We are having a FreePBX server integrated with a 3CX Server. we can generate out calls thru 3CX without any issues (External calls). but 3CX users can’t transfer there calls to Free PBX extensions. Once Dialed the extensions keeps ringing and the 3CX receives the below error.
Call or Registration to 9999@(Ln.10001@Asterix_Bridge) has failed. 0.0.0.0 replied: 408 Request Timeout; internal.
This is an invalid URI. SIP URIs can only have one @ and can’t have parentheses in the domain name part.
It’s not clear to me how much of the preceding comes from 3CX, rather than asterisk, but 0.0.0.0 isn’t a real IP address, it is used, in certain contexts, to mean I don’t care, but, if ever used as a destination, it would be plain wrong.
408 means there was no response at all. It is often generated by the requestor to indicate that, rather than being a response about a more remote timeout, so it is important to understand where this came from.
For those three reasons I think you’re asking in the wrong place.
Sorry, but people here deal with Asterisk, not FreePBX or 3CX, and even when
it’s Asterisk, chan_sip is no longer developed and is deprecated. Plenty of
people still use it, but debugging problems with it is likely to get the
simple response “upgrade to pjsip, which is supported, and then we can talk”.