SOLVED Flowroute absolutely awesome

tiggerpaws

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I have tried EVERYTHING with this companuy,
whilst I admit that their customer service is good,
they seem like they try, but they just "hit a brick wall"
in the end, their actual phone service is now WORTHLESS.

They just blame it on whatever software you are running
at the time, or configuration, I think they changed something
and now they don't want to help you because the software
you are running does not comply with the stupid little voip
adaptors they probably now sell.

It was working just fine several months ago and JUST BROKE,
just broke, did not do a thing to my old server, so figrued
might as well put up new software with PIAF, STILL DIDN'T WORK.

Everything else, vitelity works GREAT,
Google voice GREAT.

flowroute = SUCKS
did4sale = SUCKS
myvnumber = SUCKS

There is no reason to get a 404 incoming call error
in flowroute and one way audio on myvnumber when
there isn't a stinking thing wrong with PIAF or
our router!

Might as well just stick with Vitelity and Google voice
they are the only ones that work.
 

billsimon

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I would love to hear their side of the story on this one.
 

wardmundy

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A service that's "worthless" to some is not necessarily "worthless" to others. There's always some room for "operator error" in the equation.

briankelly63 is a big user of FlowRoute so perhaps he can chime in as well. At one time, he was working on obtaining a FlowRoute special for PIAF users, but I haven't heard any more about that in several months.
 
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I hear ya! I have never liked this "black box" concept we deal with in the VOIP world. With an old POTS line you have battery, you get dialtone, dial DONE.

In the VOIP world we get lulled into a sense of simplicity in the sense that we still get to use this thing that looks like a phone, we get dialtone, dial and then..... THE FUN BEGINS. It's an elaborate hoax!

We tend to forget that there is a tremendous amount of activity going on it the background related to SIP message, headers, IP packets on different ports all going on through miles of "cable" owned and managed by a hundred different people.

My issue is that we never know what's at the other end, whats changed, which route is being used, where the issue is without doing extensive, time consuming and complicated troubleshooting. I wish I had a nickel for every hour members of this board spent on debugging Google Voice problems! I'd be able to afford a secretary to type this post.

Hat's off to Anveo Direct for allowing control of routes, carriers and providing SIP traces for every call.

In response to your issue I've noticed that each carrier, depending on the equipment they are using, widely vary in their ability to tolerate different configurations, routers and header-packet anomalies. So it often occurs that one carrier works with a given set up an another doesn't or if they make a small change your hosed.

I think if we had greater visibility of what platform and software revision was at the other end in the VOIP world we'd be able to help the industry isolate some of these issues. There is also no reason that carriers couldn't put in place test servers that could be used to troubleshoot our connection to them or like I mentioned go the Anveo route.

Hang in there....:banghead:
 
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A service that's "worthless" to some is not necessarily "worthless" to others. There's always some room for "operator error" in the equation.

briankelly63 is a big user of FlowRoute so perhaps he can chime in as well. At one time, he was working on obtaining a FlowRoute special for PIAF users, but I haven't heard any more about that in several months.

I like Flowroute but their issue is depth. They have been "working" to deliver cname lookups for years now but just can't seem to make it happen, how crazy is that. They appear to be a very small operation. I did have good results in terms of outgoing calls with them. Incoming was about the same as everyone else as those DID points of presence are all handled by the same big players.
 

billsimon

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That's an interesting analysis, Brian. Some of us fell for the hoax from the other angle. Those of us who come from an IT background look at VoIP and say, "It's just a network application!" Then we realize that not only do we have to know everything about systems administration, network administration, and a bit of programming, but also everything about legacy and IP telephony.

I have no perspective but my own on this, but I would say that it's easier to get into modern IP telephony from a computing/systems/networking background than it is from a legacy telephony background. Especially when you are doing anything more than just setting up a VoIP switch and endpoints and calling it a day.
 

tiggerpaws

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I mean I really like the Security of PIAF, it keeps the dang
crackers out, and fail2ban Stops Crackers In Their Tracks!

As far as flowroute goes, "their take" on this as someone put it, is that its "misconfigured"!
Rubbish! PIAF works out of the box so they are not fooling me,
I think they just changed their mode of business or what not and
want to sell those poopy voip adapters and don't give a stool about
us pbx users.

If anyone wants to dispute it, here is my edited
status-output.txt file and I apologise if it is too
big, I also apologise if it is not enough info so I
will provide a pastebin link instead of any more of
a wall of text.


Code:
PBX in a Flash System Information
**********************************************************************
* 1. I agree that I will review the contents of the file *
**********************************************************************
/root/status-output.txt
**********************************************************************
* and redact any sensitive information *
* 2. I agree that I am wholly responsible for the editing of the file*
**********************************************************************
/root/status-output.txt
**********************************************************************
* *
* IF YOU DO NOT AGREE WITH THESE TERMS OF USE DO NOT USE THIS FILE *
* *
**********************************************************************
Date and Time = 20130826 139
PIAF color = PURPLE
Asterisk Status = ONLINE
Dahdi Status = ONLINE
MySql Status = ONLINE
SSH Status = ONLINE
Apache Status = ONLINE
Iptables Status = ONLINE
Ip6tables Status = OFFLINE
Fail2ban Status = ONLINE
IP Connect Status = ONLINE
Free Disk Space = ADEQUATE
Free Memory = ADEQUATE
NTPD Status = ONLINE
Sendmail Status = ONLINE
Samba Status = OFFLINE
Webmin Status = ONLINE
Ethernet 0 Status = ONLINE
Ethernet 1 Status = N/A
Wlan Status = N/A
PIAF Installed Version = 2.0.6.4
Freepbx Version = 2.10.0.2
Running Asterisk = 1.8.23.0
Asterisk Source Version = 1.8.23.0
Dahdi Source = 2.7.0
Libpri Source = 1.4.12
System Verified = *VERIFIED*
pbx.local on 192.168.0.63 - eth0
CentOS release 6.4 (Final) :32 Bit Kernel: 2.6.32-358.6.2.el6.i686
********************************************************************
Ifconfig output
eth0 Link encap:Ethernet HWaddr 00:03:47:F1:30:C5
inet addr:192.168.0.63 Bcast:192.168.0.255 Mask:255.255.255.0
inet6 addr: fe80::203:47ff:fef1:30c5/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:1288229 errors:0 dropped:0 overruns:0 frame:0
TX packets:1287533 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:281271226 (268.2 MiB) TX bytes:330551399 (315.2 MiB)
 
lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:71201 errors:0 dropped:0 overruns:0 frame:0
TX packets:71201 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:10794549 (10.2 MiB) TX bytes:10794549 (10.2 MiB)
 
********************************************************************
Network Configuration output ifconfig-eth0
DEVICE="eth0"
BOOTPROTO="dhcp"
DHCP_HOSTNAME="pbx.local"
HOSTNAME="pbx.local"
HWADDR="00:03:47:F1:30:C5"
IPV6INIT="yes"
MTU="1500"
NM_CONTROLLED="yes"
ONBOOT="yes"
TYPE="Ethernet"
UUID="50e138b8-746e-499c-8fb9-dcdd4cbd2df1"
********************************************************************
uname -a output
Linux pbx.local 2.6.32-358.6.2.el6.i686 #1 SMP Thu May 16 18:12:13 UTC 2013 i686 i686 i386 GNU/Linux
********************************************************************
Routing Info - route
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
192.168.0.0 * 255.255.255.0 U 0 0 0 eth0
link-local * 255.255.0.0 U 1002 0 0 eth0
default qwestmodem.doma 0.0.0.0 UG 0 0 0 eth0
********************************************************************
Free Memory free -m output
total used free shared buffers cached
Mem: 498 471 27 0 44 171
-/+ buffers/cache: 255 243
Swap: 766 15 751
********************************************************************
Disk Info df output
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/sda2 76048472 3838404 68346980 6% /
tmpfs 255356 0 255356 0% /dev/shm
/dev/sda1 99150 25434 68596 28% /boot
********************************************************************
Output of hosts file
127.0.0.1 pbx.local localhost localhost.localdomain localhost4 localhost4.localdomain4
::1 pbx.local localhost localhost.localdomain localhost6 localhost6.localdomain6
********************************************************************
********************************************************************
 
 
********************************************************************
Dmesg output
Initializing cgroup subsys cpuset
Initializing cgroup subsys cpu
Linux version 2.6.32-358.6.2.el6.i686 ([email protected]) (gcc version 4.4.7 20120313 (Red Hat 4.4.7-3) (GCC) ) #1 SMP Thu May 16 18:12:13 UTC 2013
 
***********************************************************************
ASTERISK CORE SHOW VERSION
***********************************************************************
Asterisk 1.8.23.0 built by root @ pbx.local on a i686 running Linux on 2013-08-24 03:55:13 UTC
***********************************************************************
 

tiggerpaws

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A service that's "worthless" to some is not necessarily "worthless" to others. There's always some room for "operator error" in the equation.

I don't see what operator error that you are speaking of sir,
you have made good software, and it is hard to booger up,
it works straight out of the box ( I know, I know you don't do
DAHDI, thats another story that some kind soul helped with)
so being that it works, why would flowroute suddenly break
when I made no adjustments at all to it?

All of my Google voice trunks work great, I even ran a test
with a used for mail and all that trunk and Even That one works,
so I am sorry if I just don't see where I cood go wrong with PIAF?
 

tiggerpaws

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I hear ya! I have never liked this "black box" concept we deal with in the VOIP world. With an old POTS line you have battery, you get dialtone, dial DONE.

In the VOIP world we get lulled into a sense of simplicity in the sense that we still get to use this thing that looks like a phone, we get dialtone, dial and then..... THE FUN BEGINS. It's an elaborate hoax!

We tend to forget that there is a tremendous amount of activity going on it the background related to SIP message, headers, IP packets on different ports all going on through miles of "cable" owned and managed by a hundred different people.

My issue is that we never know what's at the other end, whats changed, which route is being used, where the issue is without doing extensive, time consuming and complicated troubleshooting. I wish I had a nickel for every hour members of this board spent on debugging Google Voice problems! I'd be able to afford a secretary to type this post.

Hat's off to Anveo Direct for allowing control of routes, carriers and providing SIP traces for every call.

In response to your issue I've noticed that each carrier, depending on the equipment they are using, widely vary in their ability to tolerate different configurations, routers and header-packet anomalies. So it often occurs that one carrier works with a given set up an another doesn't or if they make a small change your hosed.

I think if we had greater visibility of what platform and software revision was at the other end in the VOIP world we'd be able to help the industry isolate some of these issues. There is also no reason that carriers couldn't put in place test servers that could be used to troubleshoot our connection to them or like I mentioned go the Anveo route.

Hang in there....:banghead:


Thank you for your kind support along with the others here.

I am 99% tempted to use up all my slowroute minutes today and
port my number over to Anveo, plus they are cheaper than slowroute
for incoming office package than slowroute is, plus, who the HECK
uses AMAZON payments in chunks of 20 dollars at a time??? How stupid.

Another apology to Ward, sorry I am still learning how to use the
inline CODE thingy so my post isn't 2000 kilometres tall when I
gotta post a config file.
 

AndyInNYC

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Just my two cents.

I use flowroute on a single user, local PIAF system and a 2 user RentPBX system. Of all of the problems I have had, flowroute was never a cause or issue. Several times they were the cure - I couldn't figure out how to correctly dial certain non-US destinations and they quickly gave me the correct formatting (without any comments about my general stupidity <g>).

I have always found them to be responsive and to at least attempt to answer questions which weren't even specific to their service (i.e. the issue was user ignorance with PIAF rather than a flowroute technical question).


Andrew
 

tiggerpaws

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I still don't understand why it does not work,
they are telling me it is a 404 error, when I clearly
have an inbound route set, is there something wrong with
having any CID and any DID incoming routed to an IVR?

I do that because where else will the calls go?

I mean I have friends calls coming into my extension,
and that works with other trunks, but when someone
calls in on the slowroute line, it always give 404 error,
this does not c ompute.
 

rjaiswal

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Could you post your sanitized flowroute trunk config? I too use flowroute, for my business and for a bunch of clients. Never had an issue with them in 2 years.

I did have a 404 error once. It was due to my firewall randomizing my sip port, so my pbx was registering with flowroute on some wierd port, and not 5060. After setting my firewall to 1:1 NAT, that cured my 404 error.

In the flowroute control panel, what port is your pbx registering to? Do you even see your pbx registered in the flowroute control panel?
 

tiggerpaws

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username=
type=friend
secret=
port=5060
insecure=port,invite
host=70.167.153.130
fromdomain=sip.flowroute.com
dtmfmode=rfc2833
context=from-flowroute-com
canreinvite=no
allow=ulaw&g729
;host=216.115.69.144
 

rjaiswal

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username=
type=friend
secret=
port=5060
insecure=port,invite
host=70.167.153.130
fromdomain=sip.flowroute.com
dtmfmode=rfc2833
context=from-flowroute-com
canreinvite=no
allow=ulaw&g729
;host=216.115.69.144

From the looks of it...

Host should be sip.flowroute.com
And context should be from-trunk
 

heykoopa

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one of the issues that I possibly see is ulaw&g729 I believe the G729 codec needs to be paid for try ulaw&g722 and in asterisk sip settings turn the g722 codec on. If your doing sip registration make sure the bottom register field is filled in on the trunk settings and if you have NAT make sure that is turned on in each extension. I have my production system that's been up 2 years without issues and last week I was building a test system from memory and stumbled into the 404 error until I made those changes. Also the 404 error seemed to pop up when I dialed 7 digit and or a 10 digit but seemed to be fine on 11 so check your outbound routes and prepend your NPA info.
 

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tiggerpaws

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Interesting, now something is starting to work,
except that no matter which line I call in on,
even though the server is answering, she always
answers with the Playing 'ss-noservice.gsm' recording.

Weird.
 

rjaiswal

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Could you post the output of the asterisk cli when you try to dial your flowroute DID?

In order to see the output, you need to ssh into your piaf pbx, and type asterisk -rvvvvv

Then dial your DID and post the messages that come up when you dial.

As a test, TEMPORARILY, enable allow anonymous inbound sip. You'll find the setting, under asterisk sip settings. Once enabled, try to dial your flowroute DID.

If it connects, then it's a port forwarding issue. You'll have to disable anonymous inbound sip and see why piaf doesn't recognize the incoming call from flowroute.

Please remember to disable anonymous sip. It's a HUGE security risk by leaving it enabled.
 

w1ve

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I've been watching this thread.. tiggerpaws.... my best suggestion to you is learn from the folks here. Configuring a trunk for a provider is complex -- often you *think* you have the settings correct and you do not. So, a slight change on the vendors side and it'll stop working. I am a flowroute customer for two years, and it has been flawless. Love the fact that you can set your outbound CID (LIDB database entry).

Question: What do you have in your registration string? Is it in the format user:[email protected]/NNNNNNNNNN where NNNNNNNNNN is your DID? Your incoming route must match that NNNNNNNNNN or you'll get the message about Number not in service. By the way, no matter what your DID is, you can make that number anything. (This is only useful if you have a single DID from them. IF multiple, you must not have the /N... at the end, and you'll get whatever they supply.) Note that there are many formats that the DIDs can come in... 10 digits, 11 digits (country code included), or even a "+" prefix on the DID.

If you want to dive in and be a bit more technical, here are some of my suggestions:

I find the best way to debug is stuff is looking at what is happening live: asterisk -rvvvv and watching the output. IF you have a routing problem, that should be very evident in the output. If that goes by too fast, look at the log at /var/log/asterisk -- the current log is called "full".

Even if you don't know the SIP protocol, it is very human-readable. You know the IP address of the flowroute server.
From the command line after asterisk -rvvvv, enter: sip set debug ip IP-OF-FLOWROUTE

(sip set debug off turns it off)
You'll see From To headers with Addresses.

If you are in a NAT situation and you are not receiving audio, then look at the IP addresses in the SIP packet, and you see any private-range address (192.168.x.x or 10.x.x.x) as an actual end-point address, you know your NAT translation is not working. This is why you would have one-way audio.

Your problems are probably not vendor related. We will get this fixed.
 

tiggerpaws

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10,000 Thank yous w1ve.

It is running so far, I told it nat=yes
AND fixed some context=from-trunk
in the incoming settings place
that they had me put on context=from-flowroute-com
for some reason of using this option in the extensions_custom.conf
file:

[from-flowroute-com]
exten => _X!,1,GotoIf($["${CALLERID(num):0:2}" != "+1"]?noplusatstart)
exten => _X!,n,NoOp(Changing Caller ID number from ${CALLERID(num)} to ${CALLERID(num):1})
exten => _X!,n,Set(CALLERID(num)=${CALLERID(num):1})
exten => _X!,n(noplusatstart),Goto(from-trunk,${EXTEN},1)

This removes the + sign and allows just the caller ID
to pass.

And to the kind soul whoever said to place the
sip_nat.conf settings
nat=yes
externip=(sterilised but I know my IP dont worry)
localnet=192.168.0.63/24

That fixed the one way audio problem, so thank
you also, anyone else I forgot I thank you also
and sorry to disturb people here, I cannot thank
you guys enough, the nat=yes also fixed a line I had
trouble with didforsale n ow works also.

Many thanks everyone.
 

w1ve

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Awesome... You've not disturbed anyone! This is a great community, always willing to help. The lesson I would take from this is don't flame vendors unless you are sure there is a problem! Most of these PSTN gateway providers are smaller shops and depend on word of mouth to feed their families. :rolleyes: There are lots of places that vet out all the providers -- try reading the http://www.dslreports.com/forum/voip forum -- lots of fodder!

If you want to know how to make your PIAF sing, this is da place!

-Gerry
 

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