[asterisk-users] Routing issue

[David Gonzalez]

Hi list

I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk
solutions and appliances.

I installed TrixBox on a litle PC @ home and a x100p card which is
recognized as a Zaptel card, I made some in/outbound routes and they seem to
work but I have a problem with SIP softphones. I created 2 estensions 1000
and 1001 they're both in different cities, when I 1000 (on the same network
as TrixBox) dial 1001 (the other city) they answer and can hear me, but I
don't hear them, and when they call *43 for echo test it plays the "You're
entering echo test..." but when it tries to start echo it just hangs up. and
the log says

    -- Executing [*43 at from-internal:1] Answer("SIP/1000-08939150", "") in
new stack
    -- Executing [*43 at from-internal:2] Wait("SIP/1000-08939150", "1") in new
stack
    -- Executing [*43 at from-internal:3] Playback("SIP/1000-08939150",
"demo-echotest") in new stack
    -- <SIP/1000-08939150> Playing 'demo-echotest' (language 'en')
  == Spawn extension (from-internal, *43, 3) exited non-zero on
'SIP/1000-08939150'
    -- Executing [h at from-internal:1] Macro("SIP/1000-08939150",
"hangupcall") in new stack
    -- Executing [s at macro-hangupcall:1] ResetCDR("SIP/1000-08939150", "w")
in new stack
    -- Executing [s at macro-hangupcall:2] NoCDR("SIP/1000-08939150", "") in
new stack
    -- Executing [s at macro-hangupcall:3] GotoIf("SIP/1000-08939150",
"1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s at macro-hangupcall:6] GotoIf("SIP/1000-08939150",
"1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s at macro-hangupcall:9] GotoIf("SIP/1000-08939150",
"1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s at macro-hangupcall:11] Hangup("SIP/1000-08939150", "") in
new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/1000-08939150' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/1000-08939150'

I'd appreciate your help a lot, I'm not if this is a forewall issue or
something wrong with my asterisk config.

Thanks a lot.

-- 
DAVID GONZALEZ H.
GNU/Linux Debian+SuSE+RedHat+LFS
TECNICO EN REDES
NETWORK ADMIN
http://www.computrabajo.com.co/cvs/dgonzalezh
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