hey peeps,
I recently got Future-Nine phone service and my own ATA (PAP2T) and had the beeps too, until I found out what's causing it.
It's called
DTMF talkoff Google it for a full explaination.
First, ... if the caller is hearing the beep tones, it's from the ATA at your house,
or the Magicjack dongle, from Magicjacks side at their office. ... If YOU hear it, and the caller doesn't it's from the callers side ATA,
or THE Magicjack side... so, the Magicjack people must set their Asterisk FreePBX settings too. The
DTMF Tx Method: must be set to "
In-Band" to filter out the voice tones that trigger the beeps.
Both Tx and Rx needs this setting for the office using Asterisk FreePBX.
Here's the link explaining it all. (if you're just using the MJ dongles, YOU can't do anything, it's all up to the MJ office to make the changes)
http://blogtech.oc9.com/index.php?optio ... 6&Itemid=8
So far it seems to have worked. By the way,
www.Future-Nine.com is really great. Their service allows you to use ANY SIP device, and also an ATA with your SIP phone together, (they'll clone your account) so the both phones will ring and you won't have to stay by the ATA side phone all the time.
I'll still keep the Magicjack as backup for making calls, not using my free Future-Nine minutes, but the future-nine quality is just like a regular POTS line... all the time, no sputter, or break-up, or drop-calls.
like MJ sometimes has ... very nice!
For those here too lazy to click the link to see the settings text at the site, here it is...
Asterisk - Intermittent beep with linksys SPA 3102, PAP2, DTMF sounds heard during conversation
We had users reporting hearing DTMF sounds during a conversation. We have solved the problem by switching DTMF mode to INBAND in the SPA 3102 config, in the PAP2 config and into Asterisk/FreePBX configuration.
1) In your SPA-3102 html interface, go to /admin/voice/advanced, Line 1 and PSTN Line and set DTMF Tx Method: INBAND in both places.
2) Since we have an asterisk extension mapped to the SPA-3102 Line1, we also went into FreePBX and set the dtmfmode to: "inband" for that extension. (was previously set to SIP-INFO)
3) We also have a SIP trunk in asterisk mapped to the SPA-3102 PSTN line. We went into FreePBX under "Trunks" and set the DTMF mode for that SIP trunk by adding "dtmfmode=inband" in both "Incoming Settings" and "Outgoing Settings" for that SIP trunk. (was previously set to SIP-INFO)
PAP2
For the PAP2 devices :
1) In your PAP2 html interface, go to /admin/advanced, Line 1 and Line 2 and set DTMF Tx Method: INBAND in both places. (was previously set to SIP-INFO)
2) For extensions mapped to the PAP2 Line 1 and Line 2, we went into FreePBX under "Extensions" and set the dtmfmode to: "inband" for those extensions. (was previously set to SIP-INFO)