[asterisk-users] Weird one way Audio situation

[Raúl Gómez C.]

Hi Steve and the rest of the list,

On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:

> 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
>
>
My Asterisk Server has two NIC with a channel bonding setup (Balance TLB)
connected to the same switch, and it does not have any firewall rule.


I'm attaching a file with the output of "sip set debug" on the CLI of a call
in this situation.

Although calls made with SIP phones have this strange behavior, when I place
a call with an analog phone connected to a FXS port of the same TDM card
(see below for full description) this does not happen.


Thanks, any help will be really appreciated...



-- 
Nacho
Linux Counter #156439



On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:

> On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. <nachogomez at gmail.com>
> 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
>
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