Grandstream UCM PBX settings

[charettepa]

I am having an issue finding the correct settings for
"Registration Expiry" and "Registration Retry Interval"

While i previously used Asterisk/FreePBX and prior to that ATA's and never had a problem
I have a Grandstream UCM PBX and the terminology is not lining up

I believe the i may have found "Registration Expiry"
but I am not sure
there is a "Max Registration/Subscription Time" and a "Default Registration Incoming/Outgoing Time"
which one should i set to 3600, if not both

for "Registration Retry Interval"
the closest i can find is "Register Timeout" which is right next to "Register Attempts" under outgoing SIP registrations
however the max i can set the value is 100

have i found the right field and if so, will this be a problem

[Liptonbrisk]

While I can't test, I would try this:

i) Default Registration Incoming/Outgoing Time 3600 seconds. The Expires value is what matters for FPL. Under 60 minutes won't work.

ii) Max Registration/Subscription Time 3600 seconds (The expiry value, I think, takes precedence).

iii) Register Timeout (next to Register Attempts) 100 seconds
While Freephoneline recommends 120s for the failed registration retry interval, 100 seconds should not produce a temporary IP ban. I would probably set Register Attempts to 3 or 4.

iv) See if you can find PBX Settings --> SIP Settings --> OPTIONS Keep Alive Interval. If you can find it (or something similar), set that to 20 seconds.
The setting might be under PBX Settings -->SIP Settings -->Basic Settings.

However, I don't have what you're using and am just guessing, unfortunately.

I'm also uncertain if I should be trying to help: "PBX connections or commercial uses are prohibited."
https://support.freephoneline.ca/hc/art ... redentials

Anyway, if you're just using FPL for residential usage (or as a hobbyist) and not for telemarketing, there shouldn't be a problem.

[charettepa]

thanks so much for your reply
ill try when i get home

on the UCM
by default the MAX is 28,800 (8 hours) by default
but the default is 1800 by default
so i could change this to 3600

for the retry i will set it to 100 (the max they allow)
im a little concerned though as i dont want to get blocked
whats weirs is that google searches show the 120 is because
FPL allows max 10 attempts per 5 minutes
but that would be 60s retry not 120s
it would only be 120s if it was 5 attempts per 10 minutes
if I knew the actual max attempts I could set a higher max retries
it really would be best to leave the default of unlimited though

I will certainly look for keep alive as you suggested
the problem is the terminology that Grandstream is using


the Grandstream UCM's are all PBX'x that are built on Asterisk 16
its beyond me why they are using non-standard language
and imposing restrictions that are not there in Asterisk

[Liptonbrisk]

for the retry i will set it to 100 (the max they allow)
im a little concerned though as i dont want to get blocked

I've stated in other threads what the rate limit was in the past.

"As May 2013, our servers will rate limit REGISTER requests to a maximum of 10 requests per 5 minutes. Each authentication round usually consumes 2 requests (digest auth), so it is a fair number given our guidelines."
https://community.freepbx.org/t/trunk-s ... ca/22479/8

but that would be 60s retry not 120s

I believe Fongo (Fongo operates Freephoneline) was just being safe for the value they published. People reboot their SIP clients too, and each reboot makes a registration attempt.

Over a decade ago, one of my accounts was temporarily IP banned for making too many registration attempts within a short interval.
I had to stop trying to register for over an hour to clear the temporary ban.
The experience was annoying. I don't know what the rate limit is currently, and I don't want to test to find out.


I'm pretty sure you won't be temporarily IP banned for trying to make one registration attempt every 100 seconds up to 3 or 4 times. You probably wouldn't be temporarily IP banned if you set max attempts to unlimited with a 100 second interval, but I'm not positive either because I don't what the current rate limit is (it might be the same as it was in 2013 though, in which case you should be fine).

[Liptonbrisk]

iv) See if you can find PBX Settings --> SIP Settings --> OPTIONS Keep Alive Interval. If you can find it (or something similar), set that to 20 seconds.
The setting might be under PBX Settings -->SIP Settings -->Basic Settings.

I keep looking at different sources and finding different info, so I'm not sure about the setting locations. Probably need your specific device model

Maybe try this:

1. Set Keep-Alive for PBX

The path might be PBX Settings-->SIP Settings-->NAT.

i) Set NAT Traversal to Keep-Alive

ii) Click "Save"/Apply

2. Navigate to Extension/Trunk --> VoIP Trunks

i) Edit your Freephoneline trunk

ii) Navigate to the Advanced Settings, and find the Keep-alive Interval or OPTIONS Keep Alive Interval setting
(maybe the setting is called "Frequency" or "Keep-alive Frequency", depending on the firmware version and device model)

iii) For value, enter 20 (seconds).

iv) Keep-alive Message: ensure it is set to OPTIONS.

v) Save/Apply settings.


Again, I'm not sure, but maybe this info gives you a hint for where to look.

[charettepa]

iv) See if you can find PBX Settings --> SIP Settings --> OPTIONS Keep Alive Interval. If you can find it (or something similar), set that to 20 seconds.
The setting might be under PBX Settings -->SIP Settings -->Basic Settings.

I keep looking at different sources and finding different info, so I'm not sure about the setting locations. Probably need your specific device model

Thank you so much for your quick replies and details
I have a UCM 6302 (latest series, mine has 2fxp and 2fxs)

[Liptonbrisk]

While I'm not 100% sure, based on what I'm reading, this is what I would try:

1. Navigate to Extension/Trunk--> VoIP Trunks--->click Add

a) Type: Register SIP Trunk
b) Provider Name: Freephoneline
c) Host Name: voip4.freephoneline.ca:6060

Use voip4.freephoneline.ca:6060 to help circumvent potential SIP ALG issues and anything (ISP) that tries to block UDP 5060.

d) Username: (Your 11-digit FPL number)
e) Authenticate ID: (Your 11-digit FPL number)

Username is used to build the "From" header (who you are claiming to be).
Authenticate ID is used for the actual challenge-response (proving it’s you).

For FPL, these are identical. However, on the Grandstream UCM, if you leave Authenticate ID blank, I think it defaults to using the Username field for authentication anyway.

f) Password: SIP Password is shown after logging in at https://www.freephoneline.ca/showSipSettings

g) Transport: UDP

h) NAT Traversal: Keep-Alive



2. Navigate to the Advanced Settings tab within the Edit Trunk menu.

a) set Local Port to random UDP port in the range from 30000 to 60000: 42718, for example. Ensure only this trunk uses this specific UDP port for SIP signalling. Don't use UDP port 42718 for any other trunk. Setting a high, random UDP port helps to circumvent SIP ALG and potential SIP Scanners/crackers.


b) Default Registration Incoming/Outgoing Time: 3600 seconds

c) Keep-Alive Interval: 20 seconds

d) Keep-Alive Message: OPTIONS

By the way, do not enable Support Subscriber. Freephoneline uses unsolicited Message Waiting Indicator. This means their server automatically pushes a NOTIFY message to your PBX whenever there is a voicemail, without your PBX needing to ask for it first. Never subscribe.

e) Click Save at the bottom.



3. The UCM has global settings that will override your trunk settings if they aren't aligned, I think.

a) Navigate to PBX Settings-->SIP Settings-->select General Tab

b) Max Registration/Subscription Time (s): set to 3600 seconds.

If this remains at the default (1800), the UCM will ignore your request for 3600 in the trunk settings. Setting this to 3600 allows the trunk to function as required by FPL.



4. In the Outbound SIP Registrations Tab,

a) Register Timeout (s): set to 100.
b) Register Attempts: set to 0 (unlimited retries). I think you'll be safe at unlimited.

e) Click Save and click Apply Changes button at the top.


Bleh. I should have told you to do step 5 earlier in this list. Oh well.


5. Navigate back to Extension/Trunk--> VoIP Trunks--> Edit FPL Trunk--> Media Settings

a) set preferred Vocoder: to 1. PCMU (G.711u)

b) While Freephoneline supports G.729, in my opinion, it sounds awful. I hate using it. Personally, I refuse to using anything other that G.711u with FPL. Up to you. Freephoneline doesn't support any other codecs.

c) DTMF Mode: People usually recommend using RFC2833 or Auto, but . . .

RFC2833 probably won't work with some banking systems (TD, for example). Inband will. Visit viewtopic.php?p=82960#p82960

Inband won't work with G.729 properly because it's a lossy codec that can wreck audio signals.

If you change the trunk to Inband but your IP phones are still sending RFC2833, I think the UCM will try to convert them.
To fix that you would navigate to Extension/Trunk --> Extensions. Then you would need to edit your primary extension(s), and navigate to the Media or Feature tab. Ensure the DTMF Mode there is also set to Inband.



Click **Save** and **Apply Changes**.



6. Navigate to **System Status** > **Trunk Status**.
7. Confirm the Freephoneline trunk is **Registered**.
8. If you have a firewall, ensure it's not blocking traffic on UDP ports 42718 (local port) and whatever your RTP port range is set to. You may also need to do the same for UDP port 6060.



Anyway, I would try something like that. Hope that helps.

[charettepa]

I don't have much in PBX > SIP > NAT

[charettepa]

Step 1 A-G is already done
but h) NAT Traversal: Keep-Alive
is not present only option is check box for NAT (on/off)

i will try all other settings you list in step 2 and above

Thank you so much for your help

[charettepa]

PBX > SIP > ToS

has a field "RTP Keep-Alive" Timeout
just below the Default Registration time

this must be the keep-alive interval you have in step 2
but i assume its not the same as the one you previously mentioned in SIP > NAT

[charettepa]

I can share any other screen you want
I have not yet Enabled the Trunk, i'm still worried about the 100s max for re-try

[Liptonbrisk]

Step 1 A-G is already done
but h) NAT Traversal: Keep-Alive
is not present only option is check box for NAT (on/off)

i will try all other settings you list in step 2 and above

Thank you so much for your help

Okay, since you only see a NAT checkbox in the trunk configuration, try this:


1) In your Extension/Trunk ---> VoIP Trunks ---> Edit Freephoneline Trunk page-->NAT. Check that box.

I think in newer UCM firmware, checking that box enables NAT handling for that specific trunk. The actual Keep-Alive behaviour is probably defined under Advanced/Global settings.

2. Since you are missing the dropdown for "Keep-Alive" in the trunk basic tab, navigate to the Advanced Settings tab of that same Freephoneline Trunk.

Then,

a) set Keep-Alive Interval to 20 seconds.

b) set Keep-Alive Message to OPTIONS (or NOTIFY since FPL accepts both, but OPTIONS is the standard for UCM).

[Liptonbrisk]

PBX > SIP > ToS

has a field "RTP Keep-Alive" Timeout
just below the Default Registration time

this must be the keep-alive interval you have in step 2

It's not. Those are two completely different mechanisms. One is sending keep-alives for the connection/registration (SIP), and the other is for the conversation (RTP) or audio stream.

1. SIP Keep-Alive (The one in Step 2)
Location: Extension/Trunk > VoIP Trunks > Advanced

This keeps the NAT association open (alive) on your router so Freephoneline (FPL) can reach your PBX. It sends a tiny SIP packet (such as an OPTIONS ping) to FPL every 20 seconds. So, this prevents your trunk from becoming Unregistered or losing incoming calls.

2. RTP Keep-Alive (The one in your image showing PBX --> SIP --> TypeofService)

This is for active calls. If you are on a call and there is a period of silence, some aggressive firewalls might think the connection is dead and cut the audio. This setting sends a dummy audio packet to keep the media stream alive.

RTP is for the audio stream.


So, SIP Keep-Alive keeps the phone line connected, and RTP Keep-Alive keeps the audio from dropping during a call.


For Freephoneline, you can generally leave RTP Keep-Alive (s) at the default of 0 (disabled), but I use 20 seconds anyway. Up to you.

SIP Keep-Alive set to 20 seconds is far more important because of Freephoneline's 3600 second registration interval. Without keep-alive packets, NAT firewalls can close the connection (NAT association); 1 hour is a long time.

[Liptonbrisk]

I can share any other screen you want
I have not yet Enabled the Trunk, i'm still worried about the 100s max for re-try

I'm pretty sure you're fine. If you're scared, set the retries to 3 for now (instead of "0" or unlimited) until you get things working.

[charettepa]

Thanks for all this
I just had to head out
I'll be back home in about 1 hour

I'm pretty sure advanced under trunk did not have much either

I'll send a full screenshot

[Liptonbrisk]

I'm looking at a UCM6xxxx SIP trunk guide and a manual. They mention "heartbeat". I'm not sure if that's the same thing as sending keep-alives to a SIP server because the manual keeps mentioning peers and devices instead.

"Enable Heartbeat
Detection
If enabled, the UCM630xA will regularly send SIP OPTIONS to the device to check
if the device is still online. The default setting is "No".

Heartbeat Frequency
When "Enable Heartbeat Detection" option is set to "Yes", configure the interval
(in seconds) of the SIP OPTIONS message sent to the device to check if the device
is still online. The default setting is 60 seconds."

https://www.grandstream.com/hubfs/Produ ... manual.pdf
(page 245)

That description seems to describe checking if other devices on the PBX are present.

I've not used your PBX, so I'm a little hesitant to suggest this (found under Advanced Settings when editing the SIP Trunk):

a) Enable Heartbeat Detection: Yes
b) Heartbeat Frequency: 20 seconds

I think that's right though based on some other configuration guides I'm reading. I figure OPTIONS can be sent to the configured Host Name (voip4.freephoneline.ca:6060, in this example)—not just peers.

(I fully admit you may need someone more familiar with your device than me; I was just trying to help).

[Liptonbrisk]

I've not used your PBX, so I'm a little hesitant to suggest this (found under Advanced Settings when editing the SIP Trunk):

a) Enable Heartbeat Detection: Yes
b) Heartbeat Frequency: 20 seconds

I think that's right though based on some other configuration guides I'm reading. I figure OPTIONS can be sent to the configured Host Name (voip4.freephoneline.ca:6060, in this example)—not just peers.

Wow, I think that's the same as triggering the qualify=yes behaviour in the underlying Asterisk engine.
And then qualifyfreq=20s

I agree with you: these UCM terms aren't intuitive.

Anyway, that's probably what you need to do.

[charettepa]

Running s little late
Will be home in 30-60
Will try all that you mentioned and advise on findings

[Liptonbrisk]

I'm probably done for this evening. I'll check back tomorrow.

[charettepa]

I have now applied all your recommended settings
except the server
I am not pointed to voip 4 with port 6060

i will change it if an issue occurs

i will activate after supper
I'll let you know how it goes

Thank you for all your help