Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. _________________________________________________________________ Discover sweet stuff waiting for you at the Messenger Cafe.? Claim your treat today! http://www.cafemessenger.com/info/info_sweetstuff.html?ocid=TXT_TAGHM_SeptHMtagline2
Did you got a response for your questions? Recently found this URL in Google SiSky http://www.yeastar.com/ProductsforAsterisk.asp Regards, Alejandro Lengua On 9/6/07, John Meksavan <jmeksavan at hotmail.com> wrote:> > Has anybody ever integrated Skype with Asterisk? If you have, which > software would you recommend to accomplish such a task? ChanSkype? And > how > reliable are the calls? Did the DTMF tones work? Thanks in advance. > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070914/92ce8c01/attachment.htm
Alejandro, Thanks for replying. I did come by this website before. I was just wandering, if anybody actually tried Skype with Asterisk. My experimentation with the Sip Protocol and Asterisk is at end because I could never get QOS with any sip provider, ie Broadvoice, Vitelity, and Teliax, when connecting directly to the "General Internet". In my past experience, Skype has been the only VOIP that works very well. If I could just integrate this with my Asterisk at work, it would really make my boss happy.>From: "Alejandro Lengua" <alejandro.lengua at gmail.com> >Reply-To: Asterisk Users Mailing List - Non-Commercial >Discussion<asterisk-users at lists.digium.com> >To: "Asterisk Users Mailing List - Non-Commercial >Discussion"<asterisk-users at lists.digium.com> >Subject: Re: [asterisk-users] Skype + Asterisk >Date: Fri, 14 Sep 2007 13:02:19 -0500 > >Did you got a response for your questions? >Recently found this URL in Google >SiSky http://www.yeastar.com/ProductsforAsterisk.asp > >Regards, >Alejandro Lengua > >On 9/6/07, John Meksavan <jmeksavan at hotmail.com> wrote: > > > > Has anybody ever integrated Skype with Asterisk? If you have, which > > software would you recommend to accomplish such a task? ChanSkype? And > > how > > reliable are the calls? Did the DTMF tones work? Thanks in advance. > > > > > >>_______________________________________________ > >Sign up now for AstriCon 2007! September 25-28th. >http://www.astricon.net/ > >--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_________________________________________________________________ Get the device you want, with the Hotmail? you love. http://www.microsoft.com/windowsmobile/mobilehotmail/default.mspx?WT.mc_ID=MobileHMTagline
On Fri, 14 Sep 2007 18:53:53 +0000, John Meksavan wrote:>Alejandro, > > Thanks for replying. I did come by this website before. I was just >wandering, if anybody actually tried Skype with Asterisk. My >experimentation with the Sip Protocol and Asterisk is at end because I >could never get QOS with any sip provider, ie Broadvoice, Vitelity, and >Teliax, when connecting directly to the "General Internet". > > In my past experience, Skype has been the only VOIP that works very well. >If I could just integrate this with my Asterisk at work, it would really >make my boss happy.Huh? In my experience QOS is that it's enirely something I have to deal with....not the upstream providers. It matters most with respect to how I manage outbound bandwidth. Other traffic across my router can cause trouble for the voip service if I don't ensure adequate bandwidth. You simply can't get assured QOS over the internet. But the problem usually isn't the internet, it's the edge. For me QOS tagging has been less usefull overall than "traffic shaping"...which is an edge process for managing bandwidth at the user-end router. Skype is seriously problematic since it will use various ports in an unpredicatable fashion. That makes it difficult to manage its traffic since you don't know where its acting. I've tried two software programs intended to interface Skype to SIP (PSGW 3.0 & ?? ) and they both had problems. Most noticably high latency since they used the Skype API to take the call back to baseband audio then re-encoded it into a SIP call. I've found that using traffic shaping and G.792 codecs makes SIP very reliable for a small office on a decent DSL connection. Michael -- Michael Graves mgraves at pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves at mstvp.com o713-861-4005 c713-201-1262 skype mjgraves fwd 54245