I am now showing the channels, but it is not showing up properly.

Eg. I have a trunk which uses channels 1-47 ( private PRI ) and a trunk that uses channels 49-71 ( PRI from telco ).

All channels appear to be coming in via the 1-47 trunk on the FOP2 panel now.

EDIT: Upon further review it seems that Asterisk is not reporting the proper DAHDI channel in use. I make a call out which goes out via DAHDI channel 49, however it is reporting it under DAHDI/i3.

I can only figure that Asterisk is grouping the trunks via an interface number.

Due to this, I changed in FOP2 buttons to use 1-2 for the first two trunk lines ( two links spanned on one group ) and 3 for the last.

This does not appear to work ( one of the trunk buttons light up but does not associate properly to the correct trunk ).

EDIT: To single out the line in FOP2admin - Buttons, you need to set it to 3-3, otherwise it will generate the lines as if going from 1 to 3. This appears to have resolved the issue.

I added the line in php.ini max_input_vars = 2500 so that I could post the buttons form properly as the newer version of PHP only permits up to 1000 by default.

I ran the autoconfig-buttons-freepbx.sh and the DAHDI lines do not appear to be showing up correctly still.

Is there a setting I am missing in getting this back working?

I set up much extensions in elastix as DAHDI devices, such as "DAHDI/i9/10" etc, but corresponding buttons don't show their status, when I make call to or from extensions. However status of DAHDI trunks works well, that is, I see channels and phone numbers on trunks.

Thank you, Nicolas!

Trunk buttons only are displayed when they are bridged. That is normal behavior in fop2.

.... Be sure to have the "all" permission in both read/write in /etc/asterisk/manager.conf...

read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate

...Another issue that I see is the dial extension with the g0/ prefix. Not that is horrible, but that behavior is new, and I think is not good, as you dial a number, the g0 is used by the dial command only...

This issue does not affect the display of channel status button. Вut I don't see channel status while DAHDI-->SIP is ringing.
**DAHDI/i1/100 ringing to SIP/6030 screenshot
[url=http://www.flickr.com/photos/56594160@N02/5227634625/:ex7qglin]*http://farm6.static.flickr.com/5124/5227634625_0854707b1a_z.jpg*[/url:ex7qglin]

And now call is answered**
[url=http://www.flickr.com/photos/56594160@N02/5227632049/:ex7qglin]*http://farm6.static.flickr.com/5288/5227632049_e8f514956e_z.jpg*[/url:ex7qglin]

....If it works of some calls, it should work for others as long as you have DAHDI/ix as channel names. In any case send me the debug output as before for the other type of calls, look at the device names there.

ОК, I will send it soon. There is no indications while DAHDI/iN/YYY rings to SIP extension. The indication is appears when call is timed out and transfered to voicemail system.
** *EDIT: The indication is appears when I pickup the phone too, but not while phone ringing. **

And to make it clear to other users, this happens with CUSTOM DAHDI TRUNKS for ISDN, not for regular DAHDI usage... right?

I think it's not a CUSTOM DAHDI TRUNKS for ISDN, it's STANDARD DAHDI TRUNKS for ISDN (DIGIUM Wildcard TE122). But new version of DAHDI and LibPRI used in Asterisk 1.8 was changed for best q931 support. Therefore, such a system of naming ISDN channels is standard for all installations of Asterisk 1.8

And to make it clear to other users, this happens with CUSTOM DAHDI TRUNKS for ISDN, not for regular DAHDI usage... right?

Hi,

Can you make some calls while capturing output from fop2_server -X 1 and then send the capture privately to me, together with the button configuration for your trunk, extensions involved?

Best regards

Yes, but I don't found where I can attach files.

Can you make some calls while capturing output from fop2_server -X 1 and then send the capture privately to me, together with the button configuration for your trunk, extensions involved? Or you can try to catch me on the live help and provide ssh access to your server so I can take a look at it?

Best regards

...
[DAHDI/g0]
type=trunk
label=E1-LDK
queuecontext=from-queue
email=splitme-1-30
channel=DAHDI/i1
channel=DAHDI/i2
channel=DAHDI/i3
channel=DAHDI/i4
channel=DAHDI/i5
...

fop2_server debug is(call from Dahdi/i1/100 to SIP/6030):

...
127.0.0.1       <- Event: ExtensionStatus
127.0.0.1       <- Privilege: call,all
127.0.0.1       <- Exten: 6030
127.0.0.1       <- Context: ext-local
127.0.0.1       <- Hint: SIP/6030
127.0.0.1       <- Status: 8
127.0.0.1       <- Server: 0

127.0.0.1       <- Event: Dial
127.0.0.1       <- Privilege: call,all
127.0.0.1       <- SubEvent: Begin
127.0.0.1       <- Channel: DAHDI/i1/100-123
127.0.0.1       <- Destination: SIP/6030-00000220
127.0.0.1       <- CallerIDNum: 100
127.0.0.1       <- CallerIDName: <unknown>
127.0.0.1       <- UniqueID: 1290686402.855
127.0.0.1       <- DestUniqueID: 1290686402.856
127.0.0.1       <- Dialstring: 6030
127.0.0.1       <- Server: 0

127.0.0.1       <- Event: NewCallerid
127.0.0.1       <- Privilege: call,all
127.0.0.1       <- Channel: SIP/6030-00000220
127.0.0.1       <- CallerIDNum: 6030
127.0.0.1       <- CallerIDName: 
127.0.0.1       <- Uniqueid: 1290686402.856
127.0.0.1       <- CID-CallingPres: 0 (Presentation Allowed, Not Screened)
127.0.0.1       <- Server: 0

192.168.50.95   => { 'btn': '15@GENERAL', 'cmd': 'settext', 'data': '100 <unknown>', 'slot': '1' }

192.168.50.95   => { 'btn': '15@GENERAL', 'cmd': 'direction', 'data': 'inbound', 'slot': '1' }

127.0.0.1       <- Event: Newstate
127.0.0.1       <- Privilege: call,all
127.0.0.1       <- Channel: SIP/6030-00000220
127.0.0.1       <- ChannelState: 5
127.0.0.1       <- ChannelStateDesc: Ringing
127.0.0.1       <- CallerIDNum: 6030
127.0.0.1       <- CallerIDName: 
127.0.0.1       <- Uniqueid: 1290686402.856
127.0.0.1       <- Server: 0

127.0.0.1       <- Event: VarSet
127.0.0.1       <- Privilege: dialplan,all
127.0.0.1       <- Channel: DAHDI/i1/100-123
127.0.0.1       <- Variable: ~HASH~SIP_CAUSE~SIP/6030-00000220~
127.0.0.1       <- Value: SIP 180 Ringing
127.0.0.1       <- Uniqueid: 1290686402.855
127.0.0.1       <- Server: 0

192.168.50.95   => { 'btn': '15@GENERAL', 'cmd': 'settimer', 'data': '0@UP', 'slot': '1' }

192.168.50.95   => { 'btn': '15@GENERAL', 'cmd': 'state', 'data': 'RINGING', 'slot': '1' }

192.168.50.95   => { 'btn': '15@GENERAL', 'cmd': 'clidnum', 'data': 'MTAw', 'slot': '1' }

192.168.50.95   => { 'btn': '15@GENERAL', 'cmd': 'clidname', 'data': 'PHVua25vd24+', 'slot': '1' }

192.168.50.95   => { 'btn': '15@GENERAL', 'cmd': 'notifyringing', 'data': '1', 'slot': '' }

192.168.50.95   => { 'btn': '15@GENERAL', 'cmd': 'settext', 'data': '100 <unknown>', 'slot': '1' }

'btn': '15@GENERAL' - is SIP/6030

In FOP 2.20 language is set only in presence.js or via the individual preferences, there is no server side language anymore (no op_lang_xx.cfg) nor the need to edit the index.html file. But you still have the language for the phonebook , and you must set it in config.php.

The autoconfiguration script will not output the i, you must modify it by hand. I do not know how to test for the type, so if you use it, it is pretty easy to modify the script anyways.

But the main question is, does it work? Do you see dahdi status?

Are you using centos5, 32 bits? If so, download fop 2.20 beta and try again, the output from the autoconfig should be:

channel=DAHDI/i1
channel=DAHDI/i2

Thanks. Result is:

1. I'm lost fop2 homemade russian translation, the op_lang_ru.cfg and lang_ru.js is not working, language setting is disappears from index.html. The 'ru' setting in Preferences box and fop2.cfg Global Config is not affected too.
2. The

echo "channel=DAHDI/i$ZAPNUM"
in autoconfig_buttons_freepbx.sh is useful for ISDN channels only. Additional FXS/FXO Dahdi channels named as usual (DAHDI/N). The FOP2 must check the channel type and the add "i" or not, depends of that type, I think.

channel=DAHDI/i1
channel=DAHDI/i2

If you are using any other distro, let me know so I can create the tarball for you.

It is not that useful if you see only one particular dialed number right?

What is the dahdi name exactly?

DAHDI/i{CHANNELNUMBER}/{DIALEDNUMBER}:{SUBADDRESS}-{SESSION} ?

I can't see :{SUBADDRESS}. I see DAHDI/i{CHANNELNUMBER}/{DIALEDNUMBER or DIALINGNUMBER (depends on call direction)}-{SESSION}

Do you want to do some beta testing for this?

Yes, sure I want

What is the dahdi name exactly?

DAHDI/i{CHANNELNUMBER}/{DIALEDNUMBER}:{SUBADDRESS}-{SESSION} ?

To make it work I will have to add support for this format in FOP2, as right now what FOP2 does is to strip only the -{SESSION} suffix and use the first part as the device name.

Do you want to do some beta testing for this?

Best regards