27 Jun
2003
27 Jun
'03
5:38 p.m.
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(a)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:
--
Jiri Kuthan http://iptel.org/~jiri/
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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(a)nic.at>
To: <serusers(a)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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