Hi, we have a similar problem. When we try to make two skype-calls at a time, only one of them has working audio. For this to happen, both calls must be ringing at the same time. Does anyone know how to fix this?
Best regards, Marcus Hunger On Thu, Oct 22, 2009 at 10:45 AM, Samir Doshi <smrdo...@gmail.com> wrote: > Hi, > > I am facing audio issue in my skype for asterisk setup. > > *Flow of the call is like this.* > > e.g. > Skype users : > test2 > > Sip users: > 1001 > 1002 <--> test2 > > This both sip users 1001 and 1002 are register in same asterisk. And also > test2 skype user is register in same asterisk. > > Now 1001 is dialing skype user test2 (skypeout). So, test2 is getting call. > But as test2 skype user is register in our asterisk, our asterisk is getting > that call (skypein). And test2 is mapped with 1002 user. So when test2 user > call comes to asterisk our asterisk is dialing SIP/1002. And 1002 is getting > calls. But when 1001 and 1002 user is connecting they are not getting audio. > But this is working fine for only skypout and skypein. But when call come > back to asterisk audio issue is coming. > > I have checked rtp debug, But getting proper packages in rtp debug. > > I am attaching image of call flow. > [image: > ?ui=2&view=att&th=1247ba299ad2be9d&attid=0.1&disp=attd&realattid=ii_1247ba299ad2be9d&zw] > > Please help me to fix the issue. > > -- > > Thanks, > Samir Doshi > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Dipl.-Inf. (FH) Marcus Hunger - hun...@sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk
<<skypeforasterisk.jpeg>>
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
