Jean, what I see is that L parameter in dial app is to specify the call time in milliseconds. Not sure if it helps, btw I see you're using ss7 technology, then I think this is for your calls outgoing the ss7 channel. For me, "sip->ss7" calls have normal ringback. But in "ss7->sip" calls, caller doesn't receive rb tone. Sorry if I misunderstood your case. Thanks. Caio On Wed, 16 Jun 2010, Jean C?rien wrote: > as an FYI, I use chan_ss7 1.3, the dial option is: > Options: (ss7/siuc/NUMBER,60,L(91580000:60000:30000)) > > and I am getting correct ringback tones (music or classic tones) from my > correspondents. I had to play with the r / R and eventually got there before > getting the tones. > > J. > > > > On Wed, Jun 16, 2010 at 2:13 PM, Claudio Furrer <elcaio at gmail.com> wrote: > > > vma, did you solved this ringback issue? > > i'm running chan-ss7 1.4 and there isn't ringback tone on incoming calls > > from > > pstn. > > > > thanks your comments. > > > > On Thu, 4 Mar 2010, vallimamod abdullah wrote: > > > > > > > > On Thursday4Mar, 2010, at 10:17 PM, <linuxgurus at gmail.com> < > > linuxgurus at gmail.com > > > > wrote: > > > > > > > You are using Digium hardware? If yes then you libss7 > > > > > > No, I am using sangoma A104d card. > > > > > > Regards, > > > - vma > > > . > > > > > > > > > > > > > > > -- > > > _____________________________________________________________________ > > > -- 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 > > >
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