Good morning, afternoon, evening …
I am installing an asterisk at home and testing with an IP phone. In particular I have version 1.6.2.13 and the IP phone is a Linksys SPA922.
I managed to install and works perfectly except that the call drop after a few seconds. After much research I discovered that there is a parameter (rtcpinterval) in rtp.conf , well changing this parameter the call lasts exactly the number of seconds indicates that line.
I show my configuration if it can be helpful:
2004 Softphone
301 IP Phone
SIP.CONF
[general]
context = general
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
videosupport=yes
[authentication]
[2004]
type=friend
context=default
username=2004
secret=2004
host = dynamic
mailbox=2004
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
[301]
type=friend
context=default
username=301
secret=301
host = dynamic
mailbox=301
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
EXTENSIONS.CONF
[general]
static=yes
writeprotect=yes
autofallthrough=yes
clearglobalvars=no
priorityjumping=yes
videosupport=yes
[default]
senddtmf=yes
exten => h,1,Hangup
exten => 2004,1,Dial(SIP/2004,30,Tt)
exten => 2004,n,Hangup
exten => 301,1,Dial(SIP/301,30,Tt)
exten => 301,n,Hangup
exten => 9000,1,NoOP(9000)
exten => 9000,n,Dial(SIP/2004,30,Tt)
exten => 9000,n,Hangup
I thank you in advance for any help or suggestions you can give me.