Hi guys,
Need your help again.
I have 2 DID’s (SIP trunk is Voip.ms)
How can I route calls from the 2nd DID?
I just migrated to Voip.ms with 2 DID’s. Issue is I cannot call in to 2nd DID, the first DID is in Outbound CallerID field. Where do I place it within FreePBX?
The error when I dial 2nd number is “All circuits are busy now.”
My inbound route contains those 2 numbers and calls are routed certain ways.
Please advise!
Thank you very much!
Some carriers require you to inspect ‘sip headers’ to correctly route inbound calls (well documented in these fora)
your outgoing call problem is unrelated to that.
Thank you Dicko,
Their support recommended to create a sub-acc and add 2nd trunk in FPBX. This might be a solution.
Indeed ‘Horses for courses’ and ’ if wishes were horses, then beggars would ride’
In other words, it is totally carrier bound. But the solution is usually trivial when you know their rules
Not VoIP.ms unless something changed.
First make sure you have your new DID setup to route to the correct place in VoIP.ms
If you have it routing in to the same sub account and pop as your main DID, then your calls will hit FreePBX.
You can watch the call hit FreePBX with sngrep from the command line. You will see it come in as an INVITE.
If it is hitting FreePBX, then you do not have your inbound route set up properly.
Since you have one inbound route working, if I had to guess, you didn’t set up the DID to route to the correct pop and account at VoIP.ms
That is NOT the answer. That is the exact opposite of the correct answer. You want and need exactly 1 trunk.
Yes, indeed 1st DID works as intended. Any idea what that option called at Voip.ms?
I would say that you essentially want two inbound routes, that can probably be best done on one ‘trunk’ which is just a confusing legacy term for a ‘network connection’ but in the SIP world trunks are kinda antiquated nonsense , call them what you want, the DID’s needs to be properly routed, and if you look at the SIP conversation ( sngrep is cool and immediate) it will be quite obvious where the ‘route’ is uniquely defined
On a failed incoming call, does anything appear in the Asterisk log? If so, paste that at pastebin.freepbx.org and post the link here.
If not, log into VoIP.ms and look at the settings for both of your DIDs. Set the broken one to be the same as the working one. If you are using complex call processing at VoIP.ms (IVR, filtering by caller ID, etc.) before sending the call to your PBX, please provide details.
thanks for reply, I don’t use call processing via Voip.ms, all happens inside of FPBX, it’s pretty default.
I have to see how I can retrieve asterisk logs.
You will find it much easier in realtime to familiarize yourself with sngrep.
After entire PBX reboot 2nd DID became routable! Wow, will it brake again?
Let us know 
Nope, because there’s no stopping you, though it might break again.
So, dear experts can you enlighten me how is it normally done in FreePBX regarding multiple DID’s and trunks.
Based what I saw in the past with Sangoma SIP, it created 2 trunks. Client had 2 DID, it was easy to route then.
Now, only one trunk but 2 DID’s. Even thought I fixed routing issue they still have some minor issue with caller ID not displaying, routing related.
So, what it the best practice in FreePBX to handle multiple DID’s (numbers) and trunk[s]?
Probably best for you to rethink what you believe is a ‘trunk’, think rather of it as a ‘connection’ to your provider over which one or many DID’s can flow in.
Also disassociate your belief that any outgoing ‘trunk’ (outgoing connection) has any one to one association to any of your DID’s which are incoming connection
(there are no ‘trunks’ per se in the SIP world that is a legacy term that only pertains to TDM/Analog connections and could be better deprecated in general FreePBX/ASterisk documentation)
DID is also misused in the SIP world. The origin is in the term direct in dialling, which is where a single trunk carried the internal extension number for each call, so that the PBX could route it without an operator. If you have a case where you only have one SIP world DID, that is nonsense; it is only direct in dialling to the ITSP, from the PSTN.
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