don't show status DAHDI after upgrade to Asterisk 1.8

Eugene

Then I see no activity on DAHDI Button
autoconfig-buttons-freepbx.sh output is

...
[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

nicolas

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? 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

Eugene

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.

nicolas

Send them via email to nicolas at house dot com dot ar.

Eugene

done

nicolas

Thanks, I found the issue. I will update the beta later tonight

Eugene

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.

nicolas

For a Mixed installation you can modify the script accordingly. 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.

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

Eugene

....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

nicolas

I talked about Custom because in the logs there is a mention of a custom trunk in freepbx. Be sure to have the "all" permission in both read/write in /etc/asterisk/manager.conf. Some asterisk versions filter out events event if you have the individual permission set (I think it is a bug in trixbox only, but maybe it is more extended than that). 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.

Eugene

.... 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]

nicolas

Hi Eugene,

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

Eugene

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

Thank you, Nicolas!

RekGRpth

Sorry for upping this theme, but I have trouble with status of extensions on DAHDI.

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.

nicolas

Check how are your device names are being set by the fop2admin module ( /usr/local/fop2/autoconfig-buttons-freepbx.sh ), and check those names with the active channels for your extensions when you have ongoing calls ( asterisk -rx "core show channels"). If they do not match, you won't see activity.

rcanavan

I have upgraded from Asterisk 1.6 to Asterisk 1.8 and found that the DAHDI channels do not show anymore as well. I have upgraded to FOP2 2.25 with the admin 1.2.10. When running autoconfig-buttons-freepbx.sh I get the DAHDI channels coming back without the i.

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

rcanavan

FYI - I removed the FOP2admin module and reinstalled ( 1.2.10 ).

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.

rcanavan

Found this post: View Post 1208

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.

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don't show status DAHDI after upgrade to Asterisk 1.8