I’ve recently upgraded from Asterisk 1.4.something to 1.8.1.1
I’m having a problem with Meetme conferencing. Let me try to explain my setup …
I’m in the UK. I don’t have any Asterisk hardware fitted to the machine that is running Asterisk. At 1.4.something I was using ztdummy for timing, now I’m using DAHDI.
My Asterisk server is behind a Broadband router running NAT.
I have a PSTN number (from Gradwell) that is delivered by IAX2. I also have a phone connected locally via a Linksys PAP2T.
Incoming calls from the PSTN number hear a voice menu which is essentially 1 for Voicemail, 2 to ring the local phone, 3 to go into a conference.
The local phone can dial extension 8001 to go to the conference.
What I’m seeing is that multiple incoming calls on the PSTN number can go to the conference and everything works, however if the local phone goes to the conference, audio from the local phone goes to the external participants, but the local phone does not hear any audio from the external participants.
If I make a call (from another PSTN number) to the incoming PSTN number and then go direct to the local phone, I have two-way audio (i.e. when Meetme is not involved everything works).
Any suggestions as to what I might look at to try to solve the one-way audio in Meetme gratefully received …
In 1.8 land, you’re looking for dahdi-dummy, that’s part of the DAHDI package, which replaced Zaptel due to trademark concerns.
Cheers.
[quote=“malcolmd”]In 1.8 land, you’re looking for dahdi-dummy, that’s part of the DAHDI package, which replaced Zaptel due to trademark concerns.
Cheers.[/quote]
I believe I’m using dahdi-dummy - if I wasn’t I wouldn’t have even been able to build Meetme into Asterisk would I?
My problem is that conferencing through Meetme works for the external participants, and (one-way only) for the internal participant. I think that says that a timing source isn’t the problem (but I could be entirely wrong!) The only problem I see is the one-way only for the internal participant.
Having fiddled with this some more, I now think the problem was that the internal phone was connecting to the meeting room in admin mode (exten => 8001,1,Meetme(1234,aciMs)) whilst the external participants were connecting in normal mode (exten => 3,1,Meetme(1234,ciMs)).
If the “a” is present in the Meetme for extension 8001, the behaviour is as in my first posting.
If I remove the “a”, then I get two-way audio on the internal phone and external participoants.
One more thing I’ve noticed whilst checking this, in the “broken” state, the internal phone does not receive the announcements of people enetering and leaving the conference.
Hi David,
I was just wondering if you ever discovered a fix to this? I am having the exact same problem now.
Thanks,
Casey.
[quote=“casey232”]Hi David,
I was just wondering if you ever discovered a fix to this? I am having the exact same problem now.
Thanks,
Casey.[/quote]
I got no further than my posting of 19 January - removing the “a” for admin mode from the exten definition for the internal phone gives me a working conference. I don’t think I have an administrator, but for my purposes that doesn’t matter.