More than 10 years ago I built an Asterisk pbx to run my home phone collection It consists of a Pentium 4 based computer with a 4 port analog FXO card and a single TE110p T1 PCI card connected to an Adtran TA850 channel bank with 23 analog extensions. Failing capacitors destroyed the motherboard so I built another system out of an old Pentium d computer, Installed freepbx distro instead of trying to roll my own Asterisk system again- forgot how to do it anyway. The result? I cannot get access to the pots lines through the FXO ports and the channel bank does not give dial tone or any of its extensions are active. There is no alarm on the T1 card - it has a green light. However, in the asterisk log I get this error.
[2018-12-24 01:05:15] NOTICE[2346] sig_pri.c: pri_check_event returned error 500 (Unknown error 500)
[2018-12-24 01:05:15] NOTICE[2346] chan_dahdi.c: Got DAHDI event: HDLC Abort (6) on D-channel of span 1
Not sure where to start, I remember there was a command line test for the Zaptel interfaces and have seen some info on a Dahdi tools command. Don’t know if something could have gotten changed in the Adtran itself but I plugged in a Carrier Access channel bank and had the same result.
zaptel is now dahdi and everything should ‘map’ over.
I don’t believe The Carrier Access I supports either PRI’s (or far end disconnect detection.) You need to identify if you have the CAI or CAll . . .
As to your FXO’s with digium hardware, they should just work (if you follow the rules - ) Maybe check that if you have both analog and TDM channels that they are mapped appropriately
Right now I have the Adtran TA850 connected the Adit 600 is just a spare I picked up and had to play with to clear the password. I originally used a non-GUI installation but I noticed that with the Freepbx distro the config files are broken up into those set by the system and those that can be manually edited. Trying to get the right settings from the GUI is where I get confused. Even setting up my sip and iax extensions took a little trial and error. I use the channel bank because sip ata’s don’t support pulse dialing and most of my phones are rotary dial except for the payphones. The channel bank is also necessary because the extensions call back to the modem and computer which configures the protel payphone collection. Is there any sample config files for a setup like mine?
Yes, the GUI will IMHO often override any ‘a little bit different’ config, feel free to disable or even uninstall it and go with the old fashioned way that ‘worked’ and likely still will
I am hoping to figure the new way out the gui seems to have some nice features. Maybe there is some place the PRI interfacing with Asterisk is explained, I have no idea how it translates through the te100p to the channel bank.
If is the word. I forgot all about how I set this up originally. There was some info on a similar setup on the old Asterisknow forums, but I cannot seem to find it anymore. At one time I understood the Zaptel hardware and Asterisk but it took weeks of trial and error before i got it working.
/etc/zaptel.conf is now /etc/dahdi/system.conf (copy it over from the old system)
and
/etc/asterisk/zapata.conf is now /etc/asterisk/chan_dahdi.conf (copy it over from the old system)
should get you there.
If you don’t have your old system then if you are not using extraordinary hardware or expecting extraordinary behaviour, then the FreePBX’ “dahdi helper” module will likely find it.
) Maybe check that if you have both analog and TDM channels that they are mapped appropriately