Asterisk 11 with public IP call problems

Asterisk Asterisk Support
1

Hello Asterisk community,

I have Asterisk 11 installed on server with public IP address. Here is part of my sip.conf file:

[general]
externip=XXX.XXX.XXX.XXX
bindport=5060
tcpenable=yes
tcpbindaddr=0.0.0.0
bindaddr=0.0.0.0
localnet=XXX.XXX.XXX.XXX/255.255.255.0
;transport=tcp,udp
rtcachefriends=yes
disallow=all
allow=ulaw
allow=alaw
allowguest=no
alwaysauthreject=yes
trustrpid=yes
sendrpid=yes
disallowed_methods=UPDATE
callerid=unknown
context=from-outside
mohinterpret=default
nat=yes
canreinvite=yes
allowsubscribe=yes
notifyhold=yes
notifyringing=yes
callcounter=yes
videosupport=no
t38pt_udptl=no
tos_sip=cs3
tos_audio=ef
tos_video=af41
tos_text=af41
cos_sip=3
cos_audio=5
cos_video=4
cos_text=3
pedantic=no

[30000]
qualify=yes
nat=yes
pickupgroup=1
callerid=test purposes
context=from-inside
canreinvite=yes
secret=XXXXXXX
host=dynamic
callgroup=1
dtmfmode=rfc2833
type=friend
disallow=all
allow=ulaw
allow=alaw

[40000]
qualify=yes
nat=yes
pickupgroup=1
callerid=test purposes
context=from-inside
canreinvite=yes
secret=XXXXXXX
host=dynamic
callgroup=1
dtmfmode=rfc2833
type=friend
disallow=all
allow=ulaw
allow=alaw

I’m able to connect to server with different kinds of phones, as well as making calls through connected SIP-trunk without problems. The only issue I have is to make calls between local extensions. I can hear voice from one phone, but not from another. I tried different types of NAT as comedia and force_rport but it is just not working. Asterisk CLI seems to not show any problems.

I will appreciate any help. Thanks in advance.

2

You have two deprecated option (one name, one value. You have a not best practice value for type. You have localnet, which is optional, but no NAT settings (nat= does count and may not be needed).

The localnet setting suggest you have a broken dual homed configuration (not mutually routable), which would be incompatible with directmedia (you use an obsolete name for this) being set.

3

Just tested between two Cisco phones and everything was ok. I need to do more tests.

4

i think its problem of your sip codec please check properly

5

Solved. I had to put “udpbindaddr=0.0.0.0” into sip.conf’s general part. Now every kind of phone I could get my hands on as well as sip phones are working fine.