RE: [Serusers] rewrite & ACK forwarding problem - sr-users

27 Jun 2003


      We know that, we just want you to start guessing harder :)
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org] 
Sent: Friday, June 27, 2003 1:46 PM
To: Sean P. Robertson; Alexander Mayrhofer; serusers@lists.iptel.org
Subject: Re: [Serusers] rewrite & ACK forwarding problem
Gentlemen,
the answer is always the same -- send us message dumps and configuration
files. Misconfiguration hunting is hard to impossible otherwise.
-Jiri
ps -- I don't think this is related to ATA, there were some other
problems with it. I think it is an error in SER configuration.
At 06:39 PM 6/27/2003, Sean P. Robertson wrote:
...
I have the same problem and posed it to the group yesterday ([Serusers]
...
Ignored 200 OK message.) So far the only workaround that I have found 
is to use the rules in my gateway to rewrite the dialed digits before 
sending them to the PSTN PRI, thus leaving the origianl URI intact for 
SIP communications.
One person told me that this is a bug in the Cisco ATA, but it happens 
on my IPDialog phones also.  It seems to me that the INVITE is being 
processed by the SER dial rules and is rewritten, but the ACK is not.
Sean
_______________________________________________
Sean Robertson
NETXUSA
p. 800-289-6389
f.  864-233-4344                  "Ask me about Voice over IP."
http://www.netxusa.com/
----- Original Message -----
From: "Alexander Mayrhofer" axelm@nic.at
To: serusers@lists.iptel.org
Sent: Friday, June 27, 2003 12:15 PM
Subject: [Serusers] rewrite & ACK forwarding problem
...
Hi,
we're running SER together with a PSTN Gateway. Before a call get's 
forwarded to the gateway, we are rewriting the request URI to make 
rewriting on the GW as simple as possible:
route {
...
   strip(3);  # +43xxx -> xxx
   prefix("0"); # xxx -> 0xxx
   rewritehostport(xxx.xxx.xxx.xxx, 5060); # request to gateway
   route(1);
   break;
...
SIP call flow looks like (record route enabled):
(1) phone -> SER
INVITE sip:*43699xxxxxxxx@nic.at43.at SIP/2.0
(2) SER -> phone
SIP/2.0 100 trying -- your call is important to us
(3) SER -> GW
INVITE sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0
(4) GW -> SER
SIP/2.0 100 Trying
(5) GW -> SER
SIP/2.0 183 Session Progress
(6) SER -> phone
SIP/2.0 183 Session Progress
(7) GW -> SER
SIP/2.0 180 Ringing
(8) SER -> phone
SIP/2.0 180 Ringing
(9) GW -> SER
SIP/2.0 200 OK
Contact: sip:0699xxxxxxxx@xx.xx.xx.xx:5060
(10) SER -> phone
SIP/2.0 200 OK
Contact: sip:0699xxxxxxx@xx.xx.xx.xx:5060
[ call established, we can talk, but ... ]
(11) phone -> SER
ACK sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0
--> Here starts the problem. That ACK (11) never gets forwarded to 
--> the
Gateway, so after a few seconds, the GW starts over at (9). Those 
three packets (9-11) repeat a few times until GW runs into a timeout 
and drops the call.
I have the impression that SER can't match the packet to the previous
...
...
requests because of the rewritten URI. Is that correct?
The only output at debug level 3 is:
Warning: sl_send_reply: I won't send a reply for ACK!!
Is that a routing goof somewhere in our scripts or is that a more 
generic problem? Is the problem that the warning indicates somehow 
related to the fact that the ACK is not being forwarded?
Help appreciated.
cheers
axelm

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--
Jiri Kuthan            http://iptel.org/~jiri/
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