Tom Waterman
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Posts posted by Tom Waterman
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Our research is indicating:
-a SIP brute force Botnet appears to in operation
-It is not a massive botnet, perhaps several hundred bots worldwide (our estimation)
-very few bots in the USA
-user agent = "Asterisk PBX"
If there is anything else that would help anyone, let me know.
Matt
As you know Matt, I am currently tracking the same thing. we are seeing hundreds of hits per day. There has to be a setting in the PBX to block them permanently. I saw one today that got an extension on the second try. Good think I have secure passwords!
Tom
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Hello all, our PBX has blacklisted about 600 IP addresses in the last 4 days. My current settings are for every 5 attempts in a 2 second span blacklist for 7 days. My question is how can I blacklist them permanently without having to type them in manually? And is anyone else seeing an increase in failed registration attempts?
Thank you,
Tom
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This is going to be a strange question so bare with me while I set the scenario up. We have a PBXnSIP here at the office with conference rooms. We have teams that travel to install our own software. Instead of have 6-10 people call in here we would like to set up a Mobile PBXnSIP on a laptop. That way the remote team can register their sip soft phones with the remote PBX and call out as needed. Many times we have no cell service in the buildings we work in. Due to bandwidth concerns at the remote site we would like to have only 1 RTP stream between the 2 sites. So we would need a conference room(or any other kind of extension) at each site and a way to link the 2. Any thoughts? Please.
Thank you in advance,
Tom
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I've downloaded and installed from the Droid market the program called "REMOTE WAVE" from Walter Yongtao Wang Version 1.7.7 and it automatically ties intself into the audio system and when I click Open in a WAV attachment I choose and select always use remote wave and Voila, it works perfectly. While I would hope the Droid OS addresses this, a $00.99 solution is more than acceptable.
The Droid OS will fix this bug in the next release. I believe it will be 2.3. It worked prior to version 2.2. Then I upgeaded to 2.2 and it is broken. It is not just the PBXnSIP files I can't play anymore it is my Vonage voicemails as well. :-)
Tom
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You can do it with a * code... *84 you would have to do it from that phone...one downside is that it also resets the voicemail greeting and maybe the password - can't remember it's been awhile since I last did it.....I believe the *84 code "Clean up extension" is used in the "Hotel" version of the software but it's also in the default version.
Yeah I would want to avoid having the user to redo all of the other extension related items. If they can put that into the new bulk options menu that will help. They can already reboot bulk phones, clear all registrations, mac addresses and cell phone numbers from related extensions a bulk email delete should be no problem. :-)
Tom
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We do not have that in the bulk action drop-down. We can possible add in the future releases.
That would be awesome! DO I need to send a feature request in or email development, or can it be passed on through the forum? I had to delete 100 messages from a user account today and it was a pain. ;-)
Tom
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Hello all, pretty simple question here. Is there a way to bulk delete voicemails from a mailbox? I am using version 4.2.0.3925 for Windows and I don't see anything like that in the much update web gui. Thank you.
Tom
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Awesome! Thanks for the quick replies.
Tom
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Hello all, we are opening a branch office out of state. We are going to have a second PBXnSIP server to service the remote office. However there will be a point to point VPN between the offices. Each office will have its own set of DIDs. I have the following questions.
Can I set up a trunk over the point to point VPN so I can send extension to extension calls over the VPN instead of sending them out the PSTN. This should also work for calls that come in to location 1 and the extension belongs to office 2. ( I assume this is possible)
Next comes to failover. If office 1 should loose its provider calls would be forwarded to office 2 and could be routed over the trunk to office 1 (assuming the point to point is still up). This would cover me in case of a provider failure.
I think both of these should work just fine. I will need two sets of extension 8xx for office 1 and 6xx for office 2.
Do I have all of my facts correct? Thank you for the assistance!
Tom
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We are using version 3.4.0.3198 (Win32) on a windows box. When you select option 5 it says " Welcome to the auto attendant" and that is it. It is just not playing the right wave file although the xml is pointing at the correct one.
Tom
Can someone help, please? I have a customer that is getting ever more upset.
Thanks in advance.
Tom
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Which version are you using? What happens if you go directly to the option 5? do you get the same Greeting?
We are using version 3.4.0.3198 (Win32) on a windows box. When you select option 5 it says " Welcome to the auto attendant" and that is it. It is just not playing the right wave file although the xml is pointing at the correct one.
Tom
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I have a customer that uses to AA. The first one if the general AA and the second is for their tech services dept. When you select option 5 in the first AA it send you to the Tech Services AA. This was working for over a year. Now when I select option 5 I get "Welcome to the Auto Attendant" and nothing else. My custom message is not played. I have verified that the audio file still exists on the PBX and I even deleted the entire AA and recreated it, with no luck. Any help is greatly appreciated.
Tom
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No need of the registration logging. All we need is SIP call log. You can get that by enabling "Log Other Messages" & keeping the log level 7 or higher.
Any word on this? I still have a customer rolled back because of this issue. :-(
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The other workaround is to use "sip:callcentric.com:5060" as outbound proxy. The DNS A record for this includes all possible proxies, so that should work as well.
NICE work guys! We appreciate it. It was solid all weekend!
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We would like to see the wireshark for the Clarity case. Please send email to support@pbxnsip.com. It is interesting to see if the PBX sees BYE.
I just sent the email. Let me know if you need anything else.
Tom
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Stability depends largely on the usage of a session border controller by the service provider. For example, if there is no SBC, you must have a routable ("public") IP address, otherwise you will always suffer from instable registrations. IPv6 will solve the problem.
Untile then, you might also check http://kiwi.pbxnsip.com/index.php/Troubles..._Trunk_Problems.
I have tried changing the Keep alive down to 20 to see if that helps. But we are having this problem 2 VoIP providers. Callcentric and Clarity. With callcentric I loose inbound calls. Outbound will continue to work. With Clarity this issue is a little different. The calls are not clearing off thier trunks. It is like they never get the bye message. Btw Steve and the folks at Clarity are GREAT to work with. Any help would be greatly appreciated. I have wireshark captures already.
Tom
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Hi,
It is not an IP conflict because the voip network is on its own lan with no connections to anything but the phones which pull dhcp from the phone server.
I will send you access to the system now.
Any word on this? We are having the same problem now.
Tom
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Anyone having any issues with callcentric? Our PBX has not dropped registration with Callcentric in days. However about every 15 minutes or so our callcentric web account shows the pbx is not there however I can still make outbound calls just fine. So it is a one way registration lost. We have made no firewall or network changes. Thanks in advance for any help.
Tom
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any ideas.... anyone?
I just upgraded a client this morning to the new version of 4.0 and I have the same problem. If nothing is registered on an extention it works fine. But if something is it rings and then when it goes to exchange it dies. This was working 30 minutes ago before I did the upgrade. Please help!
tom
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Something got screwed up. Should be working in the final release image.
It is fixed in the new version thank you!!!
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I am sorry if the title is vague, I had no idea how to word it. Let me explain what is happening. I have a user that is telling me they are receiving a couple of calls a day that do not belong to them. They are however Option 1 on the Auto Attendant, and appear no where else in the Auto Attendant such as the default time out ext or anything. So what I am looking for is a way to look in the log and see what the remote party pressed for an option in the Auto Attendant. I have logging turned way up on the pbx and it is logging to a file. I can see the call in question come to the PBX and then go to this users ext. I need to find out how. Thanks for the help.
I can see in the log where I recieved a DTMF 1 but how do I know where it came from?
Tom
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If it only happens with one specific phone, then there must be something special about it. Mabe it is the only phone which is behind a special firewall, or is has a different firmware, or maybe you just need to factor reset it and re-provision it.
After a little more investigation it turns out this is happening on all voicemails for all extensions. Theuy get 9 seconds of voicemail and then the PBX sends a disconnect message. BYE.
I have a wireshark capture of this happening. It is version 3.4.0.3201
Tom
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In any case, try to use 8.2.29 this is the "officially" released firmware now. You can delete the pnp entries in the working directory and restart the pbx to make this happen easily.
We'll take a look anyway. Maybe something else got screwed up.
I tried what you suggested and still no dice.
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Ok so I have exchange 2007 SP1 UM completely setup and it works great. You can call into Exchange and browse mail boxes dial out all that great stuff. However when you go to leave a voice mail you have about 9 seconds to do it and the PBX sends Exchange a bye message. Right in the middle of leaving the voicemail. I have a wireshark capture from Exchange is Support would like to see it. We are using Version 3.4.0.3201 Help would be greatly appreciated.
Tom
Blacklist IP addresses permanently
in Security
Posted
Are you saying that if I have a long blacklist it will hurt system performance? If so could I just blacklist the entire outside and allow on the handfull of internet addresses I need to register? But how would that affect my voip trunk provider? Would I have to add them in as well?
Thank you!!!
Tom