A familiar refrain here is: Use good phones. Use good switching gear (and a vlan for voice). If you are supporting trunking and/or remote users have a thorough understanding of your firewall to ensure it is compatible or use a SBC that can deliver the desired results.
Pick and choose your ITSP carefully too. If trunking is too porblmeatic, switch to PSTN service from the local phone company, it still works with sipx. I think unless you are VERY well rounded in telephony and SIP, reselling trunking will get you in over your head real fast. Many of the problems you list are likely related to LAN or firewall configurations. Some of what you are saying is strange to me (port numbers), because unless you really need to go outside the box you don't really have to change those things. Don't let anyone tell you voip is easy. Don't expect to start doing this without a primer in telephony functions too. Read the book. On Fri, Feb 17, 2012 at 5:02 AM, Michael Picher <[email protected]> wrote: > Tim, > > I think the message here from all involved is that cutting corners gives you > the results you have experienced. > > A VoIP system will quickly point out the problems with your network. Good > network switches, properly configured are important to larger installations. > > Cheap phones stink, my recommendation for now is to use Polycom. We're > trying to get a few others to the table but they aren't there yet with their > plugins. > > With tier 3 players for ITSP connections, you get tier 3 service. If you > want top quality voice connections, get a top quality ITSP on a dedicated > link or get traditional service and bring it in with a gateway. > > Thanks, > Mike > > On Thu, Feb 16, 2012 at 9:02 PM, Tim Ingalls <[email protected]> wrote: >> >> 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 >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > -- > Michael Picher, Director of Technical Services > eZuce, Inc. > > 300 Brickstone Square > > Suite 201 > > Andover, MA. 01810 > > O.978-296-1005 X2015 > M.207-956-0262 > @mpicher <http://twitter.com/mpicher> > www.ezuce.com > > ------------------------------------------------------------------------------------------------------------ > Hope to see you at the sipX CoLab! http://www.sipfoundry.org/sipx-colab > A gathering for - open source users, eZuce customers & eZuce partners > Get the inside track on 4.6 and a glimpse at the future of sipXecs! > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
