Hi,
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)?