Hi,
has anybody experience with 3CX-SIP-Trunks? I have installed Freeswitch
and added an extension (trunk-1). On 3CX-Side I added a SIP-Trunk
(generic sip provider) and I can see a succesfull registration. Now I
make a call to Freeswitch and want to pass it to 3CX-Sip-Trunk. But when
bridging to user/trunk-1 I get a "Originate Resulted in Error Cause: 41
[NORMAL_TEMPORARY_FAILURE]" error (with no delay).
3CX is installed as a VM in a Azure-Cloud behind NAT. I disabled local
iptables and the Firewall-Test on 3CX is all fine.
Freeswitch is on bare metal with a public ip.
When making a call, wireshark on 3CX-machine shows the INVITE followed
by "Destination unreachable (Port unreachable)". But I am not sure, if
this is a firewall-issue or something else.
Any idea?
Thanks in advance,
Thorsten
_________________________________________________________________________
Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com