Got Sip Response 423 "Interval too Brief" from ip

Asterisk Asterisk Support
1

I am new to asterisk and am running aah 2.8. I am trying to connect to a new provider that originally provided me with a grandstream 486 ata. Everything works well with the ata but when I try to go to the asterisk server I get
Got Sip Response 423 “Interval too Brief” from ip addr.

I contacted the carrier and they said their registration time should be greater than 1 hour. Their ata also shows that they are using a stun server.
Any help will be greatly appreciated.

sip config is as follows:

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.

[general]

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
externip = jam.rr.com
localnet=192.168.1.0/255.255.255.0
nat=yes

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
#include additional_a2billing_sip.conf

SIP ADDITIONAL CONFIG:

register=GS9081508@67.96.XX.XXX:5060

[318-908-1508]
username=GS9081508
user=phone
type=friend
nat=yes
insecure=very
host=67.96.XX.XXXfromuser=GS9081508
fromdomain=67.96.XX.XXX
dtmfmode=inband
dtmf=inband

canreinvite=no
authname=GS9081508

[5555]
username=5555
type=friend
secret=5555
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=5555@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=kenny <5555>

[5556]
username=5556
type=friend
secret=5556
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=5556@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=test <5556>

thank you
Kenny

2

set the maxexpiery to be more than 3600, also try disabling NAT. As long as you have externip= and localnet= set, NAT shouldn’t be needed.

3

Do I add that in sip.conf or sip_additional.conf?

4

ahh, a trixbox user.
see if there is a way to change the sip registration timeout from the trixbox web gui. If not, then it should be at the top of sip.conf, defaultexpiery= and maxexpiery=. set default to 300, max to 4000 or so.

5

not trixbox but Asterisk At Home 2.8.
I am finding that sometimes frills are more difficult to manage than a regular asterisk install over centOS.

6

i agree with that. either way the result is the same- you want to turn up the max expiery for SIP. this is probably in the gui screen that configures the sip provider…

7

The frills sometimes puzzle me. Most folks that recommend fixes say add code to sip.conf or exten.conf but when I look in those files there is limited code there, although there is a reference to other files like exten_additional.conf and sip_additional.conf. It is like aah or trixbox has added files to the database.

I get quite confused sometimes. :smile:

Kenny