Has anyone seen this strange behavior?
I have join/leave enabled, but the conference nevers asks to “say your name”.
I am running freepbx 2.8.1.4.
On asterisk 1.4.37 and 1.8.4, it does not work, but on a asterisk 1.6.2.17, it does prompt.
I have a ticket open - http://www.freepbx.org/trac/ticket/5158
Take a look at this article, it may help install it for you.
was asterisk stopped and dahdi running when you did it?
I am surprised conferences work at all without a timing module. It is required to mix audio.
I had not run a machine without hardware DAHDI and large conferences and did not realize this fact until OTTS training last week. Once I updated all of my machines to the DAHDI Dummy they conferences all take names and sound much better.
I have done that - it still doesnt show pseudo under ‘dahdi show channels’
you have it commented out ;
What is loading the pseudo interface?
Just run dahdi_genconf -vvvv then restart asterisk
Yes, you must have it loaded in Asterisk…
=========================================================================
Connected to Asterisk 1.4.26.2 currently running on vg2 (pid = 1914)
vg2*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret
pseudo default default
vg2*CLI>
vg2*CLI>
[vg2.vi1.net asterisk]# cat dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Sun May 1 00:51:10 2011
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;
; Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: Linux26) 1" (MASTER)
[vg2..net asterisk]# pwd
/etc/asterisk
[vg2..net asterisk]#timing to ask for the caller name??
the conference works fine, 10 callers sit in it for an hour - it just does not ask to record the callers name.
and dahdi_dummy is in dev:
ls -l /dev/dahdi/pseudo
crw-rw---- 1 asterisk asterisk 196, 255 Mar 29 12:37 /dev/dahdi/pseudobut it is not loaded in asterisk.
I have tried modprobing dahdi_dummy, but it still doesnt show in asterisk after a full restart.
You need DAHDI running to provide timing. You must have the DAHDI dummy channel loaded if you don’t have any physical devices.
That guide is exactly how I do my installs, but that is only the compile process, it doesnt mention anything about config settings that actually make the pseudo interface be seen by asterisk, besides that, there is now res_timing_pthread.so if you are using 1.8, which eliminates the need for dahdi if you use confcall app instead of meetme (i wonder if confcall is supported by FreePBX).
Really I would like to know what in the config makes the pseudo interface load in asterisk - I have several systems from 1.4x to 1.8x.
I have a 1.6.2 system (real box) that has a working pseudo interface in asterisk - after tar’ing the configs up and dumping them on a 1.8.4 Xen system, it also loads pseudo in asterisk - but grep’ing the configs for ‘dummy’, ‘pseudo’ or ‘dahdi’ does not show any results as to the pertinent setting that makes it come to life. Dropping the same EXACT configs on another “real box” with 1.8.4 = no pseudo in asterisk.
Here is what I am tar’ing:
tar czvf /etc/asterisk /etc/amportal.conf /etc/dahdi /var/lib/asterisk /var/spool/asterisk /var/lib/mysql /var/www/html/adminSo - there is no consistency in any of this, I can get it to work on a 1.8 system, but not another 1.8 system - it it was a matter of just upgrading to 1.6 or 1.8 I would do this, but there is no consistency.
I think you are confusing people with these statements. Yes you need dahdi running because meet-me needs it as a timing source. You do not need any hardware. Dahdi dummy is good enough.
If you do not have dahdi running conferencing will not work at all. I don’t think you can even dial in. Or you just get silence.
The problem this person is having is the same problem I am having. The Join/Leave record/announce feature is not working but conferencing is. So I doubt dahdi has anything to do with it.
just to update any watchers - this is another example of digium pulling the rug on us - https://issues.asterisk.org/view.php?id=17959 says that the core of dahdi supposedly handles all timing now, and dahdi_dummy has been removed, the documentation even backs this up - https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces BUT at the bottom it gives an entirely contradicting disclaimer that meetme is definitely dependent on dahdi_dummy.
Furthermore - dev figured dahdi was so smart, and users were so stupid, they decided that dahdi should decide the source of timing, and users should not have any method of controlling this - now we drive cars without steering wheels!
So we have asterisk revision 1.8.catch22, and digium dev blindly steaming forward.
Thanks to mustardman who tested this out and found out why it was not working. This is what he wrote in ticket #5158:
that makes no sense - its not var/lib - its /usr/lib. and that would assume someone did not compile dahdi tools (as mustardman noted) - I have compiled dahdi tools (complete). libtonezone.so is already in /usr/lib -
chan_dahdi is compiled - but that doesnt even matter - check the 2nd pbx below which DOES have pseudo loaded - is it loaded under chan_dahdi??:
Connected to Asterisk 1.8.4 currently running on pbx1 (pid = 6523)
pbx1*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pbx1*CLI> module show like dahdi
Module Description Use Count
res_timing_dahdi.so DAHDI Timing Interface 1
chan_dahdi.so DAHDI Telephony Driver w/PRI 0
codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0
app_dahdibarge.so Barge in on DAHDI channel application 0
app_dahdiras.so DAHDI ISDN Remote Access Server 0
5 modules loadedbut this one works and its even a xen vm:
Connected to Asterisk 1.8.4 currently running on pbx6 (pid = 4685)
pbx6*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
pbx6*CLI> module show like dahdi
Module Description Use Count
res_timing_dahdi.so DAHDI Timing Interface 2
chan_dahdi.so DAHDI Telephony Driver w/PRI 0
codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0
app_dahdibarge.so Barge in on DAHDI channel application 0
app_dahdiras.so DAHDI ISDN Remote Access Server 0
5 modules loadedyet another:
Connected to Asterisk 1.6.2.17.2 currently running on pbx2 (pid = 4448)
pbx2*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
pbx1*CLI> module show like dahdi
Module Description Use Count
res_timing_dahdi.so DAHDI Timing Interface 1
chan_dahdi.so DAHDI Telephony Driver w/PRI 0
app_dahdiras.so DAHDI ISDN Remote Access Server 0
codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0
app_dahdiscan.so Scan DAHDI channels application 0
app_dahdibarge.so Barge in on DAHDI channel application 0
6 modules loadednote there chan_dahdi.so is loaded, but never has a channel in use, while res_timing_dahdi loads the pseudo channel. there is no solution there, and mustardman has shown no reproduce-able solution as I too run openvz, but every single one of my vz containers is forced to use dahdi by the host, and dahdi gets compiled against the host and the guest and they still do not consistently have pseudo loaded in asterisk (some do, some dont, but all have /usr/lib/libtonezone.so - so this has nothing to do with it).