Hi Klaus.

Prefect, thanks. I used textops to replace the s= and o= lines also so now all fine. Maybe would be nice for topoh to do all this, point of the module is to secure your topology.

Thanks,
Stephen.

On 26 March 2010 15:47, Klaus Darilion <klaus.mailinglists@pernau.at> wrote:
http://sip-router.org/wiki/cookbooks/core-cookbook/devel#sip_warning_noisy_feedback

regards
klaus

Am 26.03.2010 16:39, schrieb dotnetdub:
Hi Klaus,

Thanks for reply.

Regarding the noisy feedback. Is it possible to just turn this off in
Kamailio? I've looked but the only way I can see to do it is to comment
it in source and recompile. It seems to be only transmitted certain times.

Would it not make sense for topoh module to mask all topology in the SIP
messages?

Regards,
Stephen

On 26 March 2010 12:53, Klaus Darilion <klaus.mailinglists@pernau.at
<mailto:klaus.mailinglists@pernau.at>> wrote:

   nathelper modules has a certain flag to change o= line too:
   http://www.kamailio.org/docs/modules/1.5.x/nathelper#id2468157

   Maybe mediaproxy has a similar feature?

   Regarding s= line. I think you have to change it manually using
   textops module.

   regards
   klaus

   Am 26.03.2010 13:36, schrieb dotnetdub:

       Hi List,

       We are using Kamailo 3.01 and testing topoh for starters:

       Kamailio3 = IP of Kamailio Server and Asterisk=IP of Asterisk server

       Observations:

       U 2010/03/26 12:05:56.665583 Kamailio3:5060 -> Asterisk:5060
       SIP/2.0 100 trying -- your call is important to us^@
       Via: SIP/2.0/UDP Kamailio3:5060;branch=z9hG4bK08fffea4;rport=5060^@
       From: "xxxxxxxxx" <sip:xxxxx@asterisk>;tag=as7ac63338^@
       To: <sip:Xxxxxxx@Kamailio3>^@
       Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@
       CSeq: 103 INVITE^@
       Server: kamailio (3.0.1 (i386/linux))^@
       Content-Length: 0^@
       Warning: 392 Kamailio:5060 "Noisy feedback tells:  pid=23782
       req_src_ip=Asterisk req_src_port=5060 in_uri=sip:xxxxxx@Kamailio
       out_uri=sip:Translated@*UPSTREAMPROVIDERSIP* Provider via_cnt==1"^@

       The noisy feedback contains everything that is hidden by topoh
       in the
       contact:

       Contact:
       <sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**>^

       Also:

       U 2010/03/26 12:06:00.125828 Kamailio3:5060 -> Asterisk:5060
       SIP/2.0 180 Ringing^@
       From: "xxxxxxx"<sip:xxxxxxxx@Kamailio3>;tag=as7ac63338^@
       To:
       <sip:xxxxxx@Kamailio3>;tag=fccd978-b342ea59-13c4-50022-82e974-27d2fa70-82e974^@
       Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@
       CSeq: 103 INVITE^@
       Via: SIP/2.0/UDP Asterisk:5060;branch=z9hG4bK08fffea4;rport=5060^@
       Record-Route:
       <sip:10.1.1.10;line=sr-N6IAzBclOBF4WLZfWBF7M.VLz6fLHRQ7z6srpxusg9MXgSIqHReEgJZwgSKEgSjLMc**>^@
       Record-Route: <sip:Kamailio3;lr=on;did=bff.60851b12>^@
       Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY^@
       Contact:
       <sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**>^@
       Content-Type: application/sdp^@
       Content-Length: 228^@
       ^@
       v=0^@
       *o=Upstream Switch Name 160624700 160624700 IN IP4 Upstream
       Providers IP^@*
       *s=Upstream Switch Name^@*
       *
       *
       The upstream info appears here also.

       I am using engage_media_proxy and using topoh with defaul
       parameters.

       Am I missing something ?

       Thanks,
       Stephen.





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