[OpenSER-Users] Strange Provider SIP - DSP fields in trying

[Marc LEURENT]

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One day I will learn to think twice before saying bullshits!
Thanks

Dan-Cristian Bogos a écrit :
> Marc,
> Not at all strange. As depicted by you also, you only have it set for
> 183 and 200. You need a line like this:
>  onreply_route[1] {
>       if ((isflagset(5) || isbflagset(6)) &&
> status=~"(100|180|183)|(2[0-9][0-9])") {
>  .........
> }
> 
> Cheers,
> DanB
> 
> On 9/4/07, Marc LEURENT <lftsy at free.fr> wrote:
> I already have:
> 
> onreply_route[1] {
>         xlog("!!!!  STARTING REPLY ROUTE\r\n");
>         #uac_restore_from();
>         if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") {
>                 force_rtp_proxy();
>         }
>         search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
> 
>         if (isbflagset(6)) {
>                 fix_nated_contact();
>         }
>         exit;
> }
> 
> ???? Strange ?? no?
> 
> 
> Dan-Cristian Bogos a écrit :
>>>> Hi Marc,
>>>>
>>>> this one should be handled in the same way u do with 183/200, catch it
>>>> in onreply route and force_rtpproxy()
>>>>
>>>> Cheers,
>>>> DanB
>>>>
>>>> On 9/4/07, Marc LEURENT <lftsy at free.fr> wrote:
>>>> I have a SIP provider: telecomitalia that use a cirpack!
>>>> The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...)
>>>>
>>>> When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified.
>>>> Both TRYING/RINGING with session description Connection Address/port are not replaced!
>>>>
>>>> OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address
>>>> because of force_rtp_proxy())
>>>> But other softphones like Twinkle don't understand it and still send RTP to the first IP!
>>>>
>>>>
>>>> Do you know how I can delete or modify TRYING/RINGING SDP?
>>>>
>>>> Thanks
>>>>
>>>> Below an example of a RINGING with SDP
>>>>
>>>>
>>>>
>>>> #
>>>> U 212.129.6.65:5060 -> 88.191.45.91:5060
>>>> SIP/2.0 180 Ringing.
>>>> Allow: UPDATE,REFER.
>>>> Call-ID: 866489712 at 192.168.95.47.
>>>> Contact: <sip:212.129.6.65:5060>.
>>>> Content-Type: application/sdp.
>>>> CSeq: 21 INVITE.
>>>> From: "Marc LEURENT" <sip:mleurent at sd-7501.dedibox.fr>;tag=542924903.
>>>> Record-Route: <sip:88.191.45.91;lr;ftag=542924903>.
>>>> Server: Cirpack/v4.39a (gw_sip).
>>>> To: <sip:0614730696 at sd-7501.dedibox.fr>;tag=01-07627-0003a50a-10152ba67.
>>>> Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP
>>>> 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym.
>>>> Content-Length: 303.
>>>> .
>>>> v=0.
>>>> o=cp10 118890098142 118890098144 IN IP4 212.129.46.35.
>>>> s=SIP Call.
>>>> c=IN IP4 212.129.47.194.
>>>> t=0 0.
>>>> m=audio 32896 RTP/AVP 8.
>>>> b=AS:64.
>>>> a=rtpmap:8 PCMA/8000/1.
>>>> a=ptime:10.
>>>> a=sendrecv.
>>>> m=video 65534 RTP/AVP 34 31.
>>>> a=rtpmap:34 H263/90000/1.
>>>> a=fmtp:34 .
>>>> a=rtpmap:31 H261/90000/1.
>>>> a=fmtp:31 .
>>>> a=inactive.
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>>
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