DISA Problem

pti2000

Joined
Mar 27, 2008
Messages
77
Likes
0
Points
0
#1
Disa does not work on Elastix 2.0 When you call in the return dial tone is constant and when you dial a number the system give you an error. Does any one have a fixed for this?
 

andyshawn

Joined
Apr 3, 2009
Messages
113
Likes
0
Points
0
#2
Have you checked the DTMF settings? Are you dialing into the system via the PSTN or via a VoIP handset?
 

pti2000

Joined
Mar 27, 2008
Messages
77
Likes
0
Points
0
#3
The system has a sip trunk with 8 channel, Now this works well with the older version, but not 2.0
any way I used my cell phone to call into the system via DISA and call out via the PBX
but I can't get it to work on this version.
 

andyshawn

Joined
Apr 3, 2009
Messages
113
Likes
0
Points
0
#4
Ok then. That sounds like there might be a bug in the new elastix code then. Report it in the bug section, so the developers can test it out and fix the problem.
 

Bob

Joined
Nov 4, 2007
Messages
2,400
Likes
1
Points
36
#5
pti2000,

Andy's suggestion of DTMF is not off the mark.

Have you confirmed that DTMF is working on your system (just because your SIP phones work, offers no assurance that your DTMF is working.

For testing, setup an IVR on your system, call it using a Mobile and see if you can press the numbers for the IVR and check to see if this works. If not, look at you DTMF settings.

Another failure that can occur with DTMF is poor line quality (e.g. SIP connection that is not ideal, this can fail the DTMF straight away).

So before you throw in a bug report (which I fully recommend you do)...just try to lessen the variables...this is not to say DISA is not possibly faulty, just check all options that you can....

Also, you would have rebuilt your system to use Elastix 2.0. No matter how much you think you have replicated the setup, there is always something missed (codecs etc)....I still make the mistakes....

Regards

Bob
 

dicko

Joined
Oct 24, 2008
Messages
4,099
Likes
0
Points
0
#6
To further elucidate Bob and Andy's input, Cell phones invariable use GSM, that is just the nature of the technology, so any attempt to use any inband signalling will allways fail miserably, 64k > 32k (or vise versa) = bad dtmf it's just fysix . if you don't use rfc2833 you are doomed to failure, If your VSP does the unthinkable and tries to transcode to inband (strangely some do) then you are SOL :) check with "sip debug" at the asterisk CLI to make sure that the SIP signalling is apparent to both parties
 

pti2000

Joined
Mar 27, 2008
Messages
77
Likes
0
Points
0
#7
Ok it looks like its the DTMF, Because I setup an IVR and I called in with a cell but could not make contact when I press a number. How do I fix this, where do I change the dtmf setting for this?
 

dicko

Joined
Oct 24, 2008
Messages
4,099
Likes
0
Points
0
#8
Probably with:-

dtmfmode=rfc2833

in all legs of the call, particularly the one to the cell-phone, as stated before, some VSP's don't honor that and the user will have to dig deep into his phone to extend the dtmf length, truly bogus I agree but all VSP's are not equal, try another one before you go further. (JM2CWAE)
 

Bob

Joined
Nov 4, 2007
Messages
2,400
Likes
1
Points
36
#9
pti2000 said:
Ok it looks like its the DTMF, Because I setup an IVR and I called in with a cell but could not make contact when I press a number.
Good diagnosis!! ;)

Now the painful part, and as Dicko said...not all of them are created equal..

You may need to work with your VSP and determine the right settings. Ring and talk to them about your issue, they might have already come across it with another user and will be able to give you some answers....

As Dicko mentioned, the codec could play a part as well. The less transcoding between codecs the better. Please note CODEC's and DTMF are different things, but the codec and transcoding can have an effect (as I mentioned, so can line noise and poor transcoding).

For almost all setups, I stick the GSM codec for all external voice, so this means setting your Trunks to disallow = all and allow=GSM, so that it is the only codec negotiated. If your VSP has a way of locking the codec (as we said they are not all equal), then do that also. The most likely DTMF mode is rfc2833, but try to confirm this with your VSP....

At least you are now heading in the right direction.

Regards

Bob
 

pti2000

Joined
Mar 27, 2008
Messages
77
Likes
0
Points
0
#10
Ok in my trunk setting for the sip provider I am using in peer detail this what they require

disallow=all
allow=ulaw
canreinvite=no
context=from-trunk
dtmfmode=auto

Now in all the versions before Elastix 2.0 works fine with these settings
any way can you be more specific on where and what I need to change and or maybe play with to fixed this.
 

Members online

No members online now.

Latest posts

Forum statistics

Threads
30,900
Messages
130,884
Members
17,561
Latest member
marouen
Top