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15

I found the solution after spending roughly 40 hours on this problem. There is a setting in the switch that enables "Auto DoS" protection. Apparently it considers TCP or UDP traffic that has matching source or destination ports to be a blat attack and drops the packet. This is ridiculously short-sighted since SIP traffic often (always?) relies on source ...


12

All the answers to your questions are "it depends". If you use a PBX system like Asterisk where the audio data is actually handled by the server computers you'll have much steeper CPU and I/O demands on the server computer (along with finicky reliance on timing-- something that virtual machines don't necessarily do a great job with). If you a PBX system like ...


9

nmap -v -sV localhost -p 5060 will tell you the truth. You can also use: netstat -apnt | grep 5060 or ss -aln


9

A good rule of thumb comes from old telephony .. an phone T1 line 24 voice channels and a control channel. This equates to roughly 64Kb/s per phone line. This rule works (roughly) for uncompressed VoIP. There are various codecs that do compression .. the G729 should roughly triple that, so get 72 voice channels on a T1. If you expect all 55 phones to be ...


9

Virtualizing a PBX is a challenge, due to one main aspect: There is no guaranteed scheduling of your PBX VM and the general scheduling behavior could introduce jitter. That being said you also have to think about how your line-cards (if you need S0 to some other PBX etc.) need to be presented to the VM and if things like vMotion and HA make sense. There are ...


7

I have come across the same problem and it's not just limited to asterisk. In the end we came up with something that worked well for us. We called it a SIP Loopback. Basically we signed up for an ALT sip provider (flowroute.com) and setup a script that calls out via primary SIP provider to the phone number setup with our ALT provider three times an ...


7

lsof -i:5060 will not only show if it is open but what its actually doing. Example: root@root.com# lsof -:5060 COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME asterisk 1146 root 18u IPv4 0xffffff000a053c60 0t0 UDP *:sip asterisk 1146 root 18u IPv4 0xffffff000a053c60 0t0 UDP *:sip asterisk 1146 root 18u IPv4 ...


6

VOIP. Voice over IP. It requires quite a few more things to be configured than plugging a phone into ethernet, but that puts you in the neighborhood.


5

Phone numbers and "SIP URIs like me@domain.com" are different kinds of URIs. A phone number can be represented as a URI like this: tel:+12125551212 whereas a "SIP URI like me@domain.com" can be represented as a URI like this: sip:me@domain.com Many SIP user agents don't actually use tel: URIs. Instead they just take the number that the user dialed and ...


5

SIP is IP, so yes it's routable. That said, it is tricky to get it to play well with NAT. Verify that your firewall rules are ok, both for signaling (5060) and RTP (most cases 10000-20000).


5

The publicly available SipVicious script that many of these attackers use stops the attack instantly if it receives an invalid SIP response with no From: line. You can identify SipVicious because it sets its User-Agent in the SIP requests to friendly-scanner. Using this technique against a real-world attacker, I have been able to immediately stop the flood ...


5

To keep this on-topic, and not make this opinion-based I'll just speak to the technologies you mention in your question. re: managed versus unmanaged An unmanaged switch gives you no functionality to monitor the switch. You can't see packets sent / received, error counts, etc. They're just a "black box" that moves packets. Typically you don't get virtual ...


4

If you have the bandwith (roughly 100 kbps per call), G711 should give you best call quality (and least CPU usage in case Asterisk is transcoding). Otherwise, G729 is a good choice. These two should be supported by most hard- and softphones. For more exotic codecs, check what your phones support.


4

Trixbox is pretty good (http://www.trixbox.org/). It's basically Asterisk but shake'n'bake style. Their community edition is a free version which installs Asterisk plus some fancy front-end tools among other things. It has IVR (auto-attendant), SIP support, remote extensions and other cool stuff. Try it out.


4

If you can keep your Asterisk PBX in the game, it looks like Skype is interested. The Skype for SIP Beta program maybe what you're after. This way, any SIP or IAX2 client may have Skype incoming and outgoing access.


4

The only thing I can think of is if your router has a SIP ALG and is crashing because of some bug in it when it doesn't get the ACK. If your router does have a SIP ALG you should try and find a way to turn it off. It's generally accepted now that SIP ALG's are bad news and cause far more problems than they ever solve. Update: If you're writing your own SIP ...


4

If you already have Asterisk running, setting up a task to dial a number and wait for the other side to answer and play an audio file is not that hard. But 'having Asterisk running' does not answer 'simple'. Google suggests pjsua.


4

The system is waiting for further input to determine whether the complete extension is '1' or something else that starts with '1'. Do you have other extensions in that context that start with the number 1? Check your dialplan.


3

I have looked at quite a few of the soft phone clients one of my favorites is x-lite http://www.counterpath.net/x-lite-download.html I don't like that you can't transfer (with the free version) but it seems to be a very solid client. You should also check out this comparison http://en.wikipedia.org/wiki/Comparison_of_VoIP_software


3

I have had good success using ZoIPer (formerly idefisk) under both Linux and Windows. One major advantage of that client is that it supports both IAX and SIP. The free version supports two line registration profiles and they have paid versions as well.


3

We have run into similar issues before and it is usually due to application layer gateways on the firewall or IPSec devices. These ALGs assist traffic (like SIP or FTP) that use dynamic ports in connecting through NAT devices, and they can also verify that the traffic conforms to protocol specifications. In my experience, if you are not doing NAT over the ...


3

An "unregister" is a REGISTER wherein you set the expires for one or more Contact URIs to 0. You may do so either with the Expires header - "Expires: 0" - which applies to all Contact URIs in the REGISTER, or with a parameter - "Contact: ;expires=0 - which will only affect that URI's registration. You may also unregister ALL contact URIs for your ...


3

You should do basically 2 things: Setup Asterisk server to allow proper registration of your SIP account. This is done configuring the SIP credentials at /etc/asterisk/sip.conf Configure Asterisk dialplan to map extension 55 as a dialout to your cellphone. This is done at /etc/asterisk/extension.conf You will find extensive documentation about how to do ...


3

What model is the Cisco? Depending on the Cisco hardware involved it might be easier to just terminate the RRAS there rather than trying to pass it along as SIP to some sort of virtual modem bank. But you didn't supply a model number for the existing hardware. Updated based on comment providing model information: I'm not sure I see any good reason to ...


3

With sipcmd https://github.com/tmakkonen/sipcmd you can do it on one line: sipcmd -u <login> -c <passwd> -P sip -w <sipproxy> -x "c<phonenum>;ws3000;v<audiofile>;h"


3

http://www.mitel.com/resources/242_9113-Mitel3300ControllersDataSheet.pdf Looks like it supports both SIP endpoints and SIP trunks. (which would be SIP from an ISP, yes)


3

I've seriously considered this in the past and come up with one very good reason not to virtualize. What happens if you want to connect your PBX to the standard PSTN network? Since doing so requires custom hardware it makes sense to avoid going virtual. This has the added benefit of if your SIP provider craps out you still aren't totally out of business.


3

I was thinking since these phones generate SIP traffic, I could capture that traffic for later analysis. TOTALLY wrong problem as the SIP part does not contain any audio. SIP = Session Initiation Protocol. It controls the cahnnels, but not the content. YO u need to use a SIP server that intercepts and rewrites all SIP and forces the traffic through ...


3

I asked the same question in the Asterisk IRC channel on freenode, and one of the developers responded indicating that in newer asterisk versions (I assume 1.8 and above) you need to specify callcounter=yes in your general config of sip.conf. He also mentioned that call-limit should never be zero, as that is not technically a valid option for that field. ...



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