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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