Hi Luke,

First of all thank you very much for your test efforts! You are helping so much cleaning Switchfin!

Let me answer the questions I know: 
>1) is there any way to tell from gui or CLI which switchfin build I am running?
In the 'System Info' you get the Firmware/GUI version and the date it was built
'Firmware version: UI2.17 -- Sat Jun 5 20:03:14 EEST 2010'
from the linux shell you can do cat/etc/versions

> 4) I guess similar to above question:  How do I park a call (place on hold with music) from an analog station?  is it possible?
Well most of the call features including call parking needs extra parameters in the Dial command. We use Dial without any parameters.
If you do need call parking you need in your dialplan something like (t to enable dtmf detection during the conversation and enable transferring)
exten => 6000,1,dial(dahdi/1,,t)

>7) since several builds ago (maybe since oslec??) now callerid CID data name/# does not appear any more on my FXS ports.  I checked the CDR logs and the CID info is definitely coming in OK to asterisk, and also to SIP phones but not to the FXS ports.  any idea?
Hmmm. Well I just tried without Oslec and still no CID. Are you sure that CID was working with FXS few versions ago?

>8) for voice menus, does switchfin support asterisk native .sln format?  because I created one according to the instructions at voip-info.org, and when I uploaded it and try to play it back, I receive this from asterisk CLI:
The sln format is included in the Asterisk used by the current version of Switchfin. 'core show file formats' shows the supported formats.
Do you have /persistent/sounds/record/test1.sln ?

Best Regards
Dimitar

Quick Note - To enable recording "on the fly", edit features.conf and add
automon = *1

In extensions.conf context [macro-trunkdial-failover-0.3]
change exten = 1-dial,1,Dial(${ARG1}) to exten = 1-dial,1,Dial(${ARG1},,wW)

now during a call you should be able to dial *1 to start recording, you may have some issues viewing the recording in the recording manager due to the differences in filenames.

I will look into adding this as a feature in the GUI 4.0 this weekend.

Well most of the call features including call parking needs extra parameters in the Dial command. We use Dial without any parameters.
If you do need call parking you need in your dialplan something like (t to enable dtmf detection during the conversation and enable transferring)
exten => 6000,1,dial(dahdi/1,,t)

since several builds ago (maybe since oslec??) now callerid CID data name/# does not appear any more on my FXS ports.  I checked the CDR logs and the CID info is definitely coming in OK to asterisk, and also to SIP phones but not to the FXS ports.  any idea?
Hmmm. Well I just tried without Oslec and still no CID. Are you sure that CID was working with FXS few versions ago?

for voice menus, does switchfin support asterisk native .sln format?  because I created one .....  'core show file formats' shows the supported formats.

IP04*CLI> core show file formats

Format     Name       Extensions          

ilbc       iLBC       ilbc                

gsm        wav49      WAV|wav49           

g729       g729       g729                

g722       g722       g722                

ulaw       au         au                  

alaw       alaw       alaw|al             

ulaw       pcm        pcm|ulaw|ul|mu      

slin       sln        sln|raw             

gsm        gsm        gsm                 

slin       wav        wav                 

10 file formats registered.

Do you have /persistent/sounds/record/test1.sln ?

to fix call parking you can add kK the dial cmd in
extensions.conf [macro-trunkdial-failover-0.3]

Thank you, I will keep an eye on the trunk and maybe compile a new build over the weekend if there are changes.  I may wait for Dimitar's comments on getting CID working with the FXS ports first.

this is great!  thank you again, I will compile build 207 later tonight and report back.

Do you think there is anything I can do to enable caller ID function again on my FXS port or is this something related to the switch from Zaptel to DAHDI?  (i think that was the major difference between atcom firmware and switchfin........ no?

Yes that is true, there is no problem with the outbound caller-id.

I tested two different ways, first with an incoming call rule that send all calls directly to my analog extension (6000) and then also tested via a Voice Menu and then user pressing a key to transfer to my extension.  Neither of them pass any CID to the analog station.

One Touch Recording (on the fly recording) is now possible in the GUI 4.0

The settings can be found in System Setup>Call Features
Add in your desired feature code, then make sure you have either x or/and X enabled in the Dial Options.

hi Dimitar,
wow this is a great news.
I have compiled svn revision 216 and upgraded my IP02.
I am still not receiving caller-ID to my analog extension-  do I need to reset my config files in order to get this fix?

hi Dimitar, well before I built svn 216 I completely deleted my source folder and re-downloaded everything from scratch, so that was clean.  Maybe I will cold-boot my IP02, do you think that may help?

Hi Luke

Our newest member of the team Mike suggested that in some countries ring tones effect the callerid being passed over FXS so we have added the Send CallerID After in the hardware config of the GUI try setting this to 2 and see if it helps.

Thanks
Jason

Hi Dimitar,
yes in latest svn's the Caller ID is working fine for me, thanks again for your help on this!