[Serusers] SER/PSTN Gateway conf

Greger V. Teigre greger at teigre.com
Tue May 15 17:05:44 CEST 2007


Yes, if you look at the BYE there is NO WAY SER can find out where to 
send it. The To header cannot (by RFC) be used for routing and no other 
header contains either location destination (as in Route) or aor (as in 
request uri).

To check your record routing, look at the INVITE outgoing from your SER 
towards the gw. 

If you do proper record routing, take the RFC snippets below and hit the 
vendor hard on the head. (it may seem that the GW is doing strict 
routnig, i.e. old-old RFC).
g-)

The relevant snippet from RFC3261, 12.2.1.1:

   If the route set is empty, the UAC MUST place the remote target URI
   into the Request-URI.  The UAC MUST NOT add a Route header field to
   the request.

   If the route set is not empty, and the first URI in the route set
   contains the lr parameter (see Section 19.1.1), the UAC MUST place
   the remote target URI into the Request-URI and MUST include a Route
   header field containing the route set values in order, including all
   parameters.

   If the route set is not empty, and its first URI does not contain the
   lr parameter, the UAC MUST place the first URI from the route set
   into the Request-URI, stripping any parameters that are not allowed
   in a Request-URI.  The UAC MUST add a Route header field containing
   the remainder of the route set values in order, including all
   parameters.  The UAC MUST then place the remote target URI into the
   Route header field as the last value.

Kostas Marneris wrote:
> As far as I know and understand it seems that the
> R-URI of the BYE mesg (1st line) which GW sends to SER is wrong.
>
> I expected to see :
> 	BYE sip:user at SIP_Phone_IP_Address SIP/2.0
>
> or the problem is that the 'Route:' header is missing from the BYE mesg
> so it's not loose_routed.
>
> Do you Record_Route the original INVITE ?
>
> Check also : rfc3261 / 16.12.1
>
>
> Kostas
>
>
> flavio wrote:
>   
>> 2007/5/15, Kostas Marneris <K.Marneris at otenet.gr>:
>>     
>>> Hello,
>>> The call signalling exchange is a good start but it's not enough.
>>>
>>> What some people here asked you is a capture of the SIP BYE mesg
>>> which GW sends to SER.
>>>
>>>       
>> Kostas,
>> Thank you for suggestion. I've already posted SIP BYE message found
>> with ngrep :-D. However here is:
>> U 2007/05/15 14:26:15.072602 10.28.52.105:5060 -> 10.28.19.202:5060
>> BYE sip:10.28.19.202 SIP/2.0.
>> To: <sip:06605XXXXX at 10.28.19.202>;tag=23d09968cf17c929.
>> From: "06720XXXXX"
>> <sip:06720XXXXX at 10.28.52.105>;tag=DXsf22384254031Yaz079220.
>> Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2384256106429488807479222.
>> Contact: <sip:10.28.52.105:5060>.
>> Call-ID: 238425403179219 at 10.28.52.105.
>> Max-Forwards: 70.
>> User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4.
>> CSeq: 23334 BYE.
>> Content-Length: 0.
>>
>> where:
>> 06605XXXXX is the called
>> 06720XXXXX is the callee
>> 10.28.52.105 is IP for GW
>> 10.28.19.202 is IP for SER
>>
>> Thanks for support.
>>
>>
>> ********************************
>> * (o<     ing. Patria Flavio
>> * //\      phone 0823451358
>> * V_/_  mobile 3407873357
>> *
>> ********************************
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>>
>>
>>     
>
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>   
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