On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. <[EMAIL PROTECTED]> wrote: > Hi list, > > I'm having trouble with calls placed to the PSTN (through a TDM card), > sometimes (a lot indeed) when I dial a number the callee party can't hear me > at all. > > My setup is: > > Asterisk 1.4.20.1 > Zaptel 1.4.11 > libpri 1.4.4 > Wanpipe 3.2.4 > > I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP > Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel > 2.4.16.60-0.23-smp > > I'm using the ulaw audio codec. > > There is no NAT between the Asterisk Server and the Phones (the phone and > the server are in the same network segment). > > What can it be??? > > Thanks in advance for any help/comment... > > > -- > Raul > Linux Counter #156439
Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T _______________________________________________ -- 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