Callback not working

If anyone can please point me in the right direction, for testing purposes I made this scenario successfully, person calls a U.S. DID callback then takes place, and caller gets access to a disa to dial out. I was able to configure that nicely with IVR and announcements. Then for what I want to put in production I started with a clean install freepbx I purchased a Dominican Republic DID for a monthly rate and I attempted to replicate what I did, but Im unable to do the callback. The issue might be the config file that my proider asked me to put in sip_custom.conf which has about 50 different ip’s. I took a course of asterisk like 5 years ago so im like completely lost at this point.

[] host= dtmfmode=rfc2833 dtmf=rfc2833 type=peer context=from-SendMyCall insecure=very nat=never allow=all


from-sendmycall is not valid FreePBX context.

insecure=very is not supported it’s insecure=port,invite

dtmf is not an asterisk keyword for channel_sip
allow=all is just silly, disallow all then allow the CODEC’s you want to allow

thank you for the info, I will change it, does this mean that the provider is not compatible with freeppbx? Do I have to use context for the callback to work? I was able to this so easy with google voice motif. I been trying the entire day and all I get is this call cannot be completed as dial. Any other pointer would be greatly appreciated Scott.

I can’t say if the provider is compatable with Asterisk or not. Some of the commands are old, some are nonsense.

The bottom line is they just want to allow calls from a range of IP’s and there is nothing wrong with that.

I forgot to mention the context should be from-trunk

ok perfect thanks again, I will be removing dtmfmode=rfc2833 and dtmf=rfc2833 from that config file changing the contect to from-trunk and changing insecure=port,invite. This is to be done on sip_custom.conf. Of course I would need to add the trunk but shouldn’t I put the config info under peer and user details or should it only be entered sip_custom.conf.

seems like its on the providers end this time I get a Got SIP response 603 “Decline” back from Something I wasn’t getting before.

You need to leave the dtmfmode, it’s the dtmf that’s invalid.

These settings are for inbound and all that is required.

For outbound, create a trunk as usual.

Just to make sure I understand, when the callback feature is used it’s still consider an inbound call? I’m just trying to get a clear understanding how this is going to work out.

Now im getting this retransmission error, I know its related to nat, but I have a public ip.

Seems like I was missing something in the trunk to for outbound the provider informed me that I would have to enter my user and pass and host.