DISA problem

stuck

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DISA problem-updated with CLI output

I wonder if anyone is having this problem with DISA on the new PiaF:
I dial in from my cellphone, based on CID, asterisk sends the call to DISA but I don't get a dialtone. Both the phone and FOP shows the call connected. But no dialtone at all. Even if I start dialing an internal extension nothing happens.

Update: This is the CLI output with verbosity set to 5
Hoping someone can help

-- Executing [xxxxxxxxxx@from-pstn:1] Set("SIP/xxxxxxxxxx-09c2c3b8", "__FROM_DID=xxxxxxxxxx") in new stack
-- Executing [xxxxxxxxxx@from-pstn:2] Gosub("SIP/xxxxxxxxxx-09c2c3b8", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/xxxxxxxxxx-09c2c3b8", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/xxxxxxxxxx-09c2c3b8", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/xxxxxxxxxx-09c2c3b8", "") in new stack
-- Executing [xxxxxxxxxx@from-pstn:3] GotoIf("SIP/xxxxxxxxxx-09c2c3b8", "1 ?cidok") in new stack
-- Goto (from-pstn,xxxxxxxxxx,5)
-- Executing [xxxxxxxxxx@from-pstn:5] NoOp("SIP/xxxxxxxxxx-09c2c3b8", "CallerID is "Irvine CA" <YYYYYYYYYY>") in new stack
-- Executing [xxxxxxxxxx@from-pstn:6] Goto("SIP/xxxxxxxxxx-09c2c3b8", "disa|1|1") in new stack
-- Goto (disa,1,1)
-- Executing [1@disa:1] Set("SIP/xxxxxxxxxx-09c2c3b8", "TIMEOUT(digit)=5") in new stack
-- Digit timeout set to 5
-- Executing [1@disa:2] Set("SIP/xxxxxxxxxx-09c2c3b8", "TIMEOUT(response)=10") in new stack
-- Response timeout set to 10
-- Executing [1@disa:3] DISA("SIP/xxxxxxxxxx-09c2c3b8", "no-password|from-internal") in new stack
== Spawn extension (disa, 1, 3) exited non-zero on 'SIP/xxxxxxxxxx-09c2c3b8'
 

stuck

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Any ideas?
This is preventing me from putting my PiaF box to live and replace my TB.
 

stuck

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No I have not installed extra sounds...
would it be as simple as copying the sound files from my TB over to PiaF?
 

stuck

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Will try that when I get home and post results...
Thanks fellas...

BTW I saw that thread and didn't think it would solve my problem since When I start dialing from the cellphone without hearing the dialtone sound, it doesn't pick up my DTMF from the cellphone. Anyways, can't wait to go home to try it...
 

stuck

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For some reason it didn't work. When I unzipped it using this command:
tar zxfv asterisk-extra-sounds-en-gsm-current.tar.gz it didn't create a new directory, it unzipped everything in /usr/src directory.
So I issued
chown -R asterisk:asterisk /var/lib/asterisk/sounds
command in /usr/src command. But I don't know if it was successful.

Anyways, I rebooted PiaF and tried DISA with the same result.
 

stuck

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Sounds
Just for general information, one can find the sounds at http://ftp.digium.com/pub/asterisk/r...s-1.2.1.tar.gz. And a simple make install will get them loaded quickly.

Then I tried this suggestion downloaded the file in /usr/src
tar zxfv asterisk-sounds-1.2.1.tar.gz
cd asterisk-sounds-1.2.1
make install

It seemed to install okay then I restarted Asterisk by Amportal restart and still samething. No sound on DISA

Asterisk CLI output is identical to above
 

Mic

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stuck -

The Asterisk additional sound files need to be placed in /var/lib/asterisk/sounds make sure you chown -R asterisk:asterisk /var/lib/asterisk/sounds once you cp them there!

-Mic
 

stuck

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stuck -

The Asterisk additional sound files need to be placed in /var/lib/asterisk/sounds make sure you chown -R asterisk:asterisk /var/lib/asterisk/sounds once you cp them there!

-Mic
Thanks Mic
I executed that command: chown -R asterisk:asterisk /var/lib/asterisk/sounds
from the /usr/src/asterisk-sounds-1.2.1 directory
and then confirmed that the sound files are in /var/lib/asterisk/sounds.
then rebooted...
still the same thing though...
 

stuck

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Okay update
If I set a password. Then everything works fine and it asks me for password and then give me dialtone after password is entered.
But if I remove password I get silence...it seems like when it gets to this line:
-- Executing [3@disa:3] DISA("SIP/XXXXXXXXXX-0a1e6e40", "no-password|from-internal") in new stack

the "no-password|from-internal" part is where the problem is...

This used to work in TB
 

dallas

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So why would you want set up DISA without a password? Without a password you are exposed to toll fraud. With very little effort a hacker could use your PBX to make unrestricted calls for free.
 

Mic

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I COMPLETELY AGREE WITH DALLAS!! So be sure you really want this -- you can add these lines at the end of extensions_custom.conf

; custom DISA
[custom-disa]
exten => s,1,Answer
exten => s,n,Wait(2)
exten => s,n,DISA(no-password|from-internal)

Goto your inbound route and at the bottom for Set Destination choose Custom App and enter = custom-disa,s,1

Submit - reload and within a couple of days you might have more phone bill than you can afford ;-)

****DANGER WILL ROBINSON***DANGER***
 

stuck

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I COMPLETELY AGREE WITH DALLAS!! So be sure you really want this -- you can add these lines at the end of extensions_custom.conf

; custom DISA
[custom-disa]
exten => s,1,Answer
exten => s,n,Wait(2)
exten => s,n,DISA(no-password|from-internal)

Goto your inbound route and at the bottom for Set Destination choose Custom App and enter = custom-disa,s,1

Submit - reload and within a couple of days you might have more phone bill than you can afford ;-)

****DANGER WILL ROBINSON***DANGER***

Well Tried That... and still just silence
And I've taken the risks into concern and hopefully the Callerid authentication and the dialplan/outbound route restrictions I've made will keep me safe enough...

Here is the Asterisk CLI
-- Executing [XXXXXXXXXX@from-pstn:1] Set("SIP/XXXXXXXXXX-0a172e90", "__FROM_DID=XXXXXXXXXX") in new stack
-- Executing [XXXXXXXXXX@from-pstn:2] Gosub("SIP/XXXXXXXXXX-0a172e90", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/XXXXXXXXXX-0a172e90", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/XXXXXXXXXX-0a172e90", "0?blacklisted") in new stack

-- Executing [s@app-blacklist-check:3] Return("SIP/XXXXXXXXXX-0a172e90", "")in new stack

-- Executing [XXXXXXXXXX@from-pstn:3] GotoIf("SIP/XXXXXXXXXX-0a172e90", "1 ?cidok") in new stack

-- Goto (from-pstn,XXXXXXXXXX,5)

-- Executing [XXXXXXXXXX@from-pstn:5] NoOp("SIP/XXXXXXXXXX-0a172e90", "CallerID is "Orange CA" <YYYYYYYYYY>") in new stack

-- Executing [XXXXXXXXXX@from-pstn:6] Goto("SIP/XXXXXXXXXX-0a172e90", "custom-disa|s|1") in new stack

-- Goto (custom-disa,s,1)

-- Executing [s@custom-disa:1] Answer("SIP/XXXXXXXXXX-0a172e90", "") in newstack

-- Executing [s@custom-disa:2] Wait("SIP/XXXXXXXXXX-0a172e90", "2") in new stack

-- Executing [s@custom-disa:3] DISA("SIP/XXXXXXXXXX-0a172e90", "no-password|from-internal") in new stack

== Spawn extension (custom-disa, s, 3) exited non-zero on 'SIP/XXXXXXXXXX-0a172e90'
 

wardmundy

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Just a heads up that callerID authentication for DISA doesn't keep you safe from anybody. Almost everyone running Asterisk with a good provider can spoof any callerid. I've never heard of anyone using DISA without a password. It's a really, really bad idea.
 

Mic

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s,3 should have given dial tone -No others sounds are played - worked for me...THEN I RIPPED IT BACK OUT :) from all of the feedback are you beginning to get the idea that no-password is a bad idea ;-)
 

stuck

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Well, since I got "nerd uno" to chime in and make the recommendation, I've kept my PiaF's DISA with password and replaced my TB with PiaF. Now I've just gotta reprogramm my contact list on my phone.
It just bugs me that something doesn't work in my box...
Anyways, thanks everyone for your help.
 

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