Hi
I have installed a asterisk server on raspberry (raspian).
I have created 2 internal number and one external number voip with messagnet as voip provider.
I can do and i can received call normally from number voip of messagnet, but when i type asterisk -r
i see a lot of message like this:
[Sep 1 00:33:33] NOTICE[855][C-000003e3]: chan_sip.c:26241 handle_request_invite: Call from ‘’ (89.163.146.93:5088) to extension ‘000972592634285’ rejected because extension not found in context ‘sipin’.
[Sep 1 00:33:41] WARNING[855]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission ef39a8721626a546568f168f7986445c for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
raspberrypi*CLI>
[Sep 1 00:34:12] NOTICE[855]: chan_sip.c:28468 handle_request_register: Registration from ‘“2451” sip:2451@82.48.244.33:5060’ failed for ‘195.154.55.160:5064’ - Wrong password
[Sep 1 00:34:18] NOTICE[855]: chan_sip.c:28468 handle_request_register: Registration from ‘“2551” sip:2551@82.48.244.33:5060’ failed for ‘195.154.55.160:5079’ - Wrong password
I can’t understand these warning, especial the last (wrong password).
thank you for your support.
Looks like you have 2 phones trying to register. They’re both failing because the passwords they’re sending don’t match the server’s sip passwords.
For the not found one, you will need to provide the contents of the sipin context, or if that is the wrong context, the contents of the sip.conf.
For the retransmission one, that is either a broken peer,(for which you will need to toll us something about that peer, or a NAT or firewall issue, for which you should provide sip.conf, and, at least the Contact headers from the INVITE you last received, together with the Contact header from the final response you sent to that.
Just make sure it not someone else who is trying to make calls from/hack your system.
Do you have allowguest set to yes in sip.conf?
Is this your IP, 89.163.146.93?
–Satish Barot
[general]
srvlookup=no
port=5060
udpbindaddr=0.0.0.0
context=sipin
tcpenable=no
bindaddr=0.0.0.0
nat=yes
disallow=all
allow=alaw
rtptimeout=60
rtpholdtimeout=300
register => numer of messagnet:password@sip.messagenet.it:5061/numero messagnet
[ext-provider]
type = peer
username=internal number of messagnet
fromuser=internal number of messagnet
secret=mypassword of messagnet
host=sip.messagenet.it
fromdomain=sip.messagnet.it
qualify = yes
insecure = invite,port
[office-phone](!) ; create a template for our devices
type=peer
context=from-internal
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw
allow=alaw
this is my sip.conf files
and i have open port 5060 on my router
89.163.146.93 yes this is my ip public and this 195.154.55.160 is a server located in France that probably is an server of messagnet.
the number 000972592634285 is most definitely a hacking attempt - they all start like this
really? how i can stop it?
If it’s a public IP, then the short answer is, you can’t. There are definitely some things you can do to protect yourself though. Use complex passwords for every endpoint/phone, do not allow outbound dialing from your default context, register your known sip endpoints to a different context than default (as you have), use Fail2Ban, change default ports (obscurity, but may work with dumb hackers), also be sure your firewall is patched and blocking ssh login access. Only allow access to ports you’re using for sip/rtp.
I strongly suggest you read asterisk documentation about security… just google for it, there are plenty. otherwise, you will end up with a hacker using all your credit at your telephony provider.
First thing is to close port 5060 for incoming connections on your firewall, If you have remote clients connecting to your server, you should allow explicitly their address.
J.