May be i am wrong, but if its problem with codecs then you should use Donts use allow=all, use only single codec by adding following in sip and iax disallow = all allow=alaw On Wed, Mar 20, 2013 at 3:47 AM, mohamed amine ferhi < ferhi.med.am...@gmail.com> wrote:
> Thanks for your responses. > > @Bharat I already made this change : allow=all => No changes > > @VInod the chan_ss7 version that I use is from here > http://www.netfors.com/chan_ss7_free , any other URL suggested? > The use of asterisk 1.6 is not an option for me unfortunatly. > > > 2013/3/19 Vinod Dharashive <vdharash...@gmail.com> > >> Hi >> >> Download the chan_ss7 from trunk and compiles with asterisk. I have >> tested this with asterisk 1.6. It worked for me. >> >> Cheers! >> Vinod Dharashive >> Sent from BlackBerry® on Airtel >> >> -----Original Message----- >> From: Bharat Lalcheta <bharatlalch...@gmail.com> >> Sender: asterisk-ss7-boun...@lists.digium.com >> Date: Wed, 20 Mar 2013 00:12:25 >> To: <asterisk-ss7@lists.digium.com> >> Reply-To: asterisk-ss7@lists.digium.com >> Subject: Re: [asterisk-ss7] chan_ss7 | Link established but no voice >> >> -- >> _____________________________________________________________________ >> -- 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 >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > -- > _____________________________________________________________________ > -- 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 > -- Bharat Lalcheta
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