SPA3102 through a Bell ZTE MF275R No Inbound Caller Audio

[dalexop]

Hi folks,

I have an existing, previously working SPA3102 that functioned well on the Rogers HSI network.
The Rogers Internet Model was set to act as a Hub and passed all traffic through to my Router, which is where I managed my own firewall rules.

Now I have moved to a location where I can only obtain Bell Mobility Internet Services.
The Device I have is the ZTE MF275R Internet Hub.
I have setup the Hub for Port Forwarding.
1) IP address, Port range = 16384-16482 Protocol = TCP + UDP
2) IP Address, Port Range = 5060-6080 Protocol = TCP + UDP

Outbound Calls function perfectly, we hear the caller, they hear us.
All ports increment during calls on the MTA ( Inbound + Outbound RTP & SIP packets )

Inbound Calls, we can speak to the called party, and they hear us, however we cannot hear them speak.
When this happens, the RTP Packets Sent counter increments, the RTP packets Received Counter stays the same.
SIP packets continue to increment.

I'm convinced this is a Bell Router Issue.
Has anyone come across this particular Vendor Router ?
Thanks

[dalexop]

Posting Update

More testing.

I also changed the SIP Proxy from voip.freephoneline.ca to voip4.freephoneline.ca:6060

I also opened up my Router Ports from 10,000 to 64,000 and same behavior.

No RTP packets received on inbound calls.

I've getting the feeling this is not a Bell Router issue ??.

[bridonca]

Yes and no. You biggest problem is your Cisco ATA. In my not so humble opinion, I consider it is a piece of garbage. When it works great. But if something is misconfigured, you have problems. Obihai products doe not have these problems, they just work. Better ATA all around too. So is this problem worth $50 of your time?

[dalexop]

Hi there Bridonca,
Yeah I hear you.

However I am going to give it a little longer.
You're quite right, the 3102 is a bear to get working, but once it works, it's rock solid.
I have my router & MTA protected by UPS and in the past, it was up 100 % for over two years, no reboots at all.

There is a great deal of documentation on this exact scenario, once I figure it out, I'll report back.
If not, Obhihai, here I come...
Thanks

[Liptonbrisk]

I am not familiar with the Bell router you're using.Typically it's best to stick whatever junk your ISP gives you in bridge mode if you are using your own router. Call/contact your ISP if need be. For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/

Here are a few points (most of this information comes from a RFD member):

1. Port forwarding is a security risk and should only be done when all else fails unless you want to allow crackers to hack your devices and use your services on your local area network. I would start off by disabling all port forwarding and DMZ.

You may want to google "SIP Scanners"

SIP scanners are programs written by crackers (script kiddies). They look for ways to break into your home network by scanning for open ports. Typically, they'll scan for 5060, 5061 and a few others (some scan for a lot more than that). If a port is open, they can access your ATA (and, potentially, other devices on your LAN). They will try to make free calls using your services. That's one reason why port forwarding is bad (port forwarding opens ports; it's a security issue).

If your phone is ringing constantly with caller ids that appear as 1001, 999, etc., then bots/crackers/scammers are looking (scanning ports) for ways to break into your services and devices. These are not actual phone calls. They're SIP scanners, scanning open ports on your network.



2. According to Mango,

To prevent one-way or no audio with the new FPL switches, set the following on the Voice >> SIP tab:

Handle VIA received: yes
Handle VIA rport: yes
Substitute VIA Addr: yes

3. Navigate to Line 1 (or whatever you're using for FPL)-->SIP settings, change SIP Port to a random number between 30000 and 60000

a) SIP Scanners typically don't scan in this range. b) You may be able to circumvent SIP ALG by using a high port.

Note that this is not a registration or outbound proxy port. It's the local SIP port.


4. If there are still problems, try disabling the SIP ALG feature or SPI (Stateful Packet Inspection) in whatever router or modem/router combo it is that you're using:
http://www.obihai.com/faq/sip-alg/calling-out.

To better understand why SIP ALG is horrible visit http://www.voip-info.org/wiki/view/Routers+SIP+ALG and read.

I'm of the opinion Apple routers don't offer this feature, but you might as well check. If you manage to disable SIP ALG in the router, then retest.

DLINK router users may need to log into the admin page of their router, click the "Advanced" tab and then "Firewall Settings",
navigate to "Application Level Gateway (ALG) Configuration", and uncheck SIP: http://www.support.dlink.com/emulators/ ... dv_dmz.htm

If you received a modem/router combo, from your ISP ask your ISP. It is typically better to stick the modem/router combo from your ISP in bridge mode and use an external router.

See here for an example on how to disable SIP ALG in a router: http://www.obihai.com/faq/sip-alg/disable-alg



5. Save settings.
Turn off both router and ATA. Turn on router. Wait for router to be fully up and transmitting data. Turn on ATA.
Then retest by calling your FPL phone number. If the problem is solved, don't continue.

6. Try Proxyserver voip4.freephoneline.ca:6060
visit http://forum.fongo.com/viewtopic.php?f=15&t=16350 (look at page 6 of the .pdf)

voip4.freephoneline.ca:6060 is used to try to deal with SIP ALG problems.


Retest. When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number. If the problem is solved, don't continue.

voip4.freephoneline.ca:6060 is a SIP server whose purpose is to help those with SIP ALG issues (can't disable it in the user's router, for example).



7. You're port forwarding a much larger range than you need to anyway. You shouldn't be port forwarding anything unless all else fails first.

There are the only ports used:

a. RTP (UDP) port range 16384-16482 from your router to your ATA. For reference, that range can be found under SIP-->RTP Parameters -->RTP Port Min and RTP Port Max.
This would be the audio stream

You shouldn't be port forwarding at all. But if you're going to port forward, try the RTP range, by itself, first.


b. Line (FPL)-->SIP Settings-->SIP Port:

Typically this is 5060, but it should really be set to a random high port between 30000 and 60000 to help avoid SIP Scanners/crackers and circumvent faulty SIP ALG/SPI features in routers.

c. The SIP server registration port and outbound proxy ports (If not specified, these are typically 5060). If you're using voip4.freephoneline.ca, the port is 6060.

That's it for ports. Forwarding anything else will do sweet nothing, except make it easier for people to hack you.

[Dumitru]

Hello
I have the same problem . ISp is Bell fiber and I am using same router Sipura SPA 3102. If you know how to fix it please let me know.
Thanks so much in advance !

[Liptonbrisk]

Hello
I have the same problem . ISp is Bell fiber and I am using same router Sipura SPA 3102. If you know how to fix it please let me know.
Thanks so much in advance !

These instructions for the PAP2T apply to your SPA3102: http://forums.redflagdeals.com/freephon ... #p26808549
Except for step 7 the link is http://forum.fongo.com/viewtopic.php?f=15&t=16350

[dalexop]

Hi folks, *** Problem Fixed ***

Very interesting final test.
I had to order a second Internet Access Service, so now I have two Mobility Data Services.
One from Bell Canada and the other from Rogers.

I simply plugged the SPA3102 into the Rogers HUB and the problem went away.
Moreover I did not need to configure my Rogers HUB. No Firewall rules at all. It simply works.

There is something either in the Bell Hub or Bell network stopping the RTP UPD Packets coming back.

Note:

They both employ the ZTE MF275R Rocket Hub.

Rogers Hub:
Software Version MF275RUIV1.0.7
Firmware Version MF275R1.0.7
Hardware Version dgpB

Bell Hub:
192.168.0.1
Software Version * MF275RUIV1.1.1
Firmware Version * MF275R1.1.1
Hardware Version * dgpB

[dalexop]

** Please Note +++ The Bell Router cannot be put into Bridging Mode +++ **
This seriously limits our approach.

I am not familiar with the Bell router you're using.Typically it's best to stick whatever junk your ISP gives you in bridge mode if you are using your own router. Call/contact your ISP if need be. For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/

Here are a few points (most of this information comes from a RFD member):

1. Port forwarding is a security risk and should only be done when all else fails unless you want to allow crackers to hack your devices and use your services on your local area network. I would start off by disabling all port forwarding and DMZ.

You may want to google "SIP Scanners"

SIP scanners are programs written by crackers (script kiddies). They look for ways to break into your home network by scanning for open ports. Typically, they'll scan for 5060, 5061 and a few others (some scan for a lot more than that). If a port is open, they can access your ATA (and, potentially, other devices on your LAN). They will try to make free calls using your services. That's one reason why port forwarding is bad (port forwarding opens ports; it's a security issue).

If your phone is ringing constantly with caller ids that appear as 1001, 999, etc., then bots/crackers/scammers are looking (scanning ports) for ways to break into your services and devices. These are not actual phone calls. They're SIP scanners, scanning open ports on your network.



2. According to Mango,

To prevent one-way or no audio with the new FPL switches, set the following on the Voice >> SIP tab:

Handle VIA received: yes
Handle VIA rport: yes
Substitute VIA Addr: yes

3. Navigate to Line 1 (or whatever you're using for FPL)-->SIP settings, change SIP Port to a random number between 30000 and 60000

a) SIP Scanners typically don't scan in this range. b) You may be able to circumvent SIP ALG by using a high port.

Note that this is not a registration or outbound proxy port. It's the local SIP port.


4. If there are still problems, try disabling the SIP ALG feature or SPI (Stateful Packet Inspection) in whatever router or modem/router combo it is that you're using:
http://www.obihai.com/faq/sip-alg/calling-out.

To better understand why SIP ALG is horrible visit http://www.voip-info.org/wiki/view/Routers+SIP+ALG and read.

I'm of the opinion Apple routers don't offer this feature, but you might as well check. If you manage to disable SIP ALG in the router, then retest.

DLINK router users may need to log into the admin page of their router, click the "Advanced" tab and then "Firewall Settings",
navigate to "Application Level Gateway (ALG) Configuration", and uncheck SIP: http://www.support.dlink.com/emulators/ ... dv_dmz.htm

If you received a modem/router combo, from your ISP ask your ISP. It is typically better to stick the modem/router combo from your ISP in bridge mode and use an external router.

See here for an example on how to disable SIP ALG in a router: http://www.obihai.com/faq/sip-alg/disable-alg



5. Save settings.
Turn off both router and ATA. Turn on router. Wait for router to be fully up and transmitting data. Turn on ATA.
Then retest by calling your FPL phone number. If the problem is solved, don't continue.

6. Try Proxyserver voip4.freephoneline.ca:6060
visit http://forum.fongo.com/viewtopic.php?f=15&t=16350 (look at page 6 of the .pdf)

voip4.freephoneline.ca:6060 is used to try to deal with SIP ALG problems.


Retest. When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number. If the problem is solved, don't continue.

voip4.freephoneline.ca:6060 is a SIP server whose purpose is to help those with SIP ALG issues (can't disable it in the user's router, for example).



7. You're port forwarding a much larger range than you need to anyway. You shouldn't be port forwarding anything unless all else fails first.

There are the only ports used:

a. RTP (UDP) port range 16384-16482 from your router to your ATA. For reference, that range can be found under SIP-->RTP Parameters -->RTP Port Min and RTP Port Max.
This would be the audio stream

You shouldn't be port forwarding at all. But if you're going to port forward, try the RTP range, by itself, first.


b. Line (FPL)-->SIP Settings-->SIP Port:

Typically this is 5060, but it should really be set to a random high port between 30000 and 60000 to help avoid SIP Scanners/crackers and circumvent faulty SIP ALG/SPI features in routers.

c. The SIP server registration port and outbound proxy ports (If not specified, these are typically 5060). If you're using voip4.freephoneline.ca, the port is 6060.

That's it for ports. Forwarding anything else will do sweet nothing, except make it easier for people to hack you.

[dalexop]

Hi folks,
the Problem is fixed by moving Telecom vendors.
See above for details.

I can replicate the fix by moving the SPA3102 from Bell Internet HUB to the Rogers Internet HUB.
Finally have a working Home Phone again.

[Liptonbrisk]

** Please Note +++ The Bell Router cannot be put into Bridging Mode +++ **

That's interesting. Are you positive? I ask because typically the first people you speak to when calling Bell have no clue what bridge mode is, much less, how to enable it.

This seriously limits our approach.

Well, you could have still tried steps 2, 3, and 6.

If I had to do step 7 (port forward) to get things working, I would have probably switched to something else too.

There is something either in the Bell Hub or Bell network stopping the RTP UPD Packets coming back.

If I had to guess, Bell may have SIP ALG or SPI enabled in firmware with no way to disable it. Given that you were port forwarding, it's unlikely RTP packets were even reaching the router.
http://www.voip-info.org/wiki/view/Routers+SIP+ALG (take a look at SIP ALG problems)

Anyway, that's good that the service is working for you!