I configured freepbx 188.8.131.52 on a ML10 server in Nigeria which is connected to a GOIP 32 sim gateway.
I am unable to call out and in.
Every time I call out it get “Your call can not be completed as dialled, please check the number and try again”
Every time I call into the server I get a continuous dial tone that doesn’t end.
Please see logfiles attached and here are the peer details attached
CDR says that the calls were answered but my softphone (Linphone) doesn’t see these calls.
I have attached screenshots of the asterisk cli and peers
Please help, new freepbx
User context is wrong, it should be
from-trunk. Peer context is wrong, it should be
from-trunk, but neither of those affects outbound calls. Provide a full call trace with pastebin, not a screen cap:
Okay will do that asap. thank you so much.
“your call cannot be completed as dialed” message is an indication that the number you called could not be matched to any outbound route or local extension.
Please how do I correct this?
I have an outbound route and also setup pjsip extensions.
Have you checked that the outbound route is correctly defined? Can you call other extensions?
I changed it to from-trunk for both user & peers and its still the same.
Here is my call trace for pastebin. Please help
You should work with the gateway vendor to get this figured out.
You have configured your trunk so that the gateway registers to asterisk (
host=dynamic in the user section) and you have configured an IP address or hostname for outbound calling in the Peer section. Pretty much makes no sense at all.
Maybe you’d like to share your use case here also, if you want to get any help. I am aware of many a scammer using GSM banks with throwaway SIMs and frankly have little interest in seeing another one succeed.
Thank you for taking your time to respond, please can you explain what you mean by use case?
I have shared a trace call here on pastebin, is that helpful?
Yes I can call extension, but I can not call externally and call in.
Please any suggestions?
Let’s tackle one at a time. What external number are you trying to call? Post the number and a screenshot of your outbound routes
You’ve created a chan_sip trunk, but the device is sending calls to the pjsip port. You’ve enabled anonymous calls, so the INVITE is accepted, routed to the inbound route and sent to Queue 122. From the full log, the inbound call appeared to be successful, despite the device not being configured correctly.
I am trying to call 08127738594
Thank you for looking through my paste, so all i need to do is create a pjsip trunk instead?
Post the dialing pattern of the outbound route
Those are the only dial patterns I have setup on the outbound route above in the screenshot 2.
Those dial patterns don’t allow numbers with a starting 0, because they have an N. Use all Xs.
I did this and it didnt still call out.
I created a pjsip trunk and nothing changed, I am now extremely confused.
This was the call trace after I did that