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Cette solution existe et elle est tout à fait correcte. b) Nat=route: Asterisk will send the audio to the port and ip where its receiving the audio from. Instead of relying on the addresses in the SIP and SDP messages. This will only work if the phone behind nat send and receive audio on the same port and if they send and receive the signaling on the same port. (The signaling port does not have to be the same as the RTP audio port). http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html Teste de validation ( 8- Mette à jour le contact d’un client durant un appel sortant ) http://switzernet.com/3/public/110523-astrad-wish-list/ Mais avec notre version actuelle on ne peut pas savoir le port utilisé durant un appel sortant, ni dans les attribut de la Channel en question ni dans les SIP-Headers (URI est incorrect à cause du NAT). Gets SIP peer information Doesn't work with RealTime. Description: SIPPEER(<peername>[:item]) SIPPEER(<peername>[,item]) - for Asterisk 1.6 Valid items are: ip (default): The IP address. port: The port number. (1.6) mailbox: The configured mailbox. context: The configured context. expire: The epoch time of the next expire. dynamic: Is it dynamic? (yes/no). callerid_name: The configured Caller ID name. callerid_num : The configured Caller ID number. callgroup: The configured Callgroup. (1.6) pickupgroup: The configured Pickupgroup. (1.6) codecs: The configured codecs. status: Status (if qualify=yes). regexten: Registration extension limit: Call limit (call-limit) busylevel: Configured call level for signalling busy (1.6) curcalls: Current number of calls. Only available if call-limit is set language: Default language for peer useragent: Current user agent id for peer codec[x]: Preferred codec index number 'x' (beginning with zero). accountcode: value accountcode field of CDR records for calls coming from this peer http://www.voip-info.org/wiki/view/Asterisk+func+sippeer |