[SR-Users] topoh
Klaus Darilion
klaus.mailinglists at pernau.at
Fri Mar 26 16:47:43 CET 2010
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 at pernau.at
> <mailto: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.
>
>
>
>
>
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