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 :
ici mon sip.conf :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'
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
Merci de votre aide,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)
Cordialement.
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