I just upgrade the SIPX SERVER from 3.10 to 4.0.4   But The AudioCodes still
receiving this kind of messages, now in this form

6d:21h:22m:35s SUBSCRIBE sip:[email protected]
<sip%[email protected]> SIP/2.0
Record-Route:
Via: SIP/2.0/TCP
192.168.0.120;branch=z9hG4bK-sipXecs-23d1847de02e601950a227e6233094d89246
Via: SIP/2.0/TCP
192.168.0.120;branch=z9hG4bK-sipXecs-23ce067a5be67089a6c4c26f1fa6aca6d2a7~41efb91317c490b01dba0712b5e82749
Via: SIP/2.0/UDP
192.168.0.120;branch=z9hG4bK-sipXecs-23c91029958c52321db109d93ed584126d95~7573e7cdfaf4501dcca57c7a42c07867
Via: SIP/2.0/TCP
192.168.0.84:60584;branch=z9hG4bK-d8754z-d61e5a5f3e30c620-1---d8754z-;rport=64535
Max-Forwards: 16
Contact:
To: "Vinos PC MIlagros"
From: "Silvana Quintero";tag=57679b76
Call-Id: NDM3M2VhNDM0NjIyYmI1MTc3OGI2MzRlYzI4ODc1ZTg.
Cseq: 2 SUBSCRIBE
Subje

6d:21h:22m:35s 120",response="935b7527fc54044349992817b0f636d1",algorithm=MD5
User-Agent: SMC 3455 release 1014c stamp 46298
Event: presence
Content-Length: 0
Date: Mon, 15 Mar 2010 23:04:50 GMT



6d:21h:22m:35s (      lgr_flow)(33872     )  |       | new
GetNewIndTransaction created - #16

6d:21h:22m:35s (     sip_stack)(33873     )  new AcSIPDialogAPI created - #17

6d:21h:22m:35s (      lgr_flow)(33874     )  |
|(SIPTU#16)SUBSCRIBE State:DialogIdle()

6d:21h:22m:35s (     sip_stack)(33875     )  SIPDialog(#16) changes
state from DialogIdle to DialogInitiated

6d:21h:22m:35s (     sip_stack)(33876     ) !! [ERROR]
AcTransactionUser::HandleSubscribe no event header in subscribe
request

6d:21h:22m:35s (      lgr_flow)(33877     )  ---- Outgoing SIP Message
to 192.168.0.120:5060 from SIPInterface #0 ----

6d:21h:22m:35s SIP/2.0 500 Server Internal Error
Via: SIP/2.0/TCP
192.168.0.120;branch=z9hG4bK-sipXecs-23d1847de02e601950a227e6233094d89246
Via: SIP/2.0/TCP
192.168.0.120;branch=z9hG4bK-sipXecs-23ce067a5be67089a6c4c26f1fa6aca6d2a7~41efb91317c490b01dba0712b5e82749
Via: SIP/2.0/UDP
192.168.0.120;branch=z9hG4bK-sipXecs-23c91029958c52321db109d93ed584126d95~7573e7cdfaf4501dcca57c7a42c07867
Via: SIP/2.0/TCP
192.168.0.84:60584;branch=z9hG4bK-d8754z-d61e5a5f3e30c620-1---d8754z-;rport=64535
From: "Silvana Quintero";tag=57679b76
To: "Vinos PC MIlagros";tag=1c2026663189
Call-ID: NDM3M2VhNDM0NjIyYmI1MTc3OGI2MzRlYzI4ODc1ZTg.
CSeq: 2 SUBSCRIBE
Record-Route:
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK


The Problem here is that the AudioCodes is filling this memory in some way ,
and after I time it drop the calls and show this message

6d:13h:14m:53s (    lgr_freeid)(2211117   ) !! [ERROR] no more free
IDs available

6d:13h:14m:53s (     sip_stack)(2211118   ) !! [ERROR] new
AcSIPDialogAPI can not be allocated

6d:13h:14m:53s (    lgr_freeid)(2211119   ) !! [ERROR] no more free
IDs available

6d:13h:14m:53s (     sip_stack)(2211120   ) !! [ERROR] new
AcSIPDialogAPI can not be allocated


This suscribe events are receiving by the AudioCodes every 3 to 4 mins.

Anyone know how to disable this kind of messages ?  ( reinstall the
softphones ??? maybe )

Thankx

(@'.')@ @('.'@)


2010/3/4 Scott Lawrence <[email protected]>

> On Thu, 2010-03-04 at 23:05 -0500, Andres Jaramillo wrote:
> >
> > Thanks for answer,
> >
> > Is this normal ?
>
> What's normal?
>
> > And why the audioCodes says that  One of the basic headers (To, From,
> > CSeq, Call-Id, Via) is missing in the message ?
>
> Well, neither the To nor the From headers have valid values in them
> (they are supposed to have SIP URLs - unless you edited them out when
> posting).  Not that it matters - those headers are essentially ignored
> in a SUBSCRIBE request.
>
> > Is there a way to eliminate these events ?
>
> The phones seem to be attempting to do peer-to-peer presence.  I suspect
> that could be turned off, but I don't know for sure.
>
> The routing of those requests to the gateway is a function of the
> addresses and the dial plans.  We have an outstanding issue to support
> filtering dial plan routing by method, but it's not on the road map yet.
>
> In theory, the gateway should just be rejecting these with some
> permanent error - do you have reason to believe they are causing some
> problem or are you just reading SIP logs for fun?  :-)
>
>
>
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