disa

[vortex1]

Hi.
i do have asterisk running on my system with the analog line connected on it.
My setup has an fct with a mobile card connected through a linksys spa 3000 to the fritz.
I would like to make use of the disa functions. did anybody implement disa on asterfritz?

 

[vortex1]

Hi. Will someone be so kind to help me configure DISA in my system.
I got an inbound number eg 123456 that when you call it it reaches asterisk and the extension eg 10 reacts. I would like to configure DISA so when i will call the 123456 i wll get a prompt for a passwd. After the passwd i want to be able to call as if i was an extensio eg 10 to the fritz.
I am running asterisk 1.4 in a usb to a fritz 7140.

Anyone please?

 

[rentier-s]

Hi

A little hard to say without knowing your extensions.conf. In general the DISA syntax is
DISA(passcode[,context[,callerid[,mailbox[@vmcontext]]]])

Example extensions.conf

[incomming]
exten => 12345,1,DISA(0000,default,10)

Calling 12345 will ask you for a PIN, enter 0000 and you will be given a dialtone as user 10 in context default.

 

[vortex1]

Hi. Thanks a lot for the quick answer.
My extensions.conf is
[globals]
SIPPHONES=SIP/10&SIP/11&SIP/12&SIP/13&SIP

[default]

[analog-in]
exten => s,1,Dial(${SIPPHONES},60,tr)

[gsm-in]
exten => 123456,1,Dial(${SIPPHONES},60,tr)


[analog-out]
exten => _XXXXX,1,Dial(CAPI/ANALOG/${EXTEN},60,Tt)
exten => _XXXX,1,Dial(CAPI/ANALOG/${EXTEN},60,Tt)
exten => _XXX,1,Dial(CAPI/ANALOG/${EXTEN},60,Tt)

[local]
exten => _1X,1,Dial(SIP/${EXTEN},60,tr)

[sip10]
include => local
include => analog-out

[sip11]
include => local
include => analog-out



[sip12]
include => local

[sip13]
include => local

where 123456 is my gsm phone to react to DISA.
so how will the example look like?

 

[vortex1]

Hi.
For some strange reason this is not working.
Do i need to load any modules? If yes how?
i am calling the number and it goes straight to the asterisk. then i do not get a prompt to enter a passwd. it is like i did not setup the inbound routes.

 

[rentier-s]

Anything visible on the CLI during call attempt? "asterisk -rvvv" to get verbose level 3 at least.

app_disa marked in make menuconfig and autoload=yes in modules.conf?

 

[vortex1]

Hi.
I did not make menuconfig since i used the preconfigured from
http://spblinux.de/fbox.new/cfg_asterisk14
Unfortunatelly i am seeing any disa modules in modules.conf.
Any ideas how to enable it?

 

[rentier-s]

There is a CLI command to check if the module is loaded. "core show apps like disa" or anything the like. I don't have it in mind exactly, but "help" or google might help you.

If you cannot find it that way, check for an app_disa.so in your modules directory. Depends on your environment where to find this.
In case the file exists but is not loaded, try autoload=yes in the general section of modules.conf or type load=>app_disa.so as one of the last lines.

Still I'd be interested in the CLI output during the call, as it might contain some error message about the issue.

 

[vortex1]

hi. the modules directory does not contain any disa module.
the load=>app_disa.so did not change anything.
Do i have to get the module from the http://spblinux.de/fbox.new/modules14/? ife yes how?

 

[vortex1]

Anyone that could help me??

 

[cmmehl]

Why wouldn't you use that feature on your SPA3k? Not quite sure right now what it's named as I'm far from mine (not DISA for sure), but you can certainly get that googled.

Good luck
Chris

 

[vortex1]

Hi. Thanks a lot for the quick answer.
I have not found anything about SPA3k and DISA (embedded wise).
My question is how to enable all the necessary modules on fritz to build DISA functions.

Could someone be so kind to help me with the necessary modules?
So far so good i have the extensions ready and i am missing the modules in modules.conf

 

[cmmehl]

Sorry can't help you with the asterisk modules on fritz, but I have found what the function in the SPA3k is called - PSTN-to-VoIP-GAteway.

It should enable you to do what you wanted.

Cheers
Chris

 

[vortex1]

Hi.
The SPA3k feaute - PSTN-to-VoIP-GAteway, is to transfer the signal through ethernet to the voip server (asterisk).
If i am missing a feature of DISA on SPA3k please let me know how to make it.
I would be grateful if you could tell me.

 

[cmmehl]

Hi,

my understanding of your needs was to get a dialtone, so you can continue placing calls via VoIP (i.e. your asterisk-server). This is was the mentioned function of the SPA3k does, unless I have completely misunderstood it. Could be the case, though, as I have never used it myself.

Cheers
Chris

 

[vortex1]

Since we have gone completely the wrong way. I will start from the beginning.

I would like someone to help me load the modules on a fritzbox for disa.
Could someone be so kind to help me out over here?

 

[rentier-s]

Do i have to get the module from the http://spblinux.de/fbox.new/modules14/? ife yes how?

I'm not familiar with asterfritz, but those modules seem to be already compiled. Did you try to copy the app_disa.so from your above link to your modules directory and perform a "module load app_disa.so" on the CLI?

 

[vortex1]

Where about in the following is the modules folder?
# ls
adsi.conf extconfig.conf oss.conf
adtranvofr.conf extensions.ael passwd
agents.conf extensions.conf phone.conf
alarmreceiver.conf features.conf privacy.conf
alsa.conf festival.conf queues.conf
asterisk.conf iax.conf res_mysql.conf
capi.conf iaxprov.conf res_odbc.conf
cdr.conf indications.conf rpt.conf
cdr_custom.conf logger.conf rtp.conf
cdr_manager.conf manager.conf sccp.conf
cdr_mysql.conf meetme.conf sip.conf
cdr_odbc.conf mgcp.conf sip_notify.conf
cdr_pgsql.conf misdn.conf skinny.conf
cdr_tds.conf mobile.conf voicemail.conf
codecs.conf modem.conf vpb.conf
dnsmgr.conf modules.conf zapata.conf
dundi.conf musiconhold.conf
enum.conf osp.conf
# pwd
/var/media/ftp/Generic-USBDisk-01/addons/asterisk14/etc/asterisk
#

 

[rentier-s]

A real Asterisk's is located at /var/lib/asterisk/modules/, maybe your path could look like /var/media/ftp/Generic-USBDisk-01/addons/asterisk14/var/lib/asterisk/modules/ :noidea:

Have a look at your asterisk.conf, there's a astmoddir= option, perhaps that's a hint.

 

[vortex1]

Hi. OK i have found the modules folder and i loaded the module
i am calling from my mobile the number that is supposed to be the DISA and i am getting the following
fritz*CLI>
-- Executing [211123456@voice-trunk2:1] DISA("SIP/211123456-005acde0", "0000|default|300") in new stack

== Spawn extension (voice-trunk2, 211123456, 1) exited non-zero on 'SIP/3021 17707041-005acde0'

fritz*CLI>

fritz*CLI> module load app_disa.so
[Sep 10 08:14:39] WARNING[1100]: loader.c:643 load_resource: Module 'app_disa.so' already exists.

fritz*CLI>

i am not hearing anything on my mobile, but i am trying to press the 0000. Nothing happens. Any ideas?