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 CASI'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.-- In science, one tries to tell people in such a way as to be understood by everyone something that no-one ever knew before. In poetry, it is the exact opposite. - Paul Dirac Please reply to the list; please *don't* CC me.
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) Please reply to the list; please *don't* CC me.