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Incoming calls not going through

PostPosted: 10/18/2019
by pablob
Hello,

After a power outage we had yesterday our incoming calls go straight to voice mail. I am registered through SIP Sorcery, and the SIP Settings screen shows that SIP Sorcery is connected.
The SIP Sorcery call log does not show any inbound call. The strangest thing is that I see the following in the FPL call log:
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<regular #>    <FPL #>    0:04    Oct 18, 2019 10:01:25 AM    Oct 18, 2019 10:01:29 AM    Normal call clearing       $ 0.00
<regular #>    <FPL #>    0:04    Oct 18, 2019 10:01:25 AM    Oct 18, 2019 10:01:29 AM    Unallocated (unassigned) number       $ 0.00

Why would it say it is an unallocated number?

Looking a bit further, I see that yesterday incoming calls looked like this:
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<regular #>    <FPL #>    0:07    Oct 17, 2019 8:05:47 PM    Oct 17, 2019 8:05:54 PM    Normal call clearing       $ 0.00
<regular #>    <FPL #>    0:07    Oct 17, 2019 8:05:47 PM    Oct 17, 2019 8:05:54 PM    Destination out of order       $ 0.00

even if they were after the power outage.

An incoming call used to look like this in the call log:
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<regular #>    <FPL #>    04:56    Oct 08, 2019 12:26:19 PM    Oct 08, 2019 12:31:15 PM    Normal call clearing       $ 0.00


Could it be that FPL decided that the number was not valid anymore and then put it into a list of unassigned numbers?
Any insights on this would be appreciated.

Thanks!

Pablo B.

Re: Incoming calls not going through

PostPosted: 10/18/2019
by bridonca
Power down the router and ATA, leave it off for a minute, power up, and see what happens.

Re: Incoming calls not going through

PostPosted: 10/18/2019
by pablob
Unless I go and power down the SIP Sorcery systems I don't see how that will help. :-)
I have de-registered and I will leave it de-registered for a while. Then I will re-register and see what happens. Meanwhile I'll power down things (just in case, with computers you never know), and re-register.

Re: Incoming calls not going through

PostPosted: 10/18/2019
by pablob
bridonca wrote:Power down the router and ATA, leave it off for a minute, power up, and see what happens.

pablob wrote:I have de-registered and I will leave it de-registered for a while. Then I will re-register and see what happens. Meanwhile I'll power down things (just in case, with computers you never know), and re-register.


I did all of that, and an incoming call still goes straight to voicemail with the same message in the log.

Could that be a problem in FPL's number database?

Re: Incoming calls not going through

PostPosted: 10/18/2019
by hanke
Did you check to make sure your ATA does not have the "Do Not Disturb" enabled. Think its *78 to turn on and *79 to deactivate. Each ATA has different * (Star) codes..but pretty much the same in industry.

Re: Incoming calls not going through

PostPosted: 10/21/2019
by Jake
I would say that because this happened after a power outage it more than likely has something to do with something at your house rather than at SS or FPL so checking things in your ATA is a good route to go.

After that what I would do is either program the ATA to connect to FPL directly or use a softphone like x-lite (https://www.counterpath.com/x-lite-download) with your FPL details plumbed into it.
Unregister at SS first then connect with something else to prove if your account works or not.

You could then further test things by connecting x-lite to SS instead of your ATA and see if things work.

Just keep swapping parts of the puzzle in and out to give you an idea of where the problem might be.

Re: Incoming calls not going through

PostPosted: 10/23/2019
by Liptonbrisk
pablob wrote:Hello,

After a power outage we had yesterday our incoming calls go straight to voice mail. I am registered through SIP Sorcery, and the SIP Settings screen shows that SIP Sorcery is connected.


In addition to what others have written (check light patterns on your ATA--and perhaps, try another power supply if it was wrecked during the power failure; it's possible you're dealing with a hardware failure), you should be checking connection status by logging in at https://www.freephoneline.ca/showSipSettings.
SIP Status needs to indicate "connected" and SIP User Agent needs to be something you recognize.
If you don't recognize the SIP User Agent, you've probably been hacked.

Note that only one registration per FPL account is allowed at any time. When there are multiple devices/softphones using the same account, only the most recent registration is valid. The previous device will lose registration.

I would try switching to to a different FPL server: voip.freephoneline.ca, voip2.freephoneline.ca, or voip4:freephoneline.ca:6060 (I would test with voip4.freephoneline.ca:6060 first).
If switching servers work, you (SIP Sorcery) may have been temporarily IP banned.
In that event, you need to ensure the registration period is 3600 seconds (1 hr) and that the failed registration retry timer (if that settings exists for SIP Sorcery) is 120 seconds. Too many registration attempts within a short period can result in a temporary IP ban.

If you do see that SIP status is connected at the above link and that the user agent is SIP Sorcery or something you recognize, then it stands to reason that the problem is between SIP Sorcery and your ATA.


Proper device reboot order is always modem--->router (wait for it to be fully up and running and broadcasting Wi-FI SSIDs before turning on your ATA)--->ATA.

Personally, I would ditch SIP Sorcery, which is an additional point of failure, and get an Obihai OBi2xx series ATA instead if I needed to register up to 4 different SIP accounts plus 8 additional SIP gateways all on the same phone line.

Re: Incoming calls not going through

PostPosted: 10/29/2019
by pablob
Liptonbrisk wrote:
If you do see that SIP status is connected at the above link and that the user agent is SIP Sorcery or something you recognize, then it stands to reason that the problem is between SIP Sorcery and your ATA.



From what I see from SIP Sorcery logs, the call does not even reach them (it never gets forwarded there). Last time this happened, the FPL staff had to do something on their side to fix it.
I haven't had time to do it, but the next step will be to connect directly and see if anything changes (the "number unallocated" message bothers me and makes me think nothing will change).

Liptonbrisk wrote:
Personally, I would ditch SIP Sorcery, which is an additional point of failure, and get an Obihai OBi2xx series ATA instead if I needed to register up to 4 different SIP accounts plus 8 additional SIP gateways all on the same phone line.


That's an interesting idea. I saw there are OBI3 and higher series. Would you still recommend the 2xx series instead of the seemingly higher ones?

Re: Incoming calls not going through

PostPosted: 10/29/2019
by Liptonbrisk
pablob wrote:
From what I see from SIP Sorcery logs


I'm less interested in SIP Sorcery logs at this point than I am in what the SIP status and SIP User Agent state after logging in at https://www.freephoneline.ca/showSipSettings.
SIP Status needs to indicate "connected" (otherwise incoming calls won't work), and SIP User Agent needs to be something you recognize.

And, again, I would try switching to a different proxy server (in SIP Sorcery) to rule out potential IP bans.


That's an interesting idea. I saw there are OBI3 and higher series. Would you still recommend the 2xx series instead of the seemingly higher ones?


The 300 series doesn't support Google Voice, officially (without hacked firmware, which I won't be helping people use), while the 200 series does. The 300 series is not an upgrade at all.
I will not be supporting Google Voice on these forums, but the difference is the 2xx series officially supports Google Voice.

Re: Incoming calls not going through

PostPosted: 10/29/2019
by pablob
Liptonbrisk wrote:
pablob wrote:
From what I see from SIP Sorcery logs


I'm less interested in SIP Sorcery logs at this point than I am in what the SIP status and SIP User Agent state after logging in at https://www.freephoneline.ca/showSipSettings.
SIP Status needs to indicate "connected" (otherwise incoming calls won't work), and SIP User Agent needs to be something you recognize.

And, again, I would try switching to a different proxy server (in SIP Sorcery) to rule out potential IP bans.

[quote]

SIP Sorcerry is connected, and I see in the SIP Settings screen:

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SIP Status:    connected
SIP User Agent:    www.sipsorcery.com


I have tried with all the proxies (voip[-.1,2,3,4].freephoneline.ca, both hostname and IP), and it did not change anything.

Re: Incoming calls not going through

PostPosted: 10/29/2019
by Liptonbrisk
pablob wrote:Could it be that FPL decided that the number was not valid anymore and then put it into a list of unassigned numbers?


While that's possible, that doesn't make sense to me if the incoming call is going directly to voicemail. Voicemail shouldn't work at all.

What Jake wrote makes sense to test if your FPL account works, at least.

If the only difference was a power failure, then other than a UDP timeout/corruption issue between the router and the ATA, the only other option that logically follows is hardware failure.

Re: Incoming calls not going through

PostPosted: 10/29/2019
by pablob
Problem solved!

After testing with a SIP softphone (Linphone, on Android), I saw that incoming calls were coming in.
I went back to SIP Sorcery and I realized that I was connecting to FPL using the IP of the proxy (not the domain), which worked well for outgoing calls (SIP Sorcery was having some flaky DNS issues). I went back and switched the domain to the FQDN (voip2.freephoneline.ca), and things now work. So it looks like whether you register to the FQDN or the IP makes a difference in terms of receiving calls.

What I like about SIP Sorcery is that I can register there and receive calls in both my home phone and cell phone, and I can use FPL from my cell phone without de-registering the phone at home. But I may switch to the OBI ATA, and skip SIP Sorcery, I just need to ponder that.

Thanks everyone for the help!!

Pablo.

Re: Incoming calls not going through

PostPosted: 10/29/2019
by Liptonbrisk
Liptonbrisk wrote:
I would try switching to to a different FPL server: voip.freephoneline.ca, voip2.freephoneline.ca, or voip4:freephoneline.ca:6060 (I would test with voip4.freephoneline.ca:6060 first).



pablob wrote:I have tried with all the proxies (voip[-.1,2,3,4].freephoneline.ca, both hostname and IP), and it did not change anything.


That seems to run counter to what you've posted below.

pablob wrote: I went back and switched the domain to the FQDN (voip2.freephoneline.ca), and things now work.


If the corresponding IP address changed for voip2.freephoneline.ca (or if you were temporarily IP banned), then that makes sense. Either way, I'm glad it's working for you.

What I like about SIP Sorcery is that I can register there and receive calls in both my home phone and cell phone, and I can use FPL from my cell phone without de-registering the phone at home.


FPL doesn't require registration for outgoing calls, but registration is a requirement for incoming calls. If you use a decent SIP app, such as Groundwire, that allows the user to not register the SIP account, you can use FPL to make outbound calls on your cellphone while the ATA is still registered. After Groundwire completes the outbound call, the ATA continues to receive incoming calls.

You can use https://www.freephoneline.ca/followMeSettings to ring multiple phones.
With Follow Me, the original caller ID is maintained when the call is received on your cellphone.


But I may switch to the OBI ATA, and skip SIP Sorcery


That's up to you, but I would since SIP Sorcery is an additional point of failure.

You can also forward incoming calls with an Obihai ATA, using sp1(yourcellnumbergoeshere) in X_InboundCallRoute. However, since the ATA is bridging the call (original call to your FPL # is answered by the ATA; then the ATA calls your cell number and finally bridges the two calls together), the incoming caller ID on your cellphone would show your FPL number.

The only way to avoid that with Obihai ATAs is to forward/bridge the call to a SIP URI. For example, sp1(*SIPURInumbergoeshere@sipbroker.com;ui=$1). ui=$1 retains original CID information, but that doesn't work when forwarding or bridging to regular phone numbers.