Hello,
i have a problem with the inbound calls, i receive the following errors
Unable to create channel of type ‘SIP’
@ from-sip-external: “Rejecting unknown SIP connection”.
What can i do to fix the problem?
@frank83 Pastebin your trunk configuration (you can copy/paste the peer/user/reg string) being sure to mask all usernames/passwords, and the output from the following:
asterisk -r
core set verbose 9
<place call>
exit
Copy all resulting scrollback to pastebin.com and provide the link to that here.
What’s your default TLS port assignment setting? Also, check your inbound routes for misconfiguration. You may not have that inbound call set to a destination.
Please don’t guess wildly. We haven’t seen a debug or anything to even suggest he’s using TLS. Guessing compounds badly explained issues by posing as understanding.
Hello,
i try to explain better, i use freepbx to make a proxy SIP between my dialer and a provider dialer third-party<—>trunk<---->freepbx<---->trunk<---->provider SIP
I configured the public number on my Dialer and i configured the public number on freepbx like extension and for the outbound it works well, but when i try to make an inbound call it doesn’t work.
I noticed that if i remove the number from the freepbx extension it works for the inbound but not for the outbound 
Conf Trunk dialer
*Peer
dtmfmode=rfc2833
host=192.168.231.13
type=friend
insecure=port,invite
disallow=all
allow=alaw
*User detail
dtmfmode=rfc2833
host=192.168.231.13
type=friend
insecure=port,invite
disallow=all
allow=alaw
Conf trunk Provider
*Peer
host=x.x.x.x
username=xxxxx
secret=xxxxx
type=peer
fromdomain=sip.prov.com
dtmfmode=rfc2833
*User detail
host=x.x.x.x
username=xxxxx
secret=xxxxxx
type=peer
fromdomain=sip.prov.com
dtmfmode=rfc2833
nat=auto
insecure=port,invite
RegisterString
xxxx:[email protected]
@cullenl Maybe i understood the problem…for the both trunk i have also the redundant proxy, how can manage them with freepbx? For the inbound i have only one “Set Destinatin” and if i create a clone the system goes in error "a route for this DID/CID already exists! "
For the outbound i need only to clone the trunk and change the ip address of the peer?
@frank83
First off; one host= per peer. Create multiple peers for redundancy. What you have won’t work the way you think it will.
Second, FreePBX (or in your case, an old, badly out of date fork of it), is not a SIP router. It’s a PBX. You will have nothing but trouble attempting to do what you’re doing.
Third, I’m not interested in helping you get your autodialer working. Sorry.
@cullenl Sorry but i don’t want use the dialer for the outgoing, but i would like to know how FreePBX can management two address with the SIP provider. It’s the first time that i use this IPBX…