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.
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