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Trixbox (asterisk) registered, but no calls passing in or out.

glenboro
Newbie
Posts: 2
Registered: 11-11-2008

Trixbox (asterisk) registered, but no calls passing in or out.

Hello all. I am trying to configure a trixbox pbx to use a plusnet VoIP SIP trunk.  I'm certainly no expert, but have a degree of knowledge based on research.
The trixbox server is configured with extensions that function, and was working with ISDN pri. Until I put the box into service I want to connect using a SIP trunk, so the existing pbx is connected to the existing pbx until the trixbox system replaces it. So I have disconnected the ISDN pri, and want to route all calls over a sIP trunk via plusnet VoIP.
I have connected the SIP trunk with:
Outbound Caller ID: 01XXXXXXXXX
Trunk Name: 01XXXXXXXXX
PEER Details:
fromdomaim=sip2.plus.net
host=sip2.plus.net
context=incoming
canreinvite=no
insecure=very
type=peer
authuser=UUUUUU
fromuser=UUUUUU
username=UUUUUU
secret=PASSWORD
nat==yes
Where UUUUUU=plusnet extension less the leading "4", PASSWORD=account password
User context=UUUUUU
User Details:
canreinvite=no
fromuser=UUUUUU
host=sip2.plus.net
insecure=very
nat=yes
qualify=yes
secret=PASSWORD
type=friend
username=UUUUUU
Register String:
UUUUUUTongueASSWORD@sip2.plus.net/UUUUUU

I have a default catch all outbound route with a dial pattern of:
0|.
and a default inbound route that just rings all extensions for any incoming call (no DID or Caller ID set).
I have "Allow Anonymous inbound SIP Calls?" Set to yes.  I also have opened ports 5060 and 5082 on the router.
The trixbox status page shows the trunk registered:
Host                                      Username          Refresh      State                Reg.Time               
sip2.plus.net:5060              UUUUUU            105      Registered          Tue, 11 Nov 2008 09:03:19
In SIP Peers it shows:

Name/username              Host            Dyn Nat ACL Port    Status             
UUUUUU/UUUUUU              79.135.125.154      N      5060    OK (43 ms)
201/201                    192.168.123.100  D  N      5060    OK (113 ms)
202/202                    192.168.123.102  D  N      5060    OK (113 ms)
01XXXXXXXXX/UUUUUU        79.135.125.155              5060    Unmonitored         
4 sip peers [Monitored: 3 online, 0 offline Unmonitored: 1 online, 0 offline]

Incoming calls are answered by the plusnet VoIP voicemail, and outgoing calls  get an "All circuits are busy" message.
Any help to resolve this quickly would be appreciated.
Paul Chambers.
4 REPLIES
Community Veteran
Posts: 6,258
Thanks: 433
Fixes: 39
Registered: 30-07-2007

Re: Trixbox (asterisk) registered, but no calls passing in or out.

Sorry I'm not that clued up on Asterix/Tribox but a couple of suggestions:-
Have you checked the Gradwell support for setting up Tribox https://portal.gradwell.net/support.php/knowledgebase/browse/category_id/44
sip2.plus.net is being replaced with sip.plus.net , not sure when.
Try sip.plus.net in your config instead.
Have you checked that your PlusNet voip account works directly with a softphone such as SjPhone ?
glenboro
Newbie
Posts: 2
Registered: 11-11-2008

Re: Trixbox (asterisk) registered, but no calls passing in or out.

Quote from: ianwarrilow
Sorry I'm not that clued up on Asterix/Tribox but a couple of suggestions:-
Have you checked the Gradwell support for setting up Tribox https://portal.gradwell.net/support.php/knowledgebase/browse/category_id/44
sip2.plus.net is being replaced with sip.plus.net , not sure when.
Try sip.plus.net in your config instead.
Have you checked that your PlusNet voip account works directly with a softphone such as SjPhone ?


Thank you for your response.
What is the relationship between gradwell and plusnet?  What changes need to be made to the gradwell support page recommendations when using plusnet?
Am I correct in dropping the leading number from the extention number? Yes, It works in a softphone without that leading number (I've used sjphone and voiper).
Thanks again.
RichSmol
Grafter
Posts: 709
Registered: 29-10-2007

Re: Trixbox (asterisk) registered, but no calls passing in or out.

yeh you'll need to drop the front 4 on the extension number to auth against the gradwell platform.
Generally, you need to make the following changes -
SIP server: sip.plus.net
SIP proxy port: 5060
Outbound proxy: natproxy.plus.net or nat.plus.net (1st one is a trial nat proxy but offers better performance)
Outbound proxy port: 5082
thats about the only difference really. Let me know what problems your having as I have seen an improvement in performance when asterisk users are configured on another instance of the gradwell platform which offers some further asterisk support.
cheers
Rich
Community Veteran
Posts: 6,258
Thanks: 433
Fixes: 39
Registered: 30-07-2007

Re: Trixbox (asterisk) registered, but no calls passing in or out.

Quote
What is the relationship between gradwell and plusnet?  What changes need to be made to the gradwell support page recommendations when using plusnet?

Gradwell now host the PlusTalk voip service on behalf of PlusNet.
Changes as Rich identified