[Asterisk-Users] g723.1 voicemail/conference files segfault *

[HT]

Hi,

First of all I am not sure that what I am trying to do is correct/supported,
but here is what I'm trying to test:

Some of my endpoints only have g723 codecs. Because of this I am only
allowing g723.1 codec in sip.conf and h323.conf. Calls between endpoints
work fine. I am trying to configure voicemail and meetme applications. I see
that all voice files in asterisk are in gsm format and when I try to place a
call to the voicemail/meetme I get some message saying:

File channel.c, Line 1399 (ast_set_write_format): Unable to find a path from
2 to 1

This probably means that GSM<->G.723.1 transcoding is not supported (which
is normal). However, when I try to use a voice file pre-encoded with G.723.1
codec (for example conf-onlyperson.g723) I not getting the " Unable to find
a path from 2 to 1" message, but Asterisk segfaults.

My guess is that asterisk is probably capable of playing/streaming files to
g.723.1 endpoints if the files to be played are already encoded with g723
codec. Right?

Is this feature supported first of all and has someone already tested
voicemail/meetme apps with different voice files (.g723 for ex.)?

Off the topic: where can I find the core dump? I am running asterisk on
Redhat9.

H.