Silence for first 20 seconds on returned outbound call

Hello! Asterisk Newb here.

Here’s the basic setup: We have an outbound call that calls and says “Press 1 now to be connected with a representative”. So far it’s working except for one problem: when the customer presses one, they’re sent to the representative correctly, but the representative can’t hear the customer for the first 20 seconds of the call. The customer can hear the representative just fine the entire time.

This is kinda a hard thing to google, so I haven’t had much luck, but the best I can figure is that this is some kind of codec issue that isn’t resolved until the call has already been going on for 20 seconds or so. If that’s the case, I’m not sure where to mess with the codec…ness.

The SIP looks like this:

[{COMPANY_NAME}](!) type=friend accountcode={COMPANY_NAME} directmedia=no context=itok dtmfmode=inband call-limit=30

I know that if directmedia was set to yes it could have this kind of issue, but it’s already set to no.

Any other ideas?


I figured out that the dialer part of this doesn’t matter. Now it’s a bit more simple:

When someone calls into the queue, the agent who answers can’t hear the caller for 20 seconds. Does anyone know what this could be?

Did you do a packet capture of a call? If packet capture is not possible, atleast do a “sip set debug on” in Asterisk CLI and copy/paste the output.

I am presuming your are using VoIP for your telephone connection. Is this correct?