Hi!
I'm upgrade Asterisk 1.4 to 1.8 and DAHDI E1 Channels status is disappeared from FOP2.
DAHDI channels now shows in CLI as DAHDI/iNN/XXXXXX-YY, where is

• NN a channel number
• XXXXXX a calling or called number
• YY - some number

The autoconfig-buttons-freepbx.sh output is:

[DAHDI/g0]
type=trunk
label=E1 channels
queuecontext=from-queue
extenvoicemail=*
email=splitme-1-30
channel=DAHDI/1
channel=DAHDI/2
channel=DAHDI/3/
channel=DAHDI/4/
...

I edit autoconfig-buttons-freepbx.sh as:

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

and even as:

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

but no luck :(
I'w view records like this

127.0.0.1       <- Event: Newexten
127.0.0.1       <- Privilege: dialplan,all
127.0.0.1       <- Channel: DAHDI/i1/100-3a
127.0.0.1       <- Context: macro-dial-one
127.0.0.1       <- Extension: s
127.0.0.1       <- Priority: 37
127.0.0.1       <- Application: Dial
127.0.0.1       <- AppData: SIP/6030,15,Ttr
127.0.0.1       <- Uniqueid: 1290076717.150
127.0.0.1       <- Server: 0

but it not affected on FOP2 page

Update: I try this trick

[DAHDI/g0]
type=trunk
label=E1-LDK
queuecontext=from-queue
extenvoicemail=*
email=splitme-1-30
channel=DAHDI/i1/100
channel=DAHDI/i2/100
channel=DAHDI/i3/100

100 - is the one of the extensions behind the E1 trunk
And now FOP2 shows me calls to/from that number only :)

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

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.

deleted

Then I see no activity on DAHDI Button
autoconfig-buttons-freepbx.sh output is [code]
...
[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
...
[/code]
fop2_server debug is(call from Dahdi/i1/100 to SIP/6030):[code]
...
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' }
[/code]
'btn': '15@GENERAL' - is SIP/6030

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.

done

Yes, it works, but for SIP->Dahdi calls only. There is no indication for DAHDI->SIP call until it has been transfered to voicemail. I see call indication within voicemail prompt.
Also, I think "DAHDI/i$ZAPNUM" hardcoded in autoconfig-freepbx-buttons.sh is not suitable for a mixed installation of analog and digital DAHDI cards.

....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

.... 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]

OK! It's not a bug, it's a feature.

Thank you, Nicolas!