[Asterisk-Users] Asterisk Internal sip calls I can´t send/recive

[Omar Lopez Limonta]

When i made internal call into my LAN using x-lite sip phone client I
retrive in askterisk CLI :

-----------
ERROR
----------
Verbosity is at least 6
    -- Remote UNIX connection
    -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack
    -- Called 201
May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 0afcadc1422e63800115943201a885fb at 192.168.1.44
for seqno 102 (Critical Request)
  == No one is available to answer at this time
    -- Executing VoiceMail("SIP/201-979d", "201") in new stack
    -- Playing 'vm-intro' (language 'es')
  == Spawn extension (anurix, 201, 2) exited non-zero on 'SIP/201-979d'
May 29 18:09:34 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call
C15A57EC-51A0-4157-BCE5-B09C0A99FD26 at 192.168.1.33 for seqno 52991
(Non-critical Response)
---------

(192.168.1.44 is the Asterisk HOST)

I can do outgoing calls with Zap interface without problems, only i
__can´T__ do calls into my lan with SIP phone/protocol  , i can listen
voicemail because is the second action on extesion.

These are my configuration files:

sip.conf
-------------

[203]
type=friend
qualify=yes
username=203
secret=203
host=dynamic
callerid=\"JuanI\" <203>
canreinvite=no
reinvite=no
context = anurix
transfer=yes
mailbox=203
callgroup=1
pickupgroup=1
nat=never
----------
extensions.conf
--------------
[exterior]
exten => _0.,1,Dial(Zap/1/${EXTEN:1},60,r)
exten => _0.,2,Hangup
;Contestar llamada
[entrada]
exten => s,1,Wait,11
exten => s,2,Answer
exten => s,3,Wait,1
exten => 1,1,Dial(SIP/200,60,Ttr)
exten => 2,1,Dial(SIP/201,60,Ttr)
exten => 3,1,Dial(SIP/202,60,Ttr)
exten => 4,1,Dial(SIP/203,60,Ttr)

;BUZONES DE VOZ DESAHABILITADOS

[anurix]
include => exterior
exten => 200,1,Dial(SIP/200,60,t)
exten => 200,2,Voicemail(200)
exten => 200,3,Hangup
exten => 201,1,Dial(SIP/201,60,t)
exten => 201,2,Voicemail(201)
exten => 201,3,Hangup
exten => 202,1,Dial(SIP/202,60,t)
exten => 202,2,Voicemail(202)
exten => 202,3,Hangup
exten => 203,1,Dial(SIP/203,60,t)
exten => 203,2,Voicemail(203)
exten => 203,3,Hangup


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