[Asterisk-bsd] Tracking down the ast_translator_build_path warnings with 1.2.12.1

[Thomas Sandford]

It is evident that a number of people are seeing these warnings with the new 
port version, but also that others are _not_.

In order to track down the problem it would be really useful if people who 
are using the 1.2.12.1 version on FreeBSD could provide the following 
information:

1) Whether they see these warnings or not.
2) The version of FreeBSD installed
3) Whether they are running the GENERIC, or a patched kernel (and if 
patched, in what way).
4) Any options used in building the asterisk port
5) The versions of libpri, zaptel and any other asterisk related ports - eg 
asterisk-addons
6) Any zaptel (or similar, eg Sangoma) hardware installed
7) The codecs they believe are actually in use.
8) Anything else that would help determine the circumstances, eg

"I see the problem when calling an external IAX (IaxTel) number using  a 
Sipura 841 registered directly with the * box. I can hear the called party 
but they cannot hear me" [nb this is an example given purely for 
illustration - I am _not_ seeing any problems here].

I'm principally after results from people using the FreeBSD port, but if 
anyone has seen the same warnings with a native compilation of 1.2.12.1 on 
FreeBSD (ie not using the port framework) or on another OS that would be a 
useful datapoint.

[please reply to my original message unless there is a specific 
configuration/results change you are trying to highlight, and snip my 
results below to keep the thread relatively simple]

To start the ball rolling with my setup:
1) Whether they see these warnings or not.
No
2) The version of FreeBSD installed
5.4-RELEASE, with freebsd-update security patches
3) Whether they are running the GENERIC, or a patched kernel (and if 
patched, in what way).
GENERIC kernel, plus OPTIONS SMP
4) Any options used in building the asterisk port
No
5) The versions of libpri, zaptel and any other asterisk related ports - eg 
asterisk-addons
libpri-1.2.3 , openh323-1.18.0_1 , pwlib-1.10.0,1 , spandsp-0.0.2.p26 , 
zaptel-1.0
6) Any zaptel (or similar, eg Sangoma) hardware installed
TDM400P with one FXS, one FXO module installed
7) The codecs they believe are actually in use.
ulaw
8) Anything else that would help determine the circumstances
Calls tried:
Sipura 841 registered with Asterisk instance, limited to ulaw in sip.conf, 
to asterisk demo
[ditto] to a remote asterisk (conference room) via IAX (also tried with 
SIP).

-- 
Thomas Sandford