[Asterisk-Users] SIP native bridge problem

[Rich Adamson]

> I'm having a problem, I'm not sure if it has todo with the fact that my 
> phone is behind a NAT or not, but here it is..
> 
> My problem is when I call out, my asterisk system routes the call to my 
> SIP provider, whoever, as soon as the other party answers, asterisk 
> tries to make a native bridge for the call, and then the call drops 
> instantly.
> 
> However, if I keep asterisk in the middle (by anyable transfers), no 
> bridge is made and the call stays just fine.
> 
> My setup is so: Sipura-2000 -> NAT (Netgear router) -> cable/internet -> 
> colocated asterisk server -> SIP provider
> 
> The native bride I assume is asterisk trying to connect the RTP stream 
> directly from the Sipura to my SIP provider (thus asterisk keeping it's 
> self out of the media stream), and this is exactly what I would like to 
> have.
> 
> But I can't for the life of be figure ot why it's just hanging up once 
> the bridge is made.
> 
> Does anyone have any ideas how I could fix this, this is sort of 
> important, if it's just me because of my NAT causing it, would doing so 
> part forwarding and disable NAT support on asterisk and the Sipura fix 
> this problem?

To add to what others have already said... you can try:

- canreinvite=no on the asterisk def for the Sipura
- setting udp port forwards on your nat box will be difficult and somewhat
  unpredictable as each Internet sip device that trys to reinvite to you
  _may_ use different udp port number ranges. * uses udp 10k-20k, Cisco
  phones a different range, xlite yet a different range, etc. Each sip
  device vendor can choose whatever range they want. But, if the reinvite
  is always coming from the same device, you might find out what the range
  is for _that_ device and port forward those ports. (There could also be
  an issue of exactly which IP address that device might be trying to 
  contact, your internal IP or the nat'ed external IP.)
- get another registered IP address and assign it to your sip phone.
- replace your nat box with a sip-aware box.

For the most part, a sip device behind a nat box limits you to converseing
with the * box.