[asterisk-users] Asterisk translator issue?

[Enrico Pasqualotto]

Hi all, I have a network with some asterisk in trunk with IAX2 and some
SIP/ZAP phone connect to this *.
In every call I need to use only alaw codec so in all conf file I have
set disallow=all and allow=alaw.
I try also to make some tuning of my environment removing unused codec
and application.

If I remove the codec_ulaw.so when I try to call I see this:

[Oct  5 12:15:33] WARNING[16637]: chan_iax2.c:8021 iax2_request: Unable
to create translator path for unknown to ulaw on IAX2/CTM1-283
    -- Hungup 'IAX2/CTM1-283'
[Oct  5 12:15:33] WARNING[16637]: app_dial.c:1090 dial_exec_full: Unable
to create channel of type 'IAX2' (cause 0 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)

instead I use only alaw.
Infact if I keep che codec_ulaw.so and during a call watch the used
codec all are alaw.

Anyone can explain me where is the problem?

P.S. for me is not a problem to keet one file but is interesting to know
 who want to translate who.

Thanks Enrico.
-- 
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
http://www.linkedin.com/in/epasqualotto
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/x-pkcs7-signature
Size: 3557 bytes
Desc: S/MIME Cryptographic Signature
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20071005/f2c0d85f/attachment.bin