I have a SIP exten with a SIP provider but I call to other PBX, my provider send me a SIP message (INVITE).
This happens when the other PBX transfers the call to an internal extension (internal extension of other PBX).
16:33:51.508500 IP (tos 0x0, ttl 252, id 27307, offset 0, flags [none], proto 17, length: 626) 172.31.239.100.5061 > 10.8.12.114.5060: UDP, length 598
0x0000: 0004 75e9 0795 0015 f9f5 b6f0 0800 4500 ..u...........E.
0x0010: 0272 6aab 0000 fc11 9fd1 ac1f ef64 0a08 .rj..........d..
0x0020: 0c72 13c5 13c4 025e 964e 494e 5649 5445 .r.....^.NINVITE
0x0030: 2073 6970 3a37 3435 3830 3330 4031 302e .sip:7458030@10.
0x0040: 382e 3132 2e31 3134 2053 4950 2f32 2e30 8.12.114.SIP/2.0
0x0050: 0d0a 5669 613a 2053 4950 2f32 2e30 2f55 ..Via:.SIP/2.0/U
this looks like a packet dump… SIP DEBUG from * is far more helpful.
lemme see if i got this- you have a SIP provider with an account. When you call another user of this same provider, you get an invite.
If that’s the case, this is normal. It’s called a reinvite, and it allows two SIP users to send media (voice data) straight to each other without bouncing off the SIP proxy. However I’d guess in your case it’s not working because youa re being invited to a 10.x IP (private, like 192.168). Unless you are on the same LAN or VPN as this guy, you will not be able to communicate in this fashion.
Try turning off reinvites, in sip.conf set canreinvite=no for this user, or put it in the [general] section to make it always apply. This may increase the bandwidth load on your * server.
The user on the other end is also probably misconfigured. canreinvite=no may help but if his PBX is not giving the correct IP, then it won’t work either way.
If you are both behind a NAT (and both using * of course) then you must both set externip= and localnet= in sip.conf.