[SR-Users] topoh

[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 at 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 at asterisk>;tag=as7ac63338^@
>> To: <sip:Xxxxxxx at Kamailio3>^@
>> Call-ID: 2f7a1c040f4920d5591695956ec8c42c at 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 at 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 at Kamailio3>;tag=as7ac63338^@
>> To:
>> <sip:xxxxxx at Kamailio3
>> >;tag=fccd978-b342ea59-13c4-50022-82e974-27d2fa70-82e974^@
>> Call-ID: 2f7a1c040f4920d5591695956ec8c42c at 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.
>>
>>
>>
>>
>>
>> _______________________________________________
>> sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
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