Having an issue placing outbound calls, getting a "Your number cannot be completed as dialed" error message back

Hey guys! I’m having an issue placing outboud calls while inbound calls work just fine. my ISP is providing the trunking service through an SBC on my local network that they manage. I setup a pjsip trunk to the SBC and used it for my inbound route and this works great. they asked me to trunk to an external address (172.xx.xx.xx) and use this for my outbound route though it never shows as avaliable in FreePBX and when I try to place calls I get the error message mentioned above. I am leaning towards an issue with the ports, because I provided port numbers for both contact and media to my ISP and then forwarded them to the PBX. But I’m not sure if I updated them in the PBX correctly.

I will reply with the log

Sanatized Log, can’t post them directly because I am a new user

You don’t have an outgoing route that matches the dialled number.

Did you even define a dial-pattern?
Because outherwise non of you’re outbound call match a outbound route.

So for testing, and for testing only!.
You could set the patern.
From Dashboard homescreen goto:

  • Connectivity Tab
  • Select “Outbound routes”
  • Choose you’re create outbound route and click the Edit icon.

You then see the options of outbound route.
Click on tab “Dial patterns”

Here set in the [Match pattern] field “X.” < (notice the dot)

This means “accept this outbound route for ANY digit’s”
This system is actually to limit acces, so you won’t dial to foreign countries.
or speedup a emergency call like 112 or 911.

Gotcha, do you mean it looks like I missed a setting when setting up the outbound route like a dial pattern setup incorrectly? Sorry I’m relatively new to VoIP

Thanks for the reply, I setup a dial pattern earlier in the process but it looks like it hadn’t saved. Went ahead and followed your instructions and now I’ve got an “All circuits are busy now error” so progress right? I’ll capture some fresh logs and put them in the same google drive folder I used last time.

This likely means that the provider rejected the call.
At the Asterisk command prompt type
pjsip set logger on
paste the Asterisk log for the call at pastebin.com and post the link here. I suggest this over Google Drive (or any service requiring the reader to log in), because some potential helpers on the forum don’t want to be tracked.

Done, looks like its dropping the call because the trunk is reporting as offline, I get the same error message when I try to trunk to the onsite SBC instead of the WAN address even though it reports as avaliable

Pastebin: 18266 [2024-03-25 21:49:56] VERBOSE[2527] res_pjsip_logger.c: <--- Received SIP - Pastebin.com

Here are the logs while trunked to the SBC

Pastebin: 17052 [2024-03-25 21:45:10] VERBOSE[2527] res_pjsip_logger.c: <--- Received SIP - Pastebin.com

I think Invalid URI is the primary error. I assume you have mis-configured the host name of the endpoint.

Line 17396:
Via: SIP/2.0/UDP;received=;rport=5060;branch=z9hG4bKPj37d5b629-d7d9-4377-848c-fdf87df75859

The SBC apparently rejected the call because Asterisk said it sent it from but it actually came from

There also seems to be a address involved.
Please explain the networking setup. Does the PBX have multiple NICs? Multiple pjsip transports? Routing between these private subnets?

The PBX has two NICs on two different subnets, one is my lan and has the address ( and the other NIC is plugged directly into the SBC’s LAN port (the SBC is at and the second interface of the PBX is at for inbound calls I have a trunk between the PBX and SBC which works fine. while for ourbound calls my SIP provider wants me to trunk to a WAN address (74.xx.xx.xx) this trunk never shows up as avaliable in FreePBX. Sorry I meant to include all of this info in the original post

OK, so sending via the SBC is probably wrong.

As a test, set Qualify Frequency for the 74.xx.xx.xx trunk to 0 (with Registration None). That will force it to be available and we can look at the outgoing INVITE and any replies.

It’s not clear whether you have to send the INVITE via as a gateway or as a proxy. You can try the latter by setting SIP Server to 74.xx.xx.xx and Outbound Proxy to

Good morning, I went ahead and changed the qualify frequency and set the registration to none. I also changed the proxy of the outbound trunk to the address you provided, still getting an “All Circuits are Busy Now” message. Here are the logs I collected afterwards

With the response starting on line 50246, they rejected the call with a 400. Unfortunately, there is no indication as to why. As there are many possible things to try, please paste a log of an incoming call – most providers will accept the same formats they send you.

Hey, thanks for the late response. I am truly out of my depth here, the problem has completely flipped around on me now. When I try to place calls it rings, but the call never makes it to the phone I am calling. and I get a “The caller did not pick up” message. and when I try to call the system I get "The customer you are calling is temporarily unavaliable. The only changes I made were to the proxy field and I have since removed that, any ideas? lol

Edit: there is no hint of incoming calls on the logs when I try to dial into the system, almost sounds like an issue with my provider IMO

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.