search for: alwaysauthreject

HI ALL got small question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy

...> On Thu, 2017-02-09 at 14:58 +0200, ????? ?????? wrote: > > >> so the main question is -- how to Disallow CALLS without registering >> on PBX > > sip.conf configuration > In the [general] section, define: > > > [general] > ... > allowguest=no > alwaysauthreject=yes > ... I don't think either of these configuration attributes achieve what the OP requested ("how to Disallow CALLS without registering on PBX"). In fact, I'm not sure that it's actually possible to disallow [authenticated] calls from a peer that hasn't registered!...

... using it as a tool and understanding what it does... So one part of it's toolset identifys valid SIP accounts - and I was under the impression that alwaysauthreject=yes was supposed to stop this... However, it sends a request for a highly probably non-existent account, then sends requests for probably existing accounts and I guess compares the results - account not found vs. bad username or password... It thus trivially, and very quickly finds valid accoun...

So when someone's brute forcing your server is there a way to identify the originating IPs without using a tcpdump? When I get a failed auth on the console it shows 'account at asteriskserver' then tag=as25ca5023 (or some random string, though it's a bit odd as alwaysauthreject = yes is on in sip.conf). Anyway, the logs don't show anything more useful either. Is there something obvious I'm missing? Cranking up verbosity on the console doesn't seem to do anything. hose

Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Dup...

Hello all. I was recently the victim of a SIP flood attack. I'm wondering what is the best method to prevent such things in the future. Many thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101003/2e254523/attachment.htm

...he attack continues to guess only passwords. Of course, they can not guess passwords like p(F9j43/Qgrhjv*&^3 so i'm still not worried, but this made me believe that asterisk responds differently when probing a valid sip peer name. So i was wondering through the sip.conf and found 'alwaysauthreject' which was set to default (commented out). I now set its value to yes (which i thought was the default setting). Does this setting makes the attacker believe that the first try of sip peer name was valid, but only the password was incorrect? So in this case should they stick to the first...

...| header. If the peer does exist the second REGISTER will | | | receive a response of "403 Authentication user name does | | | not match account name". If the peer does not exist the | | | response will be "404 Not Found" if alwaysauthreject is | | | disabled and "401 Unauthorized" if alwaysauthreject is | | | enabled. | +------------------------------------------------------------------------+ +--------------------------------------------...

...oesn't work. Using udptl debug, I can see packets between Asterisk and both sides (SPA112 and PSTN fax) but it seems that faxes can't agree how to send image. == sip.conf: [general] tcpenable=yes videosupport=yes transport=udp,tcp dtmfmode=rfc2833 qualify=yes directmedia=no allowguest=no alwaysauthreject=yes rtcachefriends=yes rtupdate=no callcounter=yes t38pt_udptl=yes,redundancy,maxdatagram=200 t38pt_rtp=no t38pt_tcp=no ignoresdpversion=yes disallow=all allow=alaw allow=ulaw externip=82.200.7.184 localnet=192.168.0.0/255.255.0.0 [mtt] type=peer host=80.75.130.136 fromuser=74957777777 disallow=al...

Hi all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic secret=123abcd context=internal [7002] type=friend host=dynamic secret=456abcd context=internal Am using linphone as sip client and create account on linphone with user name 7001...

...-- SIP/sipgate-0000014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register => SIP-ID:SIP-Password at sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all...

...p.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration[general]context=internalallowguest=noallowoverlap=nobindport=5060bindaddr=0.0.0.0srvlookup=nodisallow=allallow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=<IP>[7001]type=friendhost=dynamicsecret=123context=internal[7002]type=friendhost=dynamicsecret=456context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin nee...

Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples:

Is there a way to drop a ip connection to asterisk after a number of register attempts. I have been having issues with hackers doing registration scanning against our server. We block their address at the fire wall but since asterisk does not force a drop of the connect after so many bad reg attempts I can't enforce the block until they drop and try again. This allows them to run the box

...in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud. A typical sip.conf phone configuration on the remote server for the site is [general] session-timers=refuse disallow=all allow=g729:20 allow=ulaw allow=alaw fromuser=xxx useragent=xxx callcounter=yes alwaysauthreject=yes allowguest=no jbnable=yes jbforce=no jbimpl=adaptive jblog=no jbmaxsize=200 jbresyncthreshold=1000 externaddr=xx.xx.xx.xx localnet=xx.xx.xx.xx/255.255.255.0 [xxxx] type=peer user=xxxx secret=xxxx host=dynamic disallow=all allow=g729 allow=ulaw allow=alaw dtmfmode=rfc2833 context=xxxx call-limi...

We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were

...y quickly using up my 448 kbps upload limit for my home ADSL connection: any further traffic (i.e. anything I did) was then experiencing significant packet loss. Anyway, I've now implemented the "7 steps to better Asterisk security" that I found on the Digium website (deny/permit, alwaysauthreject etc.), and have been looking at fail2ban. However, when I attempted to install it (following the instructions I found on a page about fail2ban with Asterisk), I ran into a couple of issues. FWIW, I'm using Asterisk 1.4.21.2~dfsg-3+lenny1 on Debian. First, I tried uncommenting the line in...

...you can take to harden your systems. An internet search will bring > up many, but here are a couple of good ones: > > http://blogs.digium.com/2009/03/28/sip-security/ > http://www.ipcomms.net/blog/70-11-steps-to-secure-your-asterisk-ip-pbx > http://nerdvittles.com/?p=580 OK, I set alwaysauthreject = yes and I discovered a allowguest, which I set to "no", too. The PBX is behind a Firewall and I just allow UDP 5060 and 10000-10100. Now I log the SIP-pakets coming from Internet, too... Hopefully I solved my problem... Thanks Luca Bertoncello (lucabert at lucabert.de)

...9;m thinking there must be something wrong with my Asterisk configuration yet I've made no config changes that would account for the sudden (and consistent) incoming call failures. Here's the relevant portions of my sip.conf if it helps (with credentials and ips replaced by Xs): [general] alwaysauthreject=yes dtmfmode=auto disallow=all allow=ulaw register => XXXX:XXXX at XXX.XXX.XXX.XXX:5060 register => XXXX:XXXX at XXX.XXX.XXX.XXX:5060 [101] type=friend context=websage host=dynamic deny=0.0.0.0/0 permit=XXX.XXX.XXX.XXX/24 qualify=yes secret=XXXX mailbox=101 at default accountcode=101 I...

...es through just fine. So my hunch is that is something to do with the audio supplied by the server. Do I need to have alsa installed?? Any hint? sip.conf: [general] context = unauthenticated bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes videosupport = no textsupport=yes alwaysauthreject=yes allowguest=no [1001] ; grandstream 1 context = home type = friend callerid = One <1001> secret = XYZ host = dynamic mailbox = 1001 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport [1005] ; mob...