don't show status DAHDI after upgrade to Asterisk 1.8

Eugene

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

nicolas

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

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

Eugene

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:

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.
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.

nicolas

Hi,

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?

Eugene

deleted

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.

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don't show status DAHDI after upgrade to Asterisk 1.8