Martin, Thanks a lot. The problem was a turned on silence suppression on cisco ata 186. Now it seems to work perfectly. Thanks to everybody else too.
Michael On Tuesday 14 October 2003 05:04 pm, Martin Pycko wrote: > With the musiconhold and SIP-SIP call it turnes out that you need to > disable silence supporesion on your phones/gateways since the timing is > taken from the coming stream (but only for musiconhold AFAIK) > > regards > Martin > > On Tue, 14 Oct 2003, Michael Ulitskiy wrote: > > > Hi, > > > > I've found that neither Michael Manousos patch nor ztdummy driver > > do not fix musiconhold sound interruption problem up to acceptable quality > > level. Sound is choppy here anyway. > > It is my understanding (please correct me if I'm wrong) that if I have > > a Digium card in my asterisk machine, these problems should be gone > > 'cause those cards provide some reliable timing. So I have no choice > > and wish to buy a cheapest Digium card just for timing. I have no PSTN > > ports, it's pure voip environment here. > > So my question is whether any Digium card would be ok or I have to buy > > some specific card? I'm looking at X100P card as it is the cheapest one. > > Would it be enough? > > Thank you. > > > > Michael > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
