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

Dan-Cristian Bogos dan.bogos at gmail.com
Tue Sep 4 13:41:40 CEST 2007


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:
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> Hash: SHA1
>
> 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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