1.) The System is running with FreeTDS and UnixODBC for SybaseASE
Connectivity thru PhpAGI
TE110P, Rhino Channel Bank w/ 8 FXO and 16 FXS and some analog phones
Atcom320=4, BT200=10, GXP2000=1 and Polycom501=1
Panasonic Hybrid PABX
The callers can call from PSTN and access their data in SybaseASE with no problem…the headache exist when analog phones connect to i phones
and vice versa.
after knowig the cause of echo which is the “imbalance” of endpoints as Analog phones<-----Hybrid------->ip phones i tried to tweak zapata.conf
giving this values
i remove 90% of echo which is quite acceptable. but when i call out from polycom501 to TELCO (ananlog phones) the called person doesnt hear me but i hear the callee answering my calls and say hello… hello… helloooo. Poycom501 has no echo issue ever since it is put in the * system. only when i solve the other ip phones.
as i tweak the zapata.conf and changes the values everynow and then to solve the echo.
There are times that rfc2833 in polycom501 did not function, this will occur when the person called did not hear
its so ironic that when you troubleshoot and success. but in exchange as you go to another phone it will no work anymore.
could you please advice me what is the best echo canceller for this mix ip phones.
is there a asterisk hardware echo cancaller?
3)i dont like to use AGRESSIVE_CANCELLER as some said it will convert my TE110P into half duplex as in (over and out …roger roger… go ahaed)
signals
For Asterisk users that connect to the PSTN, the most common type of
echo is hybrid echo - the echo introduced by the impedance mismatch
between 2-wire and 4-wire telephone circuits. The echo manifests as a
distorted and delayed reflection of the users voice while in
conversation with an external party through the PSTN.
Asterisk itself offers a range of open source echo cancellation
routines that are moderately effective in eliminating the hybrid
mismatch echo that most Asterisk users experience. However, there are
cases in which these algorithms are not effective. To combat this,
Digium introduced DSP-based echo cancellation modules for our multi-
port T1/E1/J1 cards and our 24-port analogue card. Until now, our
users of the TDM400P and TDM800P have not been privilege to the
quality of this DSP-based echo cancellation.
Host-based Toll-Quality echo cancellation software is designed to
operate under 32-bit Linux and provides echo cancellation for
configurable tail lengths of 16ms (128 taps), 32ms (256 taps), 64ms
(512 taps), and 1024ms (1024 taps). See digium.com/en/
products/software/hpec.php
For new and existing customers of under-warranty Digium analogue
cards, this solution will be offered, with limited support, at no
charge. This solution is also available, with no support, to
customers of non-Digium Asterisk products at the per-channel rate.
While existing Digium carrier-grade echo cancellation solutions,
running on dedicated DSPs, do not impact the performance of the host-
processor, this new software canceller requires a moderate amount of
CPU time / MIPS in order to effectively quash echo. Digium recommends
that users requiring 8 channels at 1024 taps run a PC comparable to a
3.0 GHz Pentium 4, while users only requiring 4 channels at 1024 taps
may run a 2.5 GHz Pentium Celeron. The CPU requirements are such that
it is impractical to operate this echo canceller at 1024 taps for a
full T1 or E1 of channels.
Existing Digium customers may contact sales (sales@digium.com) for
more information about freely obtaining an HPEC license for use with
Digium analogue products. You will need to provide the serial number
for each of the cards that you want an HPEC licence.
The FXO ports on the CB probably need “tuned”. I think I read that you can use fxotune on Xorcom channel banks to improve echo but I’m not sure if it will work on the Rhino CB.