eh.. they told me it was a CPU problem. that v1 of the pap2na didn't have the horsepower to transcode 2 channels.
anyone know if the Sipura 2100 can do 2 simul 729 calls? -Matthew ----- Original Message ----- From: "Matt Klein" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Wednesday, February 02, 2005 3:43 PM Subject: Re: [Asterisk-Users] Linksys PAP2 / RT31P2 + multiple G.729 calls > of course it won't. neither can the ata. > > they're cheap, it was a licensing decision. > > i look forward to v2. > > -m > > On Wed, 2 Feb 2005, Matthew Boehm wrote: > > > Holy Crap!!!! > > > > I have just verified this! The linksys PAP2-NA will NOT SUPPORT 2 > > SIMULTANEOUS G729 CALLS! > > > > And I just got off the phone with some super-level technician at linksys and > > he said they knew this all along!! > > What bastards! > > > > Anyway, he told me they are comming out with the PAP2-NAv2 in a few months > > which WILL allow 2 simul G729 calls. > > > > -Matthew > > > > ----- Original Message ----- > > From: "Leonardo Gomes Figueira" <[EMAIL PROTECTED]> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <[email protected]> > > Sent: Tuesday, February 01, 2005 12:21 PM > > Subject: [Asterisk-Users] Linksys PAP2 / RT31P2 + multiple G.729 calls > > > > > >> Hi, > >> > >> anyone can confirm if the Linksys's ATA and Router (PAP2-NA and > >> RT31P2-NA) have the same limitation of just one G.729 call like the > >> Cisco ATA 186 ? > >> > >> I'm testing both appliances here and found this issue but could not > >> confirm this anywhere (nothing on the manual, no document or post from > >> any user about this). > >> > >> In my tests they use G.729 only on the first call and G.711 on the > >> others. If I disable G.729 on sip.conf for both peers they can't > >> establish a second call (ring but drop after answer). If there is > >> allow=ulaw on sip.conf I can establish 1 G.729 call and 3 G.711 with > >> RT31P2-NA (using three-way calling). > >> > >> In PAP2-NA if I mark "Use Pref Codec Only" and there is one call > >> established, when I call the PAP2 it replies with "488 Not Acceptable > > Here". > >> > >> Thanks, > >> > >> Leonardo > >> > >> -- > >> > >> Leonardo Gomes Figueira > >> [EMAIL PROTECTED] > >> _______________________________________________ > >> Asterisk-Users mailing list > >> [email protected] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
