Post your extension.conf Vashkar On Oct 13, 2011 2:49 AM, "caio" <elcaio at gmail.com> 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. > > Running asterisk v1.8.7.0, dahdi 2.5.0.1, and chan_ss7 2.1.0. > > Thanks and regards, > Claudio > > -- > _____________________________________________________________________ > -- 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20111014/83080817/attachment.htm>
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