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
