Skip navigation

Tag Archives: Asterisk

We’ve had a recent issue with IAX2 trunks whereby any DTMF tones played locally are not audible at the remote side of the connection…

Interestingly tones were audible on inbound and internal calls, however, this means that IVR’s are completely non-navigable.

The problem turned out to be due to the fact that it appears DTMF traffic was being sent out over a separate UDP port to the rest of the IAX traffic….calls sounded fine, but DTMF traffic was being blocked due to it running on port 4571.  We’ve opened the range 4569-4571 and now all is working fine….

We have some Polycom IP 330 SIP handsets connected to a Trixbox.  Unfortunately, when using the default SIP.cfg that is downloaded via TFTP to the handsets, if the phone was off the hook (ie. a dial tone was already audible) the handsets would appear to time-out and give the message “All Circuits are Busy Now”. On looking at the call in the CLI (asterisk -rvvvv) it was apparent that only the first 9 digits were being dialled.

The solution was to edit the the /tftpboot/sip.cfg and look for the line that says:

<digitmap dialplan.digitmap=”[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT” dialplan.digitmap.timeOut=”3|3|3|3|3|3″/>

and replace with the correct number of digits for calls in your country…

<digitmap dialplan.digitmap=”[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxxxx|[2-9]xxxxxxxxxxx|[2-9]xxxT” dialplan.digitmap.timeOut=”3|3|3|3|3|3″/>

You then need to restart the handsets in order for this to work.