I have configured asterisk 11 in Ubuntu 12.04.My sip.conf is like this
[general] context = unauthenticated ; default context for incoming calls allowguest = no ; disable unauthenticated calls srvlookup = yes ; enabled DNS SRV record lookup on outbound calls udpbindaddr = 0.0.0.0 ; listen for UDP requests on all interfaces tcpenable = no ; disable TCP support subscribecontext = default [office-phone](!); create a template for our devices type = friend ; the channel driver will match on username first, IP second context = LocalSets ; this is where calls from the device will enter the dialplan host = dynamic ; the device will register with asterisk nat = yes ; assume device is behind NAT ; *** NAT stands for Network Address Translation, which allows ; multiple internal devices to share an external IP address. secret = 12345 ; a secure password for this device -- DON'T USE THIS PASSWORD! dtmfmode = auto ; accept touch-tones from the devices, negotiated automatically disallow = all ; reset which voice codecs this device will accept or offer allow = alaw ; which audio codecs to accept from, and request to, the device allow = ulaw ; in the order we prefer ;allow=g729 allow = speex allow = gsm ; define a device name and use the office-phone template [0000FFFF0001](office-phone) ; define another device name using the same template [0000FFFF0002](office-phone) [0000FFFF0003](office-phone) [0000FFFF0004](office-phone)
Now I have 2 SIP phone capable of allowing GSM,G.711(aLaw/uLaw),Speex.So my problem is if I generate a call to one of this SIP phone via Asterisk server,is there a way to detect the type of codec that has been used in SIP phone(callee)?