[Asterisk-Users] NATted phones transferring calls - BUG0003710

[Alexander Lopez]

I wont go into the details of NATs and how they work, they are beyond
the scope of this fourum,even though it is important that we understand
NATs and their function as it pertains to Asterisk.

If Asterisk is not in the media path and it CANNOT transfer the call if
it is NATed. If you have canreinvite=yes, then the phone will set up the
media path between them. If you are not using NAT then asterisk can
redirect the call, but if you are using NAT in cannot.

NAT + (canreinvite=yes) = Media path direct to the phones, transfers
will not work.
!NAT + (canreinvite=yes) = Media path direct to both phones, but since
phones are on routable networks phones can transfer.

NAT + (canreinvite=no) = Asterisk stays in the media path so transfers
and all functions will work.
!NAT + (canreinvite=no) = Asterisk stays in the media path so transfers
and all functions will work, but will NOT allow use of NATed Devices.

Hope this helps.



> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Douglas Garstang
> Sent: Tuesday, March 28, 2006 10:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] NATted phones transferring calls - 
> BUG0003710
> 
> I made a call from 3254102 to 2944093. I then tried to do a 
> transfer to 3254107. 
> IP addresses have been changed to protect the innocent.
> 
> It appears this related to bug 3710. It's unclear from the 
> bug if the problem has been fixed or not. If it hasn't, then 
> this seems pretty serious and would I guess affect any 
> NAT-ted phones ability to transfer calls.
> 
> Here's the REFER that the phone at 2944093 sends directly to Asterisk:
> 
> U 216.186.128.68:5060 -> 216.186.142.203:5060 REFER 
> sip:3254102 at 216.186.142.203 SIP/2.0.
> Via: SIP/2.0/UDP 216.186.128.68;branch=z9hG4bKba3b074892377BD1.
> From: <sip:2944093 at 216.186.128.68>;tag=C06397B-C3C1D97A.
> To: "Test User" <sip:3254102 at 216.186.142.203>;tag=as33e7dd7c.
> CSeq: 2 REFER.
> Call-ID: 4053b9972e7851f455d9d16e7706d3f4 at 216.186.142.203.
> Contact: <sip:2944093 at 216.186.128.68>.
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067.
> Refer-To: 
> <sip:3254107 at ipt.oneeighty.com;user=phone?Replaces=a13c692-349
> c9c70-7bd27d33%40172.31.99.4%3Bto-tag%3Das2a8d818b%3Bfrom-tag%
> 3D3DE1A6BE-7262B959>.
> Referred-By: <sip:2944093 at 216.186.128.68>.
> Max-Forwards: 70.
> Content-Length: 0.
> 
> Asterisk then goes and complains:
> 
> Mar 27 16:25:57 NOTICE[20511]: chan_sip.c:6734 
> get_refer_info: Supervised transfer requested, but unable to 
> find callid 'a13c692-349c9c70-7bd27d33 at 172.31.99.4'.  Both 
> legs must reside on Asterisk box to transfer at this time.
> 
> The phone's real IP address is 172.31.99.4. 
> I'm not really sure what the problem is except that it works 
> fine when there's no NAT involved. I can see the real IP 
> address in the dialog. I wonder if that's what is confusing Asterisk?
> 
> Doug.
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