DTMF issue and PTP problem with ISDN

Asterisk Asterisk Support
1

Hello,

I’ve installed Trixbox 2.6.3.2 (Asterisk 1.4) on a server which has a 4 port ISDN BRI card (OpenVOX B400P) and I’ve successfully setup about 40 SIP extensions in a test enviroment with Grandstream GXP2000 and Siemens 470IP phones. Everything works so far, although I have I few problems and a question:

  1. DTMF sending doesen’t quite work: For example, I’ve setup PIN codes for certain dial patterns and when I enter the correct PIN + ‘#’ asterisk doesn’t recognize it and the prompt times out. Both phones (GPX and Siemens) have 3 checkboxes for DTMF mode (Audio, SIP Info and some RFC 2xxx) and by default they only have “Audio” checked. When I checked “SIP Info” things started to work on the asterisk end, however certain external IVRs I’ve tested still don’t react at all to DTMF sending. Funny thing is when all three are checked (that RFC2xxx checkbox added) then asterisk PIN entering doesen’t work, so it seems that only SIP Info works for asterisk while Audio does nothing. Is that normal and expected?

  2. I have trouble configuring mISDN and asterisk to work in PTP mode (as opposed to PTMP by default). I’ve edited /etc/misdn_init.conf to “ptp_te=1,2,3,4” and after reloading the modules misdn-info shows that ports work in PTP mode, however I get “all circuits are busy now…” whenever dialing out and Asterisk CLI command “show misdn config” says “ptp: no”. The problem was solved when I phoned the phone company and asked them to switch me to PTMP, but I would still like to know the right way.

  3. Is there any way with my current setup to achieve “silent intrusion”, so that when two parties are talking (internal and trunk) the third internal party can include it’s extension into the ongoing conversation? When tech support is talking to a client a developer would like to eavesdrop… :smile:

2

RFC2xxx is the preferred mode, but both phone and sip.conf need to match.

Audio will not work if you are using a codec that compresses more than u-law or A-law, and it is expensive to handle.

Asking unrelated questions will tend to get you answers for just one of them on any support forum.

3

RFC2xxx is the preferred mode, but both phone and sip.conf need to match.

Thanks for the tip, will look into sip.conf.

Audio will not work if you are using a codec that compresses more than u-law or A-law, and it is expensive to handle.

You mean expensive as in CPU intensive?

Asking unrelated questions will tend to get you answers for just one of them on any support forum.

I am aware of that, but I didn’t want to bloat the main board with a bunch of noob questions… :smile:

4

Yes. I meant CPU intensive.

5

Is there any way with my current setup to achieve “silent intrusion”,
I am using
voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
Works with 1.4 or more