Re: [Asterisk-Users] SIP client-NAT-Asterisk-NAT-SIP client. only works with canreinvite=no.

2003-09-11 Thread WipeOut . 
Can anybody explain me what does canreinvite=yes really does?

Not sure how technical an answer you want becasue it look slike you know whats going on but as I unterstand it canreinvite=no tells the UA that reinvite is not supported and so causes all the RTP traffic to be routed via the * server.. I played with many nat settings and port forwarding settings and it ended up that canreinvite=no was the solution to my problems as well.. the downside is that it requires more bandwidth at the central site but the plus side is that it works through NAT..


Any ideas on the client A to C (same LAN, same NAT box, unique outside 
IP, same * server)?

Only thing that springs to mind is to install another * box internally and then use IAX to connect the internal * box to the external one.. then the internal phone will call each other without crossing the NAT..

Later..
--
__
http://www.linuxmail.org/
Now with e-mail forwarding for only US$5.95/yr
Powered by Outblaze
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP client-NAT-Asterisk-NAT-SIP client. only works with canreinvite=no.

2003-09-11 Thread Alastair Maw 


WipeOut . wrote:

Any ideas on the client A to C (same LAN, same NAT box, unique
outside IP, same * server)?
Only thing that springs to mind is to install another * box
internally and then use IAX to connect the internal * box to the
external one.. then the internal phone will call each other without
crossing the NAT..
It shouldn't be *too* hard to change Asterisk such that it allows 
reinvites for particular netmasks. If you can ensure that your NAT 
clients are on different subnets, for example, this might be possible.

--
Alastair Maw [EMAIL PROTECTED]
MX Telecom - Systems Analyst
http://www.mxtelecom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP client-NAT-Asterisk-NAT-SIP client. only works with canreinvite=no.

2003-09-11 Thread austino 

 I have been trying to get SIP UA work with NAT but i have no been
successful has any one got  NATed ATA working(i.e an ATA witha private IP
working with NAT).
Asterisk registers the 192.168.0.3 Ip but no call go through at all,
infact there is no log of any call made on asterisk console.

can anyone please send me the sip.conf and ATA 186 configs of a NATed ATA
to working with *.
This what i have in my sip.conf

[]
type=friend
username=
transfer=yes
nat=yes
canreinvite=no
context=myata
host=dynamic
permit=0.0.0.0/0.0.0.0
accountcode=mi100

  ATA configs
IP=192.168.0.3
staticRoute=192.168.0.2
mask=255.255.255.0
dhcp=0
GkorProxy= (*'s public IP)
gateway= (*'s Public IP)
outbound Proxy=(*'s public IP)
NATIP= (host machine's Public IP)



On Thu, 11 Sep 2003, Jose Ildefonso Camargo Tolosa wrote:

 Hi!

 I have this configuration:

 SIP client A - NAT box A (real external IP) - Asterisk server (real
 IP) - (real external IP) NAT box B - SIP client B

 The echo test form any of the clients to the asterisk server is working
 just fine, even without canreinvite=no.

 When I try to call from SIP client A to B, wihtout the canreinvite=no in
 the sip.conf, the call doesn't even ring.

 Then I add the canreinvite=no to BOTH clients on the sip.conf, it starts
 to work.  The problem is that all voice data goes through my asterisk
 server, so the delay is longer.

 Also, this config doesn't work:

 SIP client A - NAT box A (real external IP, only one) - Asterisk
 server (real IP)
 SIP client C - NAT box A (real external IP, only one) - Asterisk
 server (real IP).

 When I try to call from A to C or C to A, the phone doesn't even ring,
 again, the echo test work just fine.

 SIP client A and SIP client C are in the same LAN, and both goes through
 NAT box A to the same asterisk server.

 In the case of clients A and C, the native bridge would be great,
 because it would save bandwith to both, my client, and me, and the voice
 delay would be almost nothing.

 My problem is: According to the data I got from the sip debug and the
 X-lite debug outputs, I don't see any reazon why the native bridge can't
 work, both clients gets different ports on the outside IP of the nat
 box, and that port is correctly recognized, and the reinvite packet is
 correctly sent.

 Can anybody explain me what does canreinvite=yes really does?

 Any ideas on the client A to C (same LAN, same NAT box, unique outside
 IP, same * server)?

 Thanks in advance,

 Sincerely,

 Ildefonso Camargo
 [EMAIL PROTECTED]


 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
Olaifa Augustine
General Data Engineering Services Ltd
18b oshin road,kongi bodija
p.o.box 29460, secretariate,
ibadan.
tel:- 234-2-8105156

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users