[Serusers] rewrite & ACK forwarding problem

Marcelo Schmidt mschmidt at equinix.com
Fri Jun 27 23:38:33 CEST 2003


We know that, we just want you to start guessing harder :)

-----Original Message-----
From: Jiri Kuthan [mailto:jiri at iptel.org] 
Sent: Friday, June 27, 2003 1:46 PM
To: Sean P. Robertson; Alexander Mayrhofer; serusers at 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 at nic.at>
>To: <serusers at 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 at 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 at 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 at xx.xx.xx.xx:5060>
>>
>> (10) SER -> phone
>> SIP/2.0 200 OK
>> Contact: <sip:0699xxxxxxx at xx.xx.xx.xx:5060>
>>
>> [ call established, we can talk, but ... ]
>>
>> (11) phone -> SER
>> ACK sip:0699xxxxxxxx at 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
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers at lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
>_______________________________________________
>Serusers mailing list
>serusers at lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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

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