Christopher Bautista wrote:
> Hi guys
>
> Successfully installed and compiled patch into asterisk source. For info
> purposes, we are using an old trunk version, revision 66959 to be exact.
> with the patched version our initial tests resulted in soft to no
> noticeable echo. A little adjustment
Hi guys
Successfully installed and compiled patch into asterisk source. For info
purposes, we are using an old trunk version, revision 66959 to be exact.
with the patched version our initial tests resulted in soft to no
noticeable echo. A little adjustment on zapata.conf might eliminate the soft
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> mailman mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/mailman
>
> --
>
> Message: 2
> Date: Fr
Christopher Bautista wrote:
Hi guys
Update:
PSTN -> Asterisk -> SIP extensions
landline to sip extension . noticeable echo
mobile to sip extension . no echo
PSTN -> Asterisk -> PSTN . no echo
SIP -> Asteris
Christopher Bautista wrote:
> Hi guys
>
> Update:
>
>> PSTN -> Asterisk -> SIP extensions
>
> landline to sip extension . noticeable echo
> mobile to sip extension . no echo
>
>> PSTN -> Asterisk -> PSTN . no ec
Mobile operators operates own echocancellers on every
channel, but not landlines. I operate chan_ss7 witch
SANGOMA cards with HW echocanceller - and it helps on
landlines.
On Friday 11 April 2008 06:55, Christopher Bautista wrote:
> Hi guys
>
> Update:
> > PSTN -> Asterisk -> SIP extensions
>
>
Dear Christopher,
We had same issue couple of months ago. We detected SS7 was not starting
echo canceller.
We produced a patch and I guess Matthew commited to trunk version. Please,
refer to bugs.digium.com and check the patch.
[]´s
Daniel
On Thu, Apr 10, 2008 at 10:55 PM, Christopher Bautis
Hi guys
Update:
> PSTN -> Asterisk -> SIP extensions
landline to sip extension . noticeable echo
mobile to sip extension . no echo
> PSTN -> Asterisk -> PSTN . no echo
> SIP -> Asterisk -> SIP
Hi
Can anyone help me with my problem?
We deployed an asterisk box with libss7. We used OpenVox's
D410P/D410E cards.
Everything is up and running (CIC etc.), except when we tried calling we get
echo on Voip extensions.
We tried isolating it.
PSTN -> Asterisk -> SIP extensions. noticea
