[asterisk-users] Defining sip users through mysql

[Arkaitz]

Hi all,
I had the same problem before so i tried reinstalling and using all
the defaults i could. But unfortunatelly it's the same, it seems that
mysql defined users can't access to the codec files. In both
situations the phone registers and i can see it with "sip show
peers"(with rtcachefriends=yes for mysql).
Using default context with default extensions.conf
Using sip.conf:
[linksys]
callerid="linksys"
type=friend
user=linksys
secret=linksys
context=default
host=dynamic
If I call extension 2:

    -- Executing BackGround("SIP/linksys-081a5678", "demo-moreinfo")
in new stack
    -- Playing 'demo-moreinfo' (language 'en')
    -- Executing Goto("SIP/linksys-081a5678", "s|instruct") in new stack
    -- Goto (default,s,6)
    -- Executing BackGround("SIP/linksys-081a5678", "demo-instruct")
in new stack
    -- Playing 'demo-instruct' (language 'en')
    -- Executing WaitExten("SIP/linksys-081a5678", "") in new stack
    -- Timeout on SIP/linksys-081a5678, going to 't'
    -- Executing Goto("SIP/linksys-081a5678", "#|1") in new stack
    -- Goto (default,#,1)
    -- Executing Playback("SIP/linksys-081a5678", "demo-thanks") in new stack
    -- Playing 'demo-thanks' (language 'en')
    -- Executing Hangup("SIP/linksys-081a5678", "") in new stack
All correct.

Using mysql:
mysql> select callerid,type,name,secret,context,host,allow,disallow from sip;
+----------+--------+---------+---------+---------+---------+-------------------------+----------+
| callerid | type   | name    | secret  | context | host    | allow
               | disallow |
+----------+--------+---------+---------+---------+---------+-------------------------+----------+
| Linksys  | friend | linksys | linksys | default | dynamic |
g729;ilbc;gsm;ulaw;alaw | all      |
+----------+--------+---------+---------+---------+---------+-------------------------+----------+
If I call extension 2:
    -- SIP Seeding peer from astdb: 'linksys' at linksys at 20.0.0.70:5060 for 3600
    -- Executing BackGround("SIP/linksys-081a5678", "demo-moreinfo")
in new stack
Oct  3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to gsm
Oct  3 13:16:17 WARNING[13896]: file.c:824 ast_streamfile: Unable to
open demo-moreinfo (format g729): No such file or directory
Oct  3 13:16:17 WARNING[13896]: pbx.c:5798 pbx_builtin_background:
ast_streamfile failed on SIP/linksys-081a5678 for demo-moreinfo
    -- Executing Goto("SIP/linksys-081a5678", "s|instruct") in new stack
    -- Goto (default,s,6)
    -- Executing BackGround("SIP/linksys-081a5678", "demo-instruct")
in new stack
Oct  3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to gsm
Oct  3 13:16:17 WARNING[13896]: file.c:824 ast_streamfile: Unable to
open demo-instruct (format g729): No such file or directory
Oct  3 13:16:17 WARNING[13896]: pbx.c:5798 pbx_builtin_background:
ast_streamfile failed on SIP/linksys-081a5678 for demo-instruct
    -- Executing WaitExten("SIP/linksys-081a5678", "") in new stack
Oct  3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to
find a codec translation path from ulaw to g729
Oct  3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to
find a codec translation path from ulaw to g729
    -- Timeout on SIP/linksys-081a5678, going to 't'
    -- Executing Goto("SIP/linksys-081a5678", "#|1") in new stack
    -- Goto (default,#,1)
    -- Executing Playback("SIP/linksys-081a5678", "demo-thanks") in new stack
    -- Playing 'demo-thanks' (language 'en')

I only hear demo-thanks.
Any hint please?
Thanks for your time

-- 
Arkaitz