FreePbx: Busy on incoming calls

FreePBX
1

Hey there I’ve recently been trying to setup a simple phone system to use in my everyday life. I have the outbound calls set up so I can call outside the network. The main problem I have now is that every time I try to call the inbound number I receive a busy signal. I am very new to phone systems and have read a decent amount of documentation as well as followed a couple tutorials online as to how to set it up.

Had trouble setting up outbound calls till we figured out it was the provider that had a wrong setting. Could the provider be related to my current issue?

The two main things I’ve noticed is when I use the command “sip show registry” it doesn’t show anything registered but it still works. In the debug log [CSeq: 102 OPTIONS] is one of things I think could be wrong but I don’t know enough to proceed properly. I have provided my trunk settings and my debug log to make things easier. If theres anything else I can post please say so. Thank you for any help you can provide.

type=friend
secret=Abcdefghi123
username=
host= sip provider ip
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
insecure=invite,port
fromuser=
fromdomain=
allow=all
nat=yes
port=5060

<— SIP read from UDP:192.168.0.32:5060 —>
SIP/2.0 200 OK
To: sip:[email protected]:5060;tag=2bd94efa8f5eac23i0
From: “Unknown” sip:[email protected];tag=as798f4738
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.0.35:5060;branch=z9hG4bK3379a689
Server: Cisco/SPA525G2-7.5.5a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.0.32:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:5060;branch=z9hG4bK39f77834;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4537a2ab
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.3(11.21.2)
Date: Fri, 09 Dec 2016 19:16:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

2

First check with your provider if you need to have SIP registration with them.
If yes, try the following configuration:

type=peer
secret= {password}
defaultuser= {username}
host= {sip provider hostname or ip}
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
insecure=invite
nat=no
port=5060
callbackextension=20 {any existing extension number here}
qualify=yes

Then check your registration and provide sip debug output in case of registration failure. We need only SIP messages only between your server and your provider. Make sure no information is lost due to forum formatting.

Please do not touch your router yet.

3

Since you are using NAT, make sure you include NAT=YES in the PEER settings.

You may also need to set up your incoming router (where the NAT is happening) to allow connections from your TISP to your router, and make sure that you redirect the ports your using to the right ports on your PBX.

4

This is what I’m basing the authentication on.

From the provider website which is thinktel.
We authenticate calls by both methods:
• IP address: the VIA and Contact header must contain your contact IP address and port.

• SIP credentials: the supplied username and password credentials must be sent in the authentication digest
for all SIP requests that require authorization (INVITE, UPDATE, BYE, etc).

No results with the new config. I am attempting to get the ports forwarded but the router is having trouble saving settings. Not sure if it’s related or not. I have changed all credentials to Ip’s instead oh hosts to try an go along with what there saying above.

Currently troubleshooting as we go along. The advice and amazingly quick reply is very appreciated.

Connected to Asterisk 11.21.2 currently running on localhost (pid = 2035)
Reliably Transmitting (NAT) to 192.168.0.32:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:5060;branch=z9hG4bK0f225c08;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as2ba7feeb
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.3(11.21.2)
Date: Fri, 09 Dec 2016 21:17:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.0.32:5060 —>
SIP/2.0 200 OK
To: sip:[email protected]:5060;tag=2bd94efa8f5eac23i0
From: “Unknown” sip:[email protected];tag=as2ba7feeb
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.0.35:5060;branch=z9hG4bK0f225c08
Server: Cisco/SPA525G2-7.5.5a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog '[email protected]:5060 ’ Method: OPTIONS
Reliably Transmitting (no NAT) to 208.68.17.52:5060:
OPTIONS sip:208.68.17.52 SIP/2.0
Via: SIP/2.0/UDP 192.186.119.106:5060;branch=z9hG4bK3394a44e
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as7d6d4606
To: sip:208.68.17.52
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.3(11.21.2)
Date: Fri, 09 Dec 2016 21:17:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:208.68.17.52:5060 —>
SIP/2.0 401 Unauthorized
WWW-Authenticate: Digest realm=“edm.trk.tprm.ca”,nonce=“a623e354d4d7”,stale=fals e,algorithm=MD5,qop="auth"
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
From: “Unknown” sip:[email protected];tag=as7d6d4606
To: sip:208.68.17.52;tag=209.197.130.80+1+22f707+1b25f9f
Via: SIP/2.0/UDP 192.186.119.106:5060;branch=z9hG4bK3394a44e
Server: DC-SIP/2.0
Organization: MetaSwitch
Supported: resource-priority, 100rel
Content-Length: 0

<------------->

5

the only thing I’ve done to the router so far is attempt to port forward.

6

When I use “Sip show registry” it shows as unregistered now instead of nothing being there.

7

Sorry for the spam but I ended up finding a setting in the router enabling Sip passthrough. I now get “This number is not in service jazz” instead of getting just a busy signal so in my mind this is a little progression.

8

Removed the CID and DID. able to make calls now! One happy little nerd over here. Thanks for the push forward everyone. My day has been made.

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