<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:10pt"><div style="font-family: times new roman,new york,times,serif; font-size: 10pt;"><div style="font-family: arial,helvetica,sans-serif; font-size: 13px;">>> The echo between our extensions (using Polycom 550 handsets) disappears<br>>> once I removed the Digium echo module.<br><br>> Are you routing internal calls from SIP -> DAHDI -> SIP? The digium<br>> echo module will not have any effect on pure SIP <-> SIP calls. Do<br>> you have acoustic echo cancellation active on the Polycom phones?<br><br>Internal calls should be SIP to SIP. Yes we do have the acoustic echo cancellation active on the Polycom phones.<br><br><br>>> This is my system.conf:<br>>> bchan=1-23<br>>> dchan=24<br>>> echocanceller=mg2,1-23<br><br>> Did
you use these same settings when you were using the hardware echo module?<br><br>Yes, I believe so. I asked an Asterisk expert to make sure everything is working correctly when installing the hardware module. If the setting don't look correct, what should be there when we use the hardware module?<br><br><br>Thank you!<br></div></div>
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