Tim,
All I can say is that I have been running 4.04 (yes, I should upgrade to the latest version) for over two years using Vitelity without a hitch. Yes, on occasion I have experienced authentication issues with the ITSP – all total about 15 times in the last two years – and in every case I’ve traced the problem back to my ISP, not the ITSP. I’ve also experienced some dropped audio but, again, I’ve traced the problem back to my ISP, not the ITSP. My system runs in a VM on an IBM eServer 226 with two 2.8Ghz Xeon processors and 8GB RAM. I support five Polycom desk phones, one Linksys 2102 ATA, and four softphones. I regularly have two concurrent calls via Vitelity. I support around ten DIDs on this system. I don’t do anything flashy on the system – voicemail, several auto attendants, page, call park, MoH. Everything has been rock solid. I’m not an IT guy some of the others on this list. I set SipX up from ISO and it just runs. But, I read Michael’s book before setting up my system, and have spent years watching this forum to glean insights and tips so that on the few occasions where I adjust a setting, I have a pretty good idea how to do it and the possible outcomes. If I were to make three recommendations about implementing SipX, they would be: 1) use quality hardware and phones and size appropriately; 2) if reliability and availability are critical to you or your clients, invest in PRIs, an MPLS circuit, T1 or other connectivity designed to give you 99.99% uptime; 3) Install from the ISO, let SipX handle its own DNS, and make sure that you get your pubic DNS records correct. I’m sorry that you have had a bad experience with SipX. I haven’t had the same experience. -Paul Sage Logo Paul Herron Principal Sage Craftsmen, LLC [email protected] | 202-570-7030 www.sagecraftsmen.com Happy is the home that shelters a friend. -- RALPH WALDO EMERSON From: Tim Ingalls [mailto:[email protected]] Sent: Thursday, February 16, 2012 9:03 PM To: sipx-users list Subject: [sipx-users] sipXecs Commercial Feasibility Hi Everyone. I promise I am not trying to be a troll. I have some serious questions that I hope I can ask honestly and get some honest feedback about using the free version of sipXecs as a commercial product. I implemented sipXecs about a year ago. My hope was to find something more reliable than Asterisk/Trixbox/FreePBX and easier to deploy. My purpose was to start selling phone systems and SIP trunking to businesses as a VAR. So far, after testing the system day in and day out as my home/home-office phone system, I haven't found it stable enough to feel comfortable selling it to customers. I have had a host of issues with sipXecs, and every time I think I've got the platform stable, something else fails and I get one of those barely-descriptive error messages in my email inbox. I've followed the instructions from the book, the wiki, and this forum, but still have issues every month. Some of the issues are as follows: * Routing inbound calls to an auto-attendant worked great for a long time and then just stopped working one day. After connecting the call, visitors were greeted with no sound at all. I decided after hours of trying everything to just skip the auto-attendant and deactivate it. * With both Vitelity and Voip.ms, I have problems where periodically an authentication request is rejected. Instead of re-trying immediately, sipXecs waits a full 10 minutes to try to connect again. * Nuances about how certain ITSPs (e.g., Vitelity and Voip.ms) work, and how you can and cannot connect to them without getting strange behavior like inbound audio not working, rejected authentication requests, etc., take days and weeks to isolate sometimes. These are not very well tested nor documented. I think that a serious effort at interop testing and certification should be undertaken with detailed results --warts and all-- posted so that someone can make an educated decision when selecting an ITSP to use with sipXecs. * Just a few days ago, calls that were transfered to voicemail resulted in the call failing and the ITSP routing the call to my failover phone number (my cell phone) -- this is after the call initially rang correctly. Rebooting the system fixed it for some reason. Why? * Periodically, (perhaps due to a sipviscious attack) certain services just stop working. Sometimes it is the proxy service. Sometimes it is the registrar service. Sometimes it is the NAT traversal feature as a result of temporarily not being able to reach the STUN server assigned (since there is no back-up STUN server setting). Why should these services just fail and require human intervention to restart them? Can't they just time out for a certain short period and then fix themselves? * CID doesn't work reliably. I change all of the settings as I'm told in the wiki, but it still doesn't get transmitted correctly (or at all). For some of my users, it works flawlessly, and for others it doesn't work at all. * Doing a SIP trace to isolate an issue is a pain in the neck. In Asterisk, all you have to do is type "asterisk -rvv" and you can see a dialog stream which you can read quickly. With sipXecs, you have to run a series of research tasks to find the call in question, convert the time to UTC, grep for the time stamp in a big list of calls, then create a merged XML file, then load it into SIPViewer, and then find what you are looking for. The process takes at least 5 minutes if you are an expert. Those are just a few examples. I'm always wondering what is going to go wrong next. It drives me (and my wife and kids) crazy. I never had this many problems with Trixbox. I'm not saying that sipXecs doesn't have its good points. I'm just saying that over the last year+ since I started using 4.2 and then 4.4, it has been anything but reliable. Reliability is the number one need for commercial clients. Yes, I'll admit that it could all be my fault. It probably is. But there are so many options, so many opinions, so many sources of information, (there are even so many places to set port numbers for various things) that it seems you have to do only sipXecs development for a living to be able to deploy it correctly. It is far from simple. And that complexity is part of the problem. I know that some of you have deployed many of these systems in a commercial setting, so I have to ask you, how do you do it? I'm too afraid that if I deploy sipXecs in an actual customer's location that they'll hate me within a few months and ask for their money back. How do you set everything up (selection of ITSP, etc.) so that the system is rock-solid reliable? Can we collect some rock-solid fool-proof (as much as possible) recipes that are known to work reliably every time? This seems to be something that should be placed on the wiki. I know that there are 100+ ways to configure the system (SIP trunking gateway configs, various hardware, ITSP settings, dial rules, etc.). I'm looking for just the recipes that make the system reliable. I also know that there are various conflicting opinions on this forum about what works and what doesn't. I'm looking for PROVEN opinions. This is my final shot before I give up on the platform. I'd even be willing to partner with someone who has a near-flawless system implemented and pay you to do the technical part if you can prove your solution is stable. Until I find the answer to this problem, I can't use sipXecs as the cornerstone of my business plan and will have to move on. If I can solve this issue, I'd be willing to pay for further development out of my profits. I know someone will suggest that I should just sell Ezuce's commercial products. Based on what I've experienced so far, I don't think I'd feel confident in relying on Ezuce to be the partner in question. If the open-source version has these problems, what's to say that the commercial version is any better? Does anyone else experience the same reliability issues? Also, is anyone willing to have a phone conversation about this and impart some wisdom or have a partnership conversation? -- Thanks, Tim Ingalls Shared Communications, Inc. 801-618-2102 Office
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