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FreePBX (Asterisk) Sound disappears during a conversation. How to diagnose a problem?
Good day, ladies and gentlemen!
Given
- FreePBX (Asterisk, version 11)
- Digium TDM800 board
- 4 incoming lines
- PhonerLite SoftFon is used
- Users work on a terminal server - sound and recording are forwarded.
Problems:
Main 1. During a conversation, voice transmission can be turned off at any time (as if we turn off the microphone). It is solved by quickly turning off / on the microphone in PhonerLite
And also:
2. Strong echo, but not for all subscribers.
3. When calling numbers from IVR, additional numbers are not dialed.
Who faced tell me, please, in which direction to look?
Thank you.
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Solved by quickly turning off / on the microphone in PhonerLite
2. Is there an echo cancellation module on the board? Is it configured correctly?
3. Does the softphone generate DTMF? Is the DTMF transmission method set correctly?
1. We exclude the terminal server and work from a full-fledged PC (or better, a hardphone). We use for a while. This will reveal the cause - terminal - not terminal. It happens with terminals, yes.
2. Reveal "but not for all subscribers" - dependence. Who and under what circumstances. Most likely - problems with external calls? As Rsa97 asked earlier, is there an echo cancellation module? Also, is jitter buffer configured on the ZAP channel?
Although, according to claim 1, this moment can also be "repaired".
3. How is it configured in the softphone - DTMF, and how is it configured for peers-trunk - DTMF? Might be worth switching to out-of-band!
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