Bonjour à tous,

J' essaye de connecter un SPA-3102 derniere une freebox.

J'arrive en local a passer des appels vers le réseau RTC, mais, deniere un nat je n' ai pas de son.
(j'ai d' autre probleme que je traiterai apres).

Mon serveur astersik se trouve deriere un NAT, j'ai fais une redirection de port TCP/UDP 5060 et [10000-20000] vers l' IP de mon serveur.

voici le debbug de la CLI quand je passe un appel :

Code:
jn-serveur*CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [203@local:1] Dial("SIP/HClO-00000015", "SIP/fixe-voip,10") in new stack
  == Using SIP RTP CoS mark 5
    -- Called fixe-voip
    -- SIP/fixe-voip-00000016 is ringing
    -- SIP/fixe-voip-00000016 answered SIP/HClO-00000015
    -- Packet2Packet bridging SIP/HClO-00000015 and SIP/fixe-voip-00000016
[Sep  7 13:27:48] NOTICE[1859]: chan_sip.c:21331 handle_request_subscribe: Received SIP subscribe for peer without mailbox: HClO
[Sep  7 13:28:03] WARNING[1859]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 9dcbed5c-1205-1910-93d0-002354382155@Cergy-PC for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt.
[Sep  7 13:28:03] WARNING[1859]: chan_sip.c:3806 retrans_pkt: Hanging up call 9dcbed5c-1205-1910-93d0-002354382155@Cergy-PC - no reply to our critical packet (see doc/sip-retransmit.txt).
  == Spawn extension (local, 203, 1) exited non-zero on 'SIP/HClO-00000015'
ici mon sip.conf :

Code:
[general]
context=local                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is yes)
;match_auth_username=yes        ; if available, match user entry using the
                                ; 'username' field from the authentication line
                                ; instead of the From: field.
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
                                ; Default is enabled
;realm=mydomain.tld             ; Realm for digest authentication
                                ; defaults to "asterisk". If you set a system name in
                                ; asterisk.conf, it defaults to that system name
                                ; Realms MUST be globally unique according to RFC 3261
                                ; Set this to your host name or domain name
udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to (0.0.0.0 binds to
bindport=5060
bindaddr=0.0.0.0
externalip=78.226.**.**
localnet=192.168.1.0/255.255.255.0    
disallow=all
allow=alaw
allow=g726
allow=g729
srvlookup=yes

[HClO]
secret=*****
context=local
type=friend
host=dynamic
nat=yes
canreinvite=yes

[fixe-voip]
secret=*****
directmedia=no
context=local
type=friend
host=dynamic
nat=yes
canreinvite=no


[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=from-office
        type=friend

[natted-phone](!,basic-options)   ; another template inheriting basic-options
        nat=yes
        directmedia=no
        host=dynamic

[public-phone](!,basic-options)   ; another template inheriting basic-options
        nat=no
        directmedia=yes

[my-codecs](!)                    ; a template for my preferred codecs
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw

[ulaw-phone](!)                   ; and another one for ulaw-only
        disallow=all
        allow=ulaw


et la mon extension.conf

Code:
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;---------------------------------------------------------
exten => 200,1,Dial(SIP/HClO,10)
exten => 203,1,Dial(SIP/fixe-voip,10)
exten => 212,1,Answer
exten => 212,2,Playback(demo-echotest)
exten => 212,3,Echo()
exten => 0,1,Dial(SIP/SPA/${exten},,T)
Merci de votre aide,

Cordialement.