Hi, I’m new to FreePBX, so forgive me if I’m covering something simple here.
I’ve recently setup a PBX to use in our office and am having problems with the IVR menus when calling from a SIP phone. Everything works fine calling from a pots phone (mobile or landline), I’ve also setup an internal extension to redirect to the IVR which works fine from the X-lite client I’m using to test. It all stops working as soon as I call in on the SIP trunk from a SIP handset (all calls are delivered over the sip trunk).
I’m assuming it has to be something to do with the trunk, as even the x-lite client doesn’t work if I call the external number.
I’ve read various posts about DTMf settings, but first off I’m not clear if what I’m doing is correct. As in where I’m putting the dtfmmode setting. Here’s a snip from my trunk settings, is this the correct place to set it?
Assuming it is, I’ve tried all the settings I know about ( RFC2833/inband/auto/info ), none of which have worked.
I think that pretty much covers what I’ve done so far, other thank talking to my SIP provider, which wasn’t much help. Hoping you guys can point me in the right direction and get me up and running.
Not quite sure what version info is helpful, hopefully this is:
Ok done a little more on this, realised I had installed from an outdated ISO after breaking my current system, so have now put a fresh install in of the current FreePBX distro.
Still have the same problem with DTMF tones, but I’ve now been able to follow some instructions I found for enabling dtmf logging.
This is what I see when using the internal extension
[2013-01-16 22:44:59] DTMF[4232] channel.c: DTMF begin ‘1’ received on SIP/415-0000000b
[2013-01-16 22:44:59] DTMF[4232] channel.c: DTMF begin ignored ‘1’ on SIP/415-0000000b
[2013-01-16 22:44:59] DTMF[4232] channel.c: DTMF end ‘1’ received on SIP/415-0000000b, duration 120 ms
[2013-01-16 22:44:59] DTMF[4232] channel.c: DTMF end passthrough ‘1’ on SIP/415-0000000b
This is what I get when dialing the external number, routing out over the trunk.
[2013-01-16 22:45:37] DTMF[4233] channel.c: DTMF begin ‘1’ received on SIP/415-0000000d
[2013-01-16 22:45:37] DTMF[4233] channel.c: DTMF begin passthrough ‘1’ on SIP/415-0000000d
[2013-01-16 22:45:37] DTMF[4233] channel.c: DTMF end ‘1’ received on SIP/415-0000000d, duration 100 ms
[2013-01-16 22:45:37] DTMF[4233] channel.c: DTMF end accepted with begin ‘1’ on SIP/415-0000000d
[2013-01-16 22:45:37] DTMF[4233] channel.c: DTMF end ‘1’ detected to have actual duration 79 on the wire, emulation will be triggered on SIP/415-0000000d
[2013-01-16 22:45:37] DTMF[4233] channel.c: DTMF end ‘1’ has duration 79 but want minimum 80, emulating on SIP/415-0000000d
[2013-01-16 22:45:37] DTMF[4233] channel.c: DTMF end emulation of ‘1’ queued on SIP/415-0000000d
I also dialed the number from my mobile
[2013-01-16 22:53:17] DTMF[4237] channel.c: DTMF begin '1' received on SIP/87.127.240.98-00000010
[2013-01-16 22:53:17] DTMF[4237] channel.c: DTMF begin ignored '1' on SIP/87.127.240.98-00000010
[2013-01-16 22:53:17] DTMF[4237] channel.c: DTMF end '1' received on SIP/87.127.240.98-00000010, duration 270 ms
[2013-01-16 22:53:17] DTMF[4237] channel.c: DTMF end passthrough '1' on SIP/87.127.240.98-00000010
If I call in from another SIP handset on my hosted PBX, I get nothing at all logged in dtmflogs
It’s more likely to be a DTMF setting related to the actual phone extensions and not the trunk as you have detailed above. What sort of handsets do you have? Check the DTMF mode on each extension (normally defaults to rfc2833). Check the actual phones to see how they are set up for DTMF.
Have just put the unit on a public IP address, It’s now working correctly.
Wow, did you secure the box or are you giving out free calls? Your Internet connection will also be trashed by all the script kiddies beating on your box.
