Hi,
I have asteris installed and it is working fine. i just wanted to confiure the same Audiocode mp series then i came to audiocode only rfc3261. So our asterisk 1.2 supports rfc3261? if not what i need to make this up to support rfc3261?
thanks
rfc3261 is AKA sip, or session initiation protocol. While asterisk isn’t 100% RFC compliant, it is very damn close and should work with your audiocodes gateway without any problems. You need to configure sip.conf with an entry for your gateway.
Hi,
shall i include a entry like the following my sip.conf for the ext.
1]
type=friend
secret =blak
host=dynamic
qualify=yes
conext=test
dtmfmode=rfc3261
canreinvite=no
allow=ulaw
if not , can you please advice , what other entries to be made in the same.
thanks
you almost have it…
first start with a [ for the header
no reason to use qualify if you’re on the same LAN (qualfiy makes * ping the device every now and then with a useless command)
dtmfmode is inband (tones transmitted in audio stream, needs ulaw or alaw), info (sent as SIP INFO messages) or rfc2833 (attached to RTP stream). You probably want rfc2833.
hi helix,
you mean to say it is ok to use rfc2833 in extensions.conf for rfc3216 compliant Audiocodes! or i need to use rfc2833 as dtfmode in extesnions.conf to audiocodes compatibility?
hi helix,
i did not get you answer, can you please clear the same?
sorry for the delay.
SIP (the protocol) has many RFC’s associated with it. 3216 defines how most SIP things work. The fact that you are using a SIP channel configured in SIP.conf means that you are using 3216.
RFC2833 governs transmitting DTMF tones over RTP audio (dialing after the call is setup, ie for voicemail system). Using RFC2833 for dtmfmode is very compatible with RFC3216, its sort of a helper RFC. If you are supposed to use RFC2833, Inband or Info for your dtmfmode is dependant on how the gateway is setup, it probably supports all three.
short answer:
Using RFC2833 for dtmfmode does not make Asteirsk any less RFC3216 compliant. This should work well.