I beat my head against the wall for a couple hours the other day because when the system updated to 2.5.1.5 and I added an analog extension the chan_dahdi_additional.conf file was parsed and all the “signalling” commands were spelled with one l. This did not sit well with the Asterisk 1.4.24.1 DAHDI and Sangoma drivers I had on 2 systems causing no dial tone.
You said:I added an analog extension the chan_dahdi_additional.conf file"
Please do NOT edit files without knowing what you are doing. In general you can not and SHOULD NOT edit any files if the name does not contain the word custom in it. see http://freepbx.org/configuration_files for more info.
More to the point, the file chan_dahdi_additional.conf is a FreePBX generated file so that the next time you make any change it will go away and be replaced only with the things that FreePBX knows about in it’s databases.
Part of the reason it didn’t sit well is that there are things that need to be entered into the internal asterisk database (Berkley db) when a extension is created for it to work. So please use the system the proper way and add the extension via the GUI so that everything and place is updated correctly. At the same time that’s the purpose of the GUI it checks for things like that mistake so it would not allow it to happen.
Please clarify. The GUI creates the file chan_dahdi_additional.conf which has misspelled “signaling” which in turn causes DAHDI and Sangoma drivers not to recognize the analog extensions thereby not configuring them for use. What do you suggest as the fix? I simply spelled “signalling” with two l’s and resolved the issue. I know the files are re-written every time an extension is added. And every time I have to go change the spelling or it the same problem occurs.
No that I do not.
I read your inital message as meaning “you” edited the file chan_dahdi_additional.conf. Which would be the wrong way to go about things.
You don’t say what version of FreePBX you are using so first off please make every attempt to update to the latest code.
I have the latest FreePBX code loaded (2.5.1 branch) and while I don’t have Dadhi configured on my system, I did go searching and I can’t find it mis-spelled anyplace in the code, so I’m assuming it was fixed.
Yeah, I re-read my initial post and understand why you thought I had manually changed the chan_dahdi_additional.conf file. I am running 2.5.1 as well and recently the modules were updated and the framework is now 2.5.1.5
I did the module update on Tuesday and that is when the dial tone quit working. I have two test systems I utilize for custom development to test our analog phones and one had Digium 2400 series card and one has Sangoma 400 series cards. Both quit working and Sangoma support could not figure it out so I reloaded asterisknow 1.5 and did the updates, uploaded all modules, and recompiled the drivers etc. with no success at getting the dial tone to work.
Then I contacted a guru I know and he actually answered the phone for once and after about 1/2 hour in my system called me back and informed me that “signaling” in the chan_dahdi_additional.conf should be spelled with two - l’s. When adding an extension through the GUI the file is written with one l.
No Dial Tone Example:
;;;;;;[300]
signaling=fxo_ks
Dial Tone Works Example:
;;;;;;[300]
signalling=fxo_ks
I know from my Avaya and Nortel day’s they used to spell signaling with two l’s
and I always thought it a telecom thing. Maybe asterisk is expecting that as well. Asterisk version is 1.4.24.1.
Digium confirmed that 1.4.x.x needs 2 l’s in signalling to configure dahdi to work.
Version 1.6.x.x will handle 1,2 or 3 l’s to eliminate the issue.
In order to get Caller ID to work on analog extensions with DAHDI the chan_dahdi.conf file needs to have the automatically generated configurations under [channels] eliminated and add simply:
[channels]
#include chan_dahdi_additional.conf
Then the extra l needs to be added to the automatically generated FreePBX configurations in chan_dahdi_additional.conf
I was going to tell you to file a bug report for it but it’s been pointed out that it is actually a specific error tied to asterisk 1.4.24.1. I’m sure Philippe will chime in shortly but if you are using 1.4.22 and above (thus heed Dadhi) it is highly recommended that you stay at 1.4.22 and NOT use 1.4.24.x as that whole sub series has big issues currently…