Hi , If possible Try libss7. On Fri, Oct 14, 2011 at 5:15 PM, Marek Cervenka <cervajs at fpf.slu.cz> wrote: > On 10/12/2011 10:47 PM, caio wrote: > > Hello, > > > > I have the following issue when calling from a sip endpoint to a pstn > > number. > > > > i don't know why the chan_ss7 is taking same values for called and > > calling party numbers. See below: > > > > -- Sent IAM CIC=30 ANI=202120 DNI=202110 RNI= > > > > The ss7 capture/dump shows isup with theses values as well. > > However, SIP packet is right (correct from/to, etc headers). Then, the > > call is returned with congestion tone. > > > > If I set the CALLERID(num) with the wanted number, the result is the > same. > > > > change in l4isup.c > > ALL "caller.id" to "connected.id" > > > -- > --------------------------------------- > Marek Cervenka > ======================================= > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > -- BIPIN RAGHUVANSHI OPERATION HEAD ASTERISK (DEVELOPMENT AND RESEARCH) WWW.EHORIZONS.IN 011-32323262 011-46334633 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20111015/26343ae8/attachment.htm>