: UDP: Opening socket on port 50953

 

[5] 2007/10/16 21:41:42: Identify trunk 4

 

[7] 2007/10/16 21:41:42: SIP Tx udp:4.xx.xxx.236:5060:

 

SIP/2.0 100 Trying

 

Via: SIP/2.0/UDP 4.xx.xxx.236;branch=z9hG4bK347b.a1776e47.0

 

Via: SIP/2.0/UDP 4.xx.xxx.229;branch=z9hG4bK347b.127161b4.0

 

Via: SIP/2.0/UDP 4.xx.xxx.148:5060;branch=z9hG4bK506071629460-1192517067514

 

Record-Route: <sip:4.xx.xxx.236;lr;ftag=VPSF506071629460>

 

Record-Route: <sip:4.xx.xxx.229;lr;ftag=VPSF506071629460>

 

From: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148;isup-oli=62>;tag=VPSF506071629460

 

To: <sip:+1206xxx4108@4.xx.xxx.229:5060>;tag=d7cd18744c

 

Call-ID: SEAMGC0120071017044305059728@xxx.xxx.63.11

 

CSeq: 1 INVITE

 

Content-Length: 0

 

 

 

 

 

[7] 2007/10/16 21:41:42: SIP Tx udp:4.xx.xxx.236:5060:

 

SIP/2.0 404 Not Found

 

Via: SIP/2.0/UDP 4.xx.xxx.236;branch=z9hG4bK347b.a1776e47.0

 

Via: SIP/2.0/UDP 4.xx.xxx.229;branch=z9hG4bK347b.127161b4.0

 

Via: SIP/2.0/UDP 4.xx.xxx.148:5060;branch=z9hG4bK506071629460-1192517067514

 

Record-Route: <sip:4.xx.xxx.236;lr;ftag=VPSF506071629460>

 

Record-Route: <sip:4.xx.xxx.229;lr;ftag=VPSF506071629460>

 

From: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148;isup-oli=62>;tag=VPSF506071629460

 

To: <sip:+1206xxx4108@4.xx.xxx.229:5060>;tag=d7cd18744c

 

Call-ID: SEAMGC0120071017044305059728@xxx.xxx.63.11

 

CSeq: 1 INVITE

 

Contact: <sip:+1206xxx4108@xx.xxx.173.140:5060;transport=udp>

 

Supported: 100rel, replaces, norefersub

 

Allow-Events: refer

 

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE

 

Accept: application/sdp

 

User-Agent: pbxnsip-PBX/2.0.3.1715

 

Content-Length: 0

 

 

 

 

 

[7] 2007/10/16 21:41:42: SIP Rx udp:4.xx.xxx.236:5060:

 

ACK sip:+1206xxx4108@xx.xxx.173.140:5060;transport=udp SIP/2.0

 

Via: SIP/2.0/UDP 4.xx.xxx.236;branch=z9hG4bK347b.a1776e47.0

 

From: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148;isup-oli=62>;tag=VPSF506071629460

 

Call-ID: SEAMGC0120071017044305059728@xxx.xxx.63.11

 

To: <sip:+1206xxx4108@4.xx.xxx.229:5060>;tag=d7cd18744c

 

CSeq: 1 ACK

 

Max-Forwards: 70

 

User-Agent: Bandwidth.com TRM (gold.13)

 

Content-Length: 0

 

 

 

Any idea what I'm looking at? It appears that everything is getting past the firewall, but no calls are accepted. I have a hunt group setup with the name 1206xxx4106 with an alias to 4107 (I know 4108 is mentioned in this log but the same happens to all three numbers).

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Is "localhost" your domain name on the PBX? What did you put into "Send call to extension" in the trunk?

 

As for your first question, do you mean on the software itself or the actual windows domain name?

 

Your second question was right on. I missed it. It says "Extension" and in it was an incorrect extension. I renamed the alias on the Hunt Group to 900 and changed that field in the trunk to 900 and it rang through.

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As for your first question, do you mean on the software itself or the actual windows domain name?

 

It is a common trap to rename the default domain name to something else... "localhost" has a special meaning to the PBX - it matches any name. If the name is changed and the PBX receives a request that does not 100 % match, the PBX will send a 404 code. But it seems that was not your problem.

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