At the SIP menu,
RTP Packet Size: 0.030
Erick Perez wrote:
where to change packet size?
On 3/9/07, Luki <[EMAIL PROTECTED]> wrote:
> Any gurus out there with experience with the SPA 2102 against
asterisk 1.2.14?
They work fine with Asterisk; most likely it's your wireless link
that's the
Erick Perez wrote:
where to change packet size?
Admin Login -> Advanced
Voice->SIP Tab
RTP Packet Size: .02
On 3/9/07, Luki <[EMAIL PROTECTED]> wrote:
> Any gurus out there with experience with the SPA 2102 against
asterisk 1.2.14?
They work fine with Asterisk; most likely it's your w
Via the web GUI of the phone. I don't remember exactly which screen it
is on. It defaults to .30, you need to change it to .20 to fix audio
problems with that device.
Erick Perez wrote:
where to change packet size?
On 3/9/07, Luki <[EMAIL PROTECTED]> wrote:
> Any gurus out there with exper
where to change packet size?
On 3/9/07, Luki <[EMAIL PROTECTED]> wrote:
> Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14?
They work fine with Asterisk; most likely it's your wireless link
that's the cause of your problem. The jitter buffer will only affect
receiv
Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14?
They work fine with Asterisk; most likely it's your wireless link
that's the cause of your problem. The jitter buffer will only affect
received audio, i.e. on your side, and since that is fine, you
probably don't need
Topology:
analog_phone-SPA2102-Navini_Wireless_Router--ISP--Asterisk
A ping against the asterisk server shows aprox 145ms roundtrip.
128kbps upstream
512kbps downstream
g729a as codec
signal quality of the navini router: 100%
The ATA operates correctly in every form, however somet
