Can't check VM messages all of a sudden

[kamelb]

Hello,

This must have started to happen after a recent FPBX module update but all of a sudden we can’t seem to be able to dial into VM and hear the messages!

Here’s a snippet from the log. It reaches the point where it would read the caller’s phone number and it hangs up.

-- <PJSIP/527-00001bf1> Playing 'digits/1.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-INBOX.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-and.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'digits/1.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-Old.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-message.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-onefor.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-INBOX.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-messages.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-opts.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-helpexit.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-first.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-message.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-received.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'digits/day-3.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'digits/mon-0.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'digits/h-4.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'digits/at.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'digits/9.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'digits/50.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'digits/4.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'digits/a-m.slin16' (language 'en')
-- <PJSIP/527-00001bf1> Playing 'vm-from-phonenumber.slin16' (language 'en')

[2023-01-12 12:29:47] WARNING[2062][C-0000080e]: file.c:824 ast_openstream_full: File does not exist in any format
[2023-01-12 12:29:47] WARNING[2062][C-0000080e]: file.c:1303 ast_streamfile: Unable to open (format (g729)): No such file or directory
== Spawn extension (from-internal, *98, 16) exited non-zero on ‘PJSIP/527-00001bf1’
– Executing [h@from-internal:1] Macro(“PJSIP/527-00001bf1”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/527-00001bf1”, “1?theend”) in new stack

Seems like a codec translation issue but not sure how to resolve. Like I said it used to work fine.

Any ideas?

Thanks

[jcolp (Joshua Colp)]

This sounds like a duplicate of Voice mail hangs up if play CID and play envelope is on which was fixed in the Asterisk versions listed on [ASTERISK-30248] ast_get_digit_str adds bogus initial delimiter if first character not to be spoken - Digium/Asterisk JIRA with I believe a workaround in the forum thread.

[lgaetz (Lorne Gaetz)]

The fixed versions of asterisk for this issue are in the fpbx testing repo now. @franckdanard built and published a few hr ago.