[asterisk-users] ADSI - Unable to send CAS...

[Antony Stone]

Hi.

Can anyone give me some insight into how to deal with this failure mode?

It makes no sense to me that Asterisk is even considering ADSI (where A stands 
for Analogue) on a SIP call.  How can I disable this so that the error does 
not occur?

On Friday 19 March 2021 at 11:17:36, Antony Stone wrote:

> On Thursday 18 March 2021 at 18:07:54, Antony Stone wrote:
> > Hi.
> > 
> > I'm running Asterisk 13 and Asterisk 16 using SIP trunks only (to a
> > commercial trunking provider).  I have no analogue interfaces.
> > 
> > A user reported dialling in to voicemail (the standard Asterisk Comedian
> > Mail service) from a mobile phone and being unable to select the menu
> > options.
> > 
> > The call path in that case would be:
> > 
> > Mobile phone -> mobile network provider -> SIP trunking provider ->
> > Asterisk
> > 
> > She dialled in three times within a 10 minute period.  On the first two
> > occasions she was unable to select any menu options - she was pressing
> > the buttons on the mobile dialpad and getting the confirmation "beep"
> > from the phone, but Asterisk did not register any DTMF coming through
> > and therefore did not navigate the menu system.
> > 
> > On the third attempt to dial in, the call and the menus worked as
> > expected.
> > 
> > When I reviewd the Asterisk logs for these calls afterwards, I saw on
> > both the first two calls, immediately after the dial plan went to
> > VoiceMailMain(), the message:
> > 
> > WARNING[19645][C-00000278]: res_adsi.c:250 in __adsi_transmit_messages:
> > Unable to send CAS
> 
> I've investigated a bit further, and I've had around 1200 accesses to
> VoiceMailMain so far this year, and in 5 cases the above message appeared
> immediately afterwards in the log file.
> 
> So, it clearly doesn't happen often, but when it does, it prevents the user
> from navigating the menu, and therefore I need to stop thei failure mode.
> 
> > So, I have two questions:
> > 
> > 1. Why is Asterisk even attempting to do ADSI on a SIP trunk (my
> > understanding of ADSI is that the A stands for Analogue, so it should not
> > even apply to a SIP call)?
> > 
> > 2. What do I need to do to either disable ADSI, or avoid the above error
> > message?
> > 
> > 
> > Thanks,
> > 
> > 
> > Antony.

-- 
"It would appear we have reached the limits of what it is possible to achieve 
with computer technology, although one should be careful with such statements; 
they tend to sound pretty silly in five years."

 - John von Neumann (1949)

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