Aastra XML scripts 2.3.0 Beta

[jroper]

Hi

The short answer is that I don't think you should.

t*f*t*p is not a secure protocol, and anyone can go scrabbling around in your t*f*t*p directory to find your endpoint usernames and passwords, with potentially expensive results.

You could consider creating a t*f*t*p-server in house (there is a windows version) and keeping t*f*t*p traffic safely inside your own internal network?

Joe

 

[Speedy2k]

Ok i have done more testing with the CallerID Superfecta add-on, and i don't know why, but it seems to not be working perfectly.
Here is the problem i come into.

If i try the 188002247737 that is the AccesD number here, i got that:

Caisses Desjardins - Centre de Servi ...

That is OK! but as soon as i try to enter a local number, it doesn't seems to work on the phone but it work on the CallerID superfecta Page and even if i try it directly.

try with our old buisiness number:
8195661661

If i try it directly from Google Chrome, i got that:

Denis Chicoine

If i try it in curl i got the same thing, but if i try it on the phone, i got:

(N/A)

I dont know why.

Here is what my asterisk.conf file look like for the Outgoing Caller id::

[Outgoing]
# Lookup application enabled (1) or not(0), default is 1
enabled=1
# External lookup uri, phone adds the number at the end of the uri
# example for superfecta module external=http://maint:[email protected]/admin/modules/superfecta/bin/callerid.php?thenumber=
external=http://maint:[email protected]/admin/modules/superfecta/bin/callerid.php?thenumber=

and if i try the same command in curl:

root@PIAF16BETA:~ $ curl http://maint:[email protected]/admin/modules/superfecta/bin/callerid.php?thenumber=8195661661
Denis Chicoineroot@PIAF16BETA:~ $

So it must be ok, the question i got, is the phone have any cache for is callerID lookup, like if it as the number into a file somewhere will take this one first ??

And the other thing i see, when the call is ringing i see the CallerID Name, but as soon as the call is answered, the display show two time callerID num we have dialed, it would be really nice to be able to pass the CallerID name even when connected. Thanx a lot!

Here is more information:

here is the access log when i try the 8195661661 from the phone:

127.0.0.1 - maint [25/Apr/2010:14:38:04 -0400] "GET /admin/modules/superfecta/bin/callerid.php?thenumber=8195661661 HTTP/1.0" 200 5 "-" "-"
10.1.0.198 - - [25/Apr/2010:14:38:04 -0400] "GET /aastra/asterisk/outgoing.php?number=8195661661 HTTP/1.0" 200 409 "-" "Aastra57i MAC:00-08-5D-11-C5-E2 V:2.5.3.2002-SIP"
10.1.0.198 - - [25/Apr/2010:14:38:05 -0400] "GET /aastra/asterisk/outgoing.php?action=refresh&number=CALLING HTTP/1.0" 200 113 "-" "Aastra57i MAC:00-08-5D-11-C5-E2 V:2.5.3.2002-SIP"
10.1.0.198 - - [25/Apr/2010:14:38:06 -0400] "GET /aastra/asterisk/outgoing.php?action=refresh&number=CALLING HTTP/1.0" 200 113 "-" "Aastra57i MAC:00-08-5D-11-C5-E2 V:2.5.3.2002-SIP"

Why is it refreshing every seconds??

And here is the error log:

[Sun Apr 25 14:38:05 2010] [error] [client 127.0.0.1] PHP Notice:  Undefined offset:  7 in /var/www/html/admin/modules/superfecta/bin/source-Trunk_Provided.php on line 40
[Sun Apr 25 14:38:05 2010] [error] [client 127.0.0.1] PHP Notice:  Undefined index:  Default_Asterisk_Phonebook in /var/www/html/admin/modules/superfecta/bin/callerid.php on line 309
[Sun Apr 25 14:38:05 2010] [error] [client 127.0.0.1] PHP Notice:  Undefined index:  Default_White_Pages in /var/www/html/admin/modules/superfecta/bin/callerid.php on line 309
[Sun Apr 25 14:38:05 2010] [error] [client 127.0.0.1] PHP Notice:  Undefined index:  Default_CanPagesCa in /var/www/html/admin/modules/superfecta/bin/callerid.php on line 309
[Sun Apr 25 14:38:05 2010] [error] [client 127.0.0.1] PHP Notice:  Undefined index:  Default_Yellow_Pages in /var/www/html/admin/modules/superfecta/bin/callerid.php on line 309

 

[mark-hc]

Hi

The short answer is that I don't think you should.

t*f*t*p is not a secure protocol, and anyone can go scrabbling around in your t*f*t*p directory to find your endpoint usernames and passwords, with potentially expensive results.

You could consider creating a t*f*t*p-server in house (there is a windows version) and keeping t*f*t*p traffic safely inside your own internal network?

Joe

The is basically the response I received from Lylix.

In the past I've just ran the scripts on the local PBX box and everything worked out.

If I run the scripts on the Lylix server, do I copy the contents of the t*f*t*p folder to a secure FTP site?

I'm uncertain how to accomplish splitting up the files among remote servers.

 

[gizmosrcool]

The DND button partially works for me. It actually does the function, but it does not show visually if it is active or not. Neither the red light comes on and it does not show "DND Activated" or something similar on the lcd screen.

Also the install does not work for a new install. I had to download and install 2.5.3 firmware and also install php-ldap. I used yum for the php and found the firmware as an rpm on trixbox yum download site.

Thx, T.

 

[aastra1]

We need more details

The DND button partially works for me. It actually does the function, but it does not show visually if it is active or not. Neither the red light comes on and it does not show "DND Activated" or something similar on the lcd screen.

Also the install does not work for a new install. I had to download and install 2.5.3 firmware and also install php-ldap. I used yum for the php and found the firmware as an rpm on trixbox yum download site.

Please give us more details on your configuration, Asterisk version, freePBX version, freePBX mode, freePBX DEVSTATE configuration...
The DND LED is now updated via a manager event collected by one of the aastra-daemons so first make sure that both daemons are running. We have some problems with certain versions of Asterisk as the manager events have changed.

Regards

aastra1

 

[aastra1]

t*f*t*p can probably be replaced by HTTP

The is basically the response I received from Lylix.

In the past I've just ran the scripts on the local PBX box and everything worked out.

If I run the scripts on the Lylix server, do I copy the contents of the t*f*t*p folder to a secure FTP site?

I'm uncertain how to accomplish splitting up the files among remote servers.

I am not familiar with Lylix but if t*f*t*p is an issue, I believe you can switch to HTTP (or HTTPS) for the config files as long as there is a /tftpboot directory that the Apache user (usually asterisk) can read and write.
I will do some tests and publish how to do that, the mDNSResponder allows http, it probably just needs a couple of changes in aastra.cfg.

Will publish ASAP.

Regards

aastra1

 

[Speedy2k]

Please give us more details on your configuration, Asterisk version, freePBX version, freePBX mode, freePBX DEVSTATE configuration...
The DND LED is now updated via a manager event collected by one of the aastra-daemons so first make sure that both daemons are running. We have some problems with certain versions of Asterisk as the manager events have changed.

Regards

aastra1

I have the same problem here, it seems that the LED are not working with the Call Fwd., NPD and the Follow-Me. The Voice Mail one look OK. When i log the phone in and from time to time, i got a Cannot Display Merssage with the Done buttons in the screen.

I have Asterisk 1.6.2.6 with FreePBX 2.7.0.2

The DND key doesn't show anything when pressed!



Another thing i already talk about is the apply_prf script that isn't recreating the language setting, so if you have set-up a phone in french by login in with the phone, when you do an apply_prf on this extension, the NPD key goes back to DND and all the keys are back to english, so in a french system, the apply_prf script isn't usable.

 

[aastra1]

Asterisk 1.6.2.6

I have the same problem here, it seems that the LED are not working with the Call Fwd., NPD and the Follow-Me. The Voice Mail one look OK. When i log the phone in and from time to time, i got a Cannot Display Merssage with the Done buttons in the screen.

I have Asterisk 1.6.2.6 with FreePBX 2.7.0.2

The DND key doesn't show anything when pressed!

To all Asterisk 1.6.2.6 users, I finally found the time to upgrade my PIAF server to this version, the problem comes from the fact that the command used to change the devstate we used in 1.6.0 and 1.6.1 is now obsolete with 1.6.2 so the code has been updated and the fix will be part of Beta3 that I will publish later this week.

Regards

aastra1

 

[Phone_User]

Is there a way to increase the volume when listening to voicemail when using the voice mail app?

Thanks

Rob

 

[edisoninfo]

I installed the latest beta2 scripts on my 57i which has the latest firmware. All seems to work except for the voicemail button. If I press it when there are no messages, it displays the "no messages" fine. If I press it when there is a message, the lower buttons all change to things like fwd, pause, etc., a messages pops up on the phone very very briefly that says "call failed" and the phone starts what sounds like a busy signal. At which time the phone is now completely frozen and a power cycle is the only thing that gets it back. I've watched the httpd error_log file and don't see any errors. What should I look for next?

 

[carlosmp]

aastra1,

I was playing with the latest, and it seems that the demo-device-user.prf still has key 8 as logout, while key 1 is login while there is no user.

I did see it was changed in demo-user, but not in demo-device-user.prf, which is where it's really needed...

 

[edisoninfo]

I installed the latest beta2 scripts on my 57i which has the latest firmware. All seems to work except for the voicemail button. If I press it when there are no messages, it displays the "no messages" fine. If I press it when there is a message, the lower buttons all change to things like fwd, pause, etc., a messages pops up on the phone very very briefly that says "call failed" and the phone starts what sounds like a busy signal. At which time the phone is now completely frozen and a power cycle is the only thing that gets it back. I've watched the httpd error_log file and don't see any errors. What should I look for next?

Is anyone else having problems with the voicemail button on the 57i phones?

 

[Socs28]

I have these installed in 4 locations and it is happening in one. No idea yet what it is.

 

[carlosmp]

A few questions

I was setting up a client with the new scripts, and noticed that a few things don't seem to be working properly. We're using device and user mode and Beta2 of the 2.3 scripts. For the phones we are using 31i and 55i sets across the office.

- DND - The DND light will initially light up if a user presses it, but we are having problems with users wanting to deactivate. The function will not deactivate.

- Follow-me - On the 55i, even though the phone shows deactived, asterisk shows it as disabled, the light is still on. On the 31i, the button says please wait, and does nothing. The light will also not sync with the server setting. On the 55i (and on my 57i) the phone continues to display Follow-me activated.

I was doing a bit of poking/reading and noticed that the action uri sections are not in the demo-device-user.prf file, but they are in the demo-user.prf . The demo-device-nouser.prf file has a single reference.

Is there a reason that these settings are not in the demo-device-user file? It seems to be in demo-user.prf on a per 51/53 basis based on the uri info as per the docs.

 

[vcallaway]

I'm having an issue with voice mail.

Asterisk 1.4.21.2
6757i phone
aastra-xml-scripts-2.3.0-1-Beta2

When I try to listen to a voice mail or record anything it missdials.

The log output shows this:

06:52:05 AM    192.168.1.188    Aastra57i MAC:00-08-5D-23-8C-D4 V:2.5.3.2002-SIP    vmail_4_asterisk    ext=2500, user=2500 ,action=rec_greetings
06:52:06 AM    192.168.1.188    Aastra57i MAC:00-08-5D-23-8C-D4 V:2.5.3.2002-SIP    outgoing_asterisk    action=check, number=vmail

So it looks like it is dialing the word vmail instead of the number.

I tried using the demo-user.prf to see if it was something I did but no difference.

 

[aastra1]

Volume increase

Hi Rob,

Is there a way to increase the volume when listening to voicemail when using the voice mail app?

Unfortunately as the display is "locked" when we play the message no key besides the displayed softkeys are available to the user, this means:

• no switch to handset/headset (unless you are fast)
• no volume control

We lock the display to avoid the user to lose the XML display if the user pressees the wrong keys.

In the new firmware 2.6.0 we just released we made a change on the locking mechanism which now allows access to the telephony keys.

This is implemented in Beta 4 that we will release ASAP.

Regards

Aastra1

 

[aastra1]

Logout key in device/user mode

Hi Carlosmp,

I was playing with the latest, and it seems that the demo-device-user.prf still has key 8 as logout, while key 1 is login while there is no user.

I did see it was changed in demo-user, but not in demo-device-user.prf, which is where it's really needed...

Looks like we forgot the Device/user mode as it is not very common. Fixed in Beta 4 which will be released ASAP, 6730i, 6731i, 53i, 9143i now have logout on key 1.

Regards

Aastra1

 

[aastra1]

Voicemail issues

Hello there,

I'm having an issue with voice mail.

Asterisk 1.4.21.2
6757i phone
aastra-xml-scripts-2.3.0-1-Beta2

When I try to listen to a voice mail or record anything it missdials.

I tried using the demo-user.prf to see if it was something I did but no difference.

It is perfectly normal that the phone dials "vmail" and not a number as it is a custom destination we have added in the Asterisk dialing plan. What is not normal is that you get a "call failed".

Make sure that you have:

• /etc/asterisk/extensions_aastra.conf which includes the aastra-vm section
• /var/lib/asterisk/agi-bin/aastra-vm.php in place as it is called by the dialplan when the phone dials "vmail"

We will also need the traces from Asterisk to understand why the call fails. Can someone publish the Asterisk traces when the phone tries to play a message?

One thing you can do is manually add a speeddial key dialing "vmail" and get the traces.

Also please give us more details on your configuration: Asterisk version, freePBX version, USEDEVSTATE and freePBX mode, that would help us.

Regards

aastra1

 

[aastra1]

Beta4 will fix issues with Asterisk 1.6.2

Hello Speedy2K,

I have the same problem here, it seems that the LED are not working with the Call Fwd., NPD and the Follow-Me. The Voice Mail one look OK. When i log the phone in and from time to time, i got a Cannot Display Merssage with the Done buttons in the screen.
I have Asterisk 1.6.2.6 with FreePBX 2.7.0.2
The DND key doesn't show anything when pressed!

Asterisk 1.6.2 manager interface is not fully compatible with the previous versions... code has been updated and will be in Beta4. We also fixed the device/user mode in Beta4 as well as other things.

Another thing i already talk about is the apply_prf script that isn't recreating the language setting, so if you have set-up a phone in french by login in with the phone, when you do an apply_prf on this extension, the NPD key goes back to DND and all the keys are back to english, so in a french system, the apply_prf script isn't usable.

Fixed in Beta4, variable naming issue. Allons bon, cela doit marcher aussi en francais... non?

Regards

aastra1

 

[vcallaway]

Checked the files and made sure they are what you specified.

Here is a clip of the trace:

    -- Executing [vmail@from-internal:1] Macro("SIP/2500-b760bbe0", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/2500-b760bbe0", "AMPUSER=2500") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/2500-b760bbe0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/2500-b760bbe0", "1|Set|REALCALLERIDNUM=2500") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/2500-b760bbe0", "AMPUSER=2500") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/2500-b760bbe0", "AMPUSERCIDNAME=Vince Callaway") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/2500-b760bbe0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/2500-b760bbe0", "AMPUSERCID=2500") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/2500-b760bbe0", "CALLERID(all)="Vince Callaway" <2500>") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/2500-b760bbe0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] NoOp("SIP/2500-b760bbe0", "Using CallerID "Vince Callaway" <2500>") in new stack
    -- Executing [vmail@from-internal:2] Set("SIP/2500-b760bbe0", "_NODEST=") in new stack
    -- Executing [vmail@from-internal:3] Macro("SIP/2500-b760bbe0", "record-enable|2500|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/2500-b760bbe0", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/2500-b760bbe0", "recordingcheck|20100526-142910|1274909350.414") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_additional.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'aastra-xml' logged on from 127.0.0.1
  == Manager 'aastra-xml' logged off from 127.0.0.1
  recordingcheck|20100526-142910|1274909350.414: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/2500-b760bbe0", "") in new stack
    -- Executing [vmail@from-internal:4] Macro("SIP/2500-b760bbe0", "dialout-trunk|7|vmail||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/2500-b760bbe0", "DIAL_TRUNK=7") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/2500-b760bbe0", "0?sub-pincheck|s|1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2500-b760bbe0", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/2500-b760bbe0", "DIAL_NUMBER=vmail") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/2500-b760bbe0", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/2500-b760bbe0", "OUTBOUND_GROUP=OUT_7") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/2500-b760bbe0", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/2500-b760bbe0", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/2500-b760bbe0", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/2500-b760bbe0", "outbound-callerid|7") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/2500-b760bbe0", "0|SetCallerPres|") in new stack

USEDEVSTATE = true
freepbx 2.5.2.3
asterisk 1.4.21.2


I also tried this on a brand new install to listen to a voice mail. Had to disconnect the phone to hangup.