Preparing for 2.0 Release Candidate 1

[Michael]

I'm quite close to getting another RC prepared. Thanks to everyone for the feedback, testing and bug fixing! Here is a developer snapshot of things so far. With all of the newly added hardware support, perhaps this will bring in even more reports (easier to try things out if your cards and hard drive work...).

Developer Snapshot r1390:
 - http://downloads.askozia.com/pbx/snapshots/r1390

Changes since RC1 (r1365)
 - added support for many single-port Analog and single-port ISDN cards
 - added a Dialplan Application which reads back the system IP
 - many new countries indication tones supported
 - added support for SATA controllers
 - increased support for multiple-port ISDN cards
 - increased support for IDE controllers
 - fixed missed-call notifications
 - fixed external phones which use Analog or ISDN Providers
 - several small fixes to call groups
 - more robust analog module detection (by Giovanni Vallesi)
 - serial console settings fixed (by Stephane Billiart)
 - COMpact 3000 : Analog ports now supported
 - COMpact 3000 : ISDN configuration generation fixed
 - COMpact 3000 : upgrading firmware via the WebGUI now much safer

[supergus]

Hi Michael,

I've installed the r1390 image and found following strange behaviour :
I just add one phone and while adding it to an existing group I remark that all the phone drag&dropable list were mixed up :

The name of phone doesn't correspond anymore with the extension number. It seems that the name is correct but not the extension. This is true only in the group "Dialplan: Call Groups: Edit "

BR.

[Michael]

Thanks for the report.

That is something on my list for today. I didn't get to it for this snapshot. Call groups are not behaving well. I had a report of this awhile back and confirmed it yesterday myself.

[helvetic]

Hello I just Upgraded my Deciso ISDN box with the RC1 Firmware. Since then my DISA Application isn't working anymore. The Call is answered and just hung up same moment. The same application was working great in V1.03.

Code:

1,Answer()
2,Authenticate(123456)
3,DISA(no-password,SIP-PHONE-8686558934b85d7d2100bf)
4,Hungup()

[helvetic]

Some Feedback about Outgoing Caller ID for ISDN, the caller ID Options "Send string defined above" form the Provider Page is ignored, for all calls the Main MSN is used (not the one set in the field "main number", the main MSN defined by Swisscom), this worked in V1.0.3. However I never got the Option "Send phones' Caller ID" working, not in V1 neither in V2.

My setup:

Deciso 4 Port ISDN box, Swisscom ISDN with MSN

Please let me know if you need more information for troubleshooting, I can also give remote access, if this will help.

Update: Just Discovered another Topic with same Problem,
http://forums.askozia.com/index.php/topic,961.msg3644.html

This solved the Problem with the ignored Outgoing MSN under the Provider Page, however I still cant get the "Send phones' Caller ID" working. Probably this is not an ISDN specif problem, i discovered same behaviour with the VOIP Providers "Send string defined above" is working but "Send phones' Caller ID" is not. I use same format everywhere 'Name <number>'

[Michael]

Hello I just Upgraded my Deciso ISDN box with the RC1 Firmware. Since then my DISA Application isn't working anymore. The Call is answered and just hung up same moment. The same application was working great in V1.03.

This is because only the necessary applications are loaded in 2.0 to cut down on memory usage. You will need to open the Integrator Panel (press 'ESC' anywhere in the GUI). From there you can edit the modules.conf file and override it with "autoload=yes" (see screenshot). That will load all of the Asterisk modules on boot.

[Michael]

This solved the Problem with the ignored Outgoing MSN under the Provider Page, however I still cant get the "Send phones' Caller ID" working. Probably this is not an ISDN specif problem, i discovered same behaviour with the VOIP Providers "Send string defined above" is working but "Send phones' Caller ID" is not. I use same format everywhere 'Name <number>'

Glad that's working! Is this perhaps the same everywhere that the number must be formatted this way? If so, the help text in the GUI should be updated.

Your telephony provider must allow that. By default, AskoziaPBX tries to send the phones' caller id but usually the provider will override it with your 'known' information. With ISDN you need "CLIP no screening" activated on your line so you are allowed to set your own Caller ID information freely.
 - http://www.voip-info.org/wiki/view/ISDN+Features

[Michael]

I've installed the r1390 image and found following strange behaviour :
I just add one phone and while adding it to an existing group I remark that all the phone drag&dropable list were mixed up :

The name of phone doesn't correspond anymore with the extension number. It seems that the name is correct but not the extension. This is true only in the group "Dialplan: Call Groups: Edit "

@supergus : Could you tell me the exact steps needed to recreate this bug? I have tried for more than an hour now to break call groups without success. Something like:
 - added SIP phone 104
 - added SIP phone 105
 - added Call Group 100, added 104 and 105
 - added IAX phone 106
 - added 106 to Call Group 100
 - broken config

Thanks in advance! Others have reported this but I've not yet been able to track it down.

[helvetic]

Hello I just Upgraded my Deciso ISDN box with the RC1 Firmware. Since then my DISA Application isn't working anymore. The Call is answered and just hung up same moment. The same application was working great in V1.03.

This is because only the necessary applications are loaded in 2.0 to cut down on memory usage. You will need to open the Integrator Panel (press 'ESC' anywhere in the GUI). From there you can edit the modules.conf file and override it with "autoload=yes" (see screenshot). That will load all of the Asterisk modules on boot.

Ty for getting back to me, I edited the modules.conf as described in your screenshot and rebooted but still the same problem.

[helvetic]

Your telephony provider must allow that. By default, AskoziaPBX tries to send the phones' caller id but usually the provider will override it with your 'known' information. With ISDN you need "CLIP no screening" activated on your line so you are allowed to set your own Caller ID information freely.
 - http://www.voip-info.org/wiki/view/ISDN+Features

I'm not sure if we talking about the same. Im not trying to use a number outside my MSN Range. On the ISDN Provider Page under Caller ID Options i switch to Send phones' Caller ID and leave the text filed above empty. On the Phone Accounts Page I enter in the Caller ID field Name <MSN-Number>. Maybe its a misunderstanding of this feature on my side, but like this I expect the caller ID is set dynamically and depending on the SIP Account calling out. In other words the Caller ID is then overwritten by value from Caller ID filed on the SIP Phone Account Page.

[Michael]

Ty for getting back to me, I edited the modules.conf as described in your screenshot and rebooted but still the same problem.

Could you post some log output when this fails?

I'm not sure if we talking about the same. Im not trying to use a number outside my MSN Range. On the ISDN Provider Page under Caller ID Options i switch to Send phones' Caller ID and leave the text filed above empty. On the Phone Accounts Page I enter in the Caller ID field Name <MSN-Number>. Maybe its a misunderstanding of this feature on my side, but like this I expect the caller ID is set dynamically and depending on the SIP Account calling out. In other words the Caller ID is then overwritten by value from Caller ID filed on the SIP Phone Account Page.

That is what should be happening but your provider has to accept what you have put in there. Do you need to put in your entire number? With or without region code? Only the msn? These are things I'm not sure of...

[helvetic]

That is what should be happening but your provider has to accept what you have put in there. Do you need to put in your entire number? With or without region code? Only the msn? These are things I'm not sure of...

I can set every of my MSN under the ISDN Provider section Caller ID section -> Outgoing Presentation -> Send string defined above. I test with every MSN within my Number Range if entered with regionalcode and without the leading Zero (= last nine digits of the Number) its working when set here under the provider page but not with the MSN entered in same format and under the Phone Account Page.

I used same format Name <last-nine-digits-of-MSN>

[helvetic]

Could you post some log output when this fails?

Thank you again with your help here

Code:

Feb 25 18:23:19 asterisk[1045]: WARNING[1098]: file.c:650 in ast_openstream_full: File agent-pass does not exist in any format
Feb 25 18:23:19 asterisk[1045]: WARNING[1098]: file.c:953 in ast_streamfile: Unable to open agent-pass (format 0x8 (alaw)): No such file or directory
Feb 25 18:23:19 asterisk[1045]: WARNING[1098]: file.c:650 in ast_openstream_full: File vm-goodbye does not exist in any format
Feb 25 18:23:19 asterisk[1045]: WARNING[1098]: file.c:953 in ast_streamfile: Unable to open vm-goodbye (format 0x8 (alaw)): No such file or directory

The DISA Application

Code:

1,Answer()
2,Authenticate(123456)
3,DISA(no-password,SIP-PHONE-8686558934b85d7d2100bf)
4,Hungup()

[Michael]

I can set every of my MSN under the ISDN Provider section Caller ID section -> Outgoing Presentation -> Send string defined above. I test with every MSN within my Number Range if entered with regionalcode and without the leading Zero (= last nine digits of the Number) its working when set here under the provider page but not with the MSN entered in same format and under the Phone Account Page.

I used same format Name <last-nine-digits-of-MSN>

That will not work. The internal extension of the phone will be sent as the MSN. This is the current behavior which is not very flexible. Maybe another field should be made available which allows the user to set this phone's caller id when dialing out through a provider? For each provider?

I really hate complexity...if anyone has some straightforward suggestion on this, I'm welcome to it.


Regarding your DISA application. Try setting the channel language. I believe there is a mistake in the way the en -> en-us symlink is created and that is why your application cannot find those two prompts, thus creating an error and exiting.

Code:

1,Answer()
2,Set(CHANNEL(language)=en-us)
3,Authenticate(123456)
4,DISA(no-password,SIP-PHONE-8686558934b85d7d2100bf)
5,Hangup()

[helvetic]

Maybe another field should be made available which allows the user to set this phone's caller id when dialing out through a provider? For each provider?

I really hate complexity...if anyone has some straightforward suggestion on this, I'm welcome to it.

I think that would be best solution for the moment as every provider seems to accept the Caller ID in some specific format only. I made same experience with different VOIP Providers. I personally would like to have that option on a provider base.

Regarding your DISA application. Try setting the channel language. I believe there is a mistake in the way the en -> en-us symlink is created and that is why your application cannot find those two prompts, thus creating an error and exiting.

Thanks allot, that solved my Problem !!! I really appreciate your help