I’ve just setup an AsteriskNow+FreePBX install, connected some softphones via SIP, that works fine, setup a IVR which I can test via the 7777 feature code, works fine;
Now I’ve setup a sip trunk with orbtalk, entered the details in the sip config and configured an inbound rule any:any to forward to my IVR, but the phone calls through, using tcpdump I can see the sip initiation traffic, but the call is not answered, in the asterisk cli I do not see any activity, setup details below;
the SIP trunk works fine when I test it on a x-lite phone
Any help is much appreciated;
Host dnsmgr Username Refresh State Reg.Time
sipgw3.orbtalk.co.uk:5060 N IPT-****** 45 Registered Wed, 13 Feb 2013 08:37:38
1 SIP registrations.
SIP OUT SETTINGS;
SIP IN SETTINGS;
Default Route (Name)
any DID / and CID
route IVR Menu
To test; get the inbound route to go to an extension and see if both parties can hear each other first, as the ivr may not be able to detect someone is on the other end… If not, is most likely your NAT settings.
The context needs to be from-trunk.
- Inbound route to go to an extension did nothing different
- There is no NAT in place this is a public IP
The thing that gets me is that I do not see the incoming call on the asterisk CLI, surly I should see it there even if it is not answered?
OK. Not familiar with orbtalk; Are the above setting provided by orbtalk? Did they offer a registration string for authentication or IP?
yes they provided the sip gateway user and pass, I’m having them take a look at the setting at the moment.
btw - on your inbound route, have you set “any DID / and CID” as you said above? Wouldn’t you need to have a DID point to the IVR?
Just been talking it through with orbtalk, VERY helpful I would recommend them, I can now see it hitting the astrisk cli but I’m getting;
[2013-02-13 14:30:06] NOTICE: chan_sip.c:23016 handle_request_invite: Call from ‘IPT-44xxxxx’ (18.104.22.168:5060) to extension ‘44xxxxxxxxxx’ rejected because extension not found in context ‘custom-wmsx’.
Resolved, I had context ‘custom-wmsx’ in the sip config nd it was not needed.
Using Orbtalk and can’t see the incoming call in the asterisk cli. Could you let me know what config orbtalk suggested you change to make it work please.