I am in the process of migrating from my current (Asterisk 220.127.116.11 / Zaptel 1.4 ) server to a new server which will be running Asterisk 1.4 and DAHDI.
The current server uses the ZTDUMMY source for timing and has NO Digium cards to interface with my PBX. I strictly use SIP trunks to connect with my Iwatsu PBX.
The purpose of the asterisk server is to use MEETME to conference callers via SIP channels coming from my Iwatsu PBX. Some of these callers are located directly on my Iwatsu PBX using TDM or IP phones and some are calling from outside sources arriving via PRI’s which terminate into our Iwatsu PBX.
So far the experience is great. Very little complaints about echo.
Now I am moving to a new server and diving into the world of DAHDI. I am also using a Sangoma UT50 Voice Time USB Sync tool as a clock source (hoping it would be more “business class” than using ZTDUMMY).
My question is:
According to all the reading I have done, DAHDI is configured on a PER-CHANNEL basis unlike Zaptel.
Since I have no physical channels (only my Sangoma USB time source) and all calls arrive via SIP, is anything required in my DAHDI config to enable the same level of echo cancellation as I had with Zaptel?
Currently in my zapata.conf I use the following:
In some cases I have required echotraining=yes but have not had that variable set for some time now as I have no echo problems.
Please help me configure DAHDI so that my echo cancellation settings are similar to the ones I used with Zaptel and not ignored for SIP / MEETME.
Thanks in advance,