Problems with asterisk/trixbox

I am facing following problems with asterisk:

  1. The sip softphones do not work beyond local lan (no sound when connected from outside). Can dial, but no sound. Please see this post (ref. voipnovice: … ic_id=2466 )

  2. Both the sip and iax softphones could not send dtmf signals to remote auto attendants. The remote autoattendant simply says either ‘no signals received’ or 'denies to provide service to rotary dial telephone’
    Please refer to voipnovice (mine) posts here: … 16&forum=2

  3. I could not locate how to make the hardphone connected to fxs of sipura3201 ring when PSTN calls arrive. The spa3102 is configured according to the

  4. I could not create a dial plan so that the user can choose between either voip or pstn services connected to the spa3102.

I am trying to fix this for a few days but could not. All suggestions appreciated. Thanks in advance.

I have been facing the above problems for more than 3 weeks and I consulted with everywhere on the net (asteriskguru, tb forum, voip-info) and the books (asteriskTFOT, Switching to Voip, Voip Hacks), but could not locate the solutions of the problems.

I am counting on the experts here. Thank you!

i’ll be quite frank … the fact you’re using TrixBox is going to actually put people (me included) off helping you. it’s got the potential to be the biggest can of worms imaginable. not that it’s really bad, but when it goes wrong there is so much config to wade through.

your 1-way audio is more likely your firewall/NAT interface than anything else. what ports/protocols are you forwarding and what does rtp.conf contain ?

DTMF issue, i posted to the thread you hijacked. a simple test extension and a few minutes experimenting would fix that.

[test-dtmf]; include this somewhere [from-internal] ?? exten => 456,1,AnswerRead(mydtmf) exten => 456,2,Read(mydtmf) exten => 456,3,Wait(2) exten => 456,4,SayDigits(${mydtmf}) exten => 456,5,Hangup()


so are your 2 outstanding issues the SPA configs ? first of all, go over the nerd very carefully. it would be most unlike Ward to post something that doesn’t work !!

have you looked on voxilla for help with the dialplan ? although i would say you would be better off routing all the calls through Asterisk.

Dear Baconbuttie:

Thank you for your response. The reason that I installed TB is because of the hype. Now I am thinking of using standalone version of asterisk or use astlinux or compile one on my own specific to my needs.

Like you said, TB keeps me whining around to several configurations. I prefer slackware/debian minimal install and use asterisk on it and I am sure the cpu load will be much lesser. Thanks for your kind suggestion.

Second, I could not understand what do you mean by “hijacked” the thread? I did according to you said i.e. adding dtmfmode=auto in the sip.conf and it worked, thanks to you and I think I have given due credit for your advice. If I missed to admire, here I extend the same to you. :wink:

I have


in the rtp.conf and I forwarded the same range of UDP ports for RTP, 5004 to 5089 for SIP (UDP and TCP), 4569 for IAX, 3478-3479 for STUN, 49152-49153 for RTP Multimedia and also put the box in the DMZ (my router was not giving access the softphones outside my firewall to the server even after forwarding the relevant ports until the server was put in DMZ).

And about the dtmf, it is fine internally but I am talking about inability to send dtmf tones to the remote IVR servers. I would certainly include the test-dtmf and check it as advised. But it does not seem to read out the dialled digits after I dialled the 456 ext if I understood right (please be considerate for some stupid questions because I am very new to asterisk world).

Thanking you a zillion. :smile: