No Incoming Audio for new "VOIP unlock key"

[sbordun3]

Hello,

My friend purchased a VOIP unlock key today for Cisco SPA112 and I am helping him to configure his device. During a call I get no incoming audio. I have configured his SIP device according to the recommended settings:
http://support.freephoneline.ca/entries ... redentials

CIsco SPA112 F/W 1.3.2(014)
Proxy: voip.freephoneline.ca
Preferred Codec: G711u/G729a
SIP Port: 5060 UDP
NO STUN

The problem exists only for a new unlock keys!
I Configured my friend's SPA112 with my own VOIP unlock key that I purchased couple of years ago and everything works fine. But when I changed the settings back to his unlocked VOIP key the problem with "no incoming audio" appeared again.

Any help would be greatly appreciated.

Thanks!

[Jake]

This might need to be looked at on their end. Open a support ticket and then PM Fongo Support the ticket number. See my sig for links.

[sbordun3]

HI Jake,

I created a ticket (Request #27332) and PM it.
I hope someone from fongo support will help me .

Thx

[bridonca]

What do you have for a router? Not only does SIP VOIP need an open SIP port, which is usually 5060, it also needs RTP port(s). Cisco ATA devices seem to need more than one, though, 4 should be enough. If you can get your router to direct these ports (which are the default RTP ports for Cisco ATA gear) to the Cisco ATA, 16384, 16385, 16386, and 16387, I believe you will fix your incoming audio issue.

Some routers are smart enough to route the correct ports, some need some help as I described above, and others are absolute junk when it comes to port forwarding and SIP VOIP.

[sbordun3]

router: CISCO DPC3825
I have 3 different SIP devices connected to this router:
1) Grandstream GXP2000 - third-party SIP phone
2) Linksys PAP2T - with 2 different FreePhoneLIne numbers (working fine)
3) SiSCO SPA112 - with my friend's newly unlocked FreePhoneline SIP key ( i hear for the first 1 second and then the incoming audio disappear)

No ports are forwarded on a router and no STUN configured.
I assume that PAP2T works fine with 2 Freephoneline numbers without any port forwarding means that no ports should be forwarded for the SPA112.
I have done 1 test:
1) I disabled one of the Freephoneline numbers on the PAP2T and configured it on SPA112. The audio problem disappeared
2) After that I configured the newly unlocked SIP Freephoneline number on the SPA112 and the "audio in" problem appeared again.
This test is proving that no port forwarding is required on my router because my old Freephoneline number works fine on SPA112 ! It is something wrong with the configuration on the server for the newly registered Freephoneline #.

DO you (technical support) have access to the server settings to compare my old Freephoneline account and the newly unlocked one? I belive you will fine the difference in configuration and will be able to fix my problem.

Thanks

[Fongo Support]

router: CISCO DPC3825
I have 3 different SIP devices connected to this router:
1) Grandstream GXP2000 - third-party SIP phone
2) Linksys PAP2T - with 2 different FreePhoneLIne numbers (working fine)
3) SiSCO SPA112 - with my friend's newly unlocked FreePhoneline SIP key ( i hear for the first 1 second and then the incoming audio disappear)

No ports are forwarded on a router and no STUN configured.
I assume that PAP2T works fine with 2 Freephoneline numbers without any port forwarding means that no ports should be forwarded for the SPA112.
I have done 1 test:
1) I disabled one of the Freephoneline numbers on the PAP2T and configured it on SPA112. The audio problem disappeared
2) After that I configured the newly unlocked SIP Freephoneline number on the SPA112 and the "audio in" problem appeared again.
This test is proving that no port forwarding is required on my router because my old Freephoneline number works fine on SPA112 ! It is something wrong with the configuration on the server for the newly registered Freephoneline #.

DO you (technical support) have access to the server settings to compare my old Freephoneline account and the newly unlocked one? I belive you will fine the difference in configuration and will be able to fix my problem.

Thanks

Hi there,

Can you please confirm that someone replied to you on the support ticket?

Please update this thread with the latest information.

Thanks,

[sbordun3]

HI,

I received the message from the Support: "I will get someone to look into this asap."

How long should I wait?
Thanks

[sbordun3]

HI,

The problem was resolved by Technical Support: "... device is stuck into a bad registration. I have forced a registration on the account, please power off device for 2 minutes and power back on. "

Thank You for a help

[dreadnok]

This might need to be looked at on their end. Open a support ticket and then PM Fongo Support the ticket number. See my sig for links.

I have the same issue as the original poster. I set up my inlaws Obi 110 and NetComm 3G10WVR. I set up the port forwarding on the router as required. I get no incoming audio with their Voip Unlock Key. I put in my own Voip Unlock Key and it works fine.
I put in a Ticket # 27983 and sent a pm to Fongo Support. I got an email back from Fongo Support right away saying they do not offer support for Voip unlock keys and the ticket will be closed. I used the links in your sig to open the ticket. Are you able to help me? Thanks.

[Jake]

This might need to be looked at on their end. Open a support ticket and then PM Fongo Support the ticket number. See my sig for links.

I have the same issue as the original poster. I set up my inlaws Obi 110 and NetComm 3G10WVR. I set up the port forwarding on the router as required. I get no incoming audio with their Voip Unlock Key. I put in my own Voip Unlock Key and it works fine.
I put in a Ticket # 27983 and sent a pm to Fongo Support. I got an email back from Fongo Support right away saying they do not offer support for Voip unlock keys and the ticket will be closed. I used the links in your sig to open the ticket. Are you able to help me? Thanks.

It could be that something is getting a bit mixed up when you are switching accounts. There should be no reason why the two accounts are different, but I have found their system can be a bit finicky at times.

After you change the details, try powering down the ATA for 5-10 minutes before powering it back up again and see if that helps.

[bridonca]

If you are using multiple VOIP devices and multiple accounts, You have to make sure each VOIP device has it's own RTP port(s) You should use other unused ports other than the default, when using multiple SIP devices to minimize conflict, and flakey results.

Though it is always a possible Fongo issue, I would be more apt to believe your router is completely confused now, and should be powered down and start over. Same with the modem and ATA devices for good measure. I strongly recommend you set each ATA to use different RTP ports from each other. You have to set that up on the router and each SIP device. I would not worry about the sip port being the same, because that is essentially an outgoing port.

[dreadnok]

This might need to be looked at on their end. Open a support ticket and then PM Fongo Support the ticket number. See my sig for links.

It could be that something is getting a bit mixed up when you are switching accounts. There should be no reason why the two accounts are different, but I have found their system can be a bit finicky at times.

After you change the details, try powering down the ATA for 5-10 minutes before powering it back up again and see if that helps.

I powered off the ATA and waited 10 minutes and still have the same issue. The only thing I am changing on the ATA is the username and password on the ATA setup. If I punch in my freephoneline number and password it works with no issues. If I go back into the setup and punch in his freephoneline number and password I can make and receive calls but I can't hear the person on the other end but they can hear me.

How do I get support to look at the account? When I put in a ticket from the link in your sig I got a reply that they don't provide customer support for freephoneline and closed the ticket. The original poster who had the same issue said Technical support fixxed his issue because account was stuck in Bad Registration.

[dreadnok]

If you are using multiple VOIP devices and multiple accounts, You have to make sure each VOIP device has it's own RTP port(s) You should use other unused ports other than the default, when using multiple SIP devices to minimize conflict, and flakey results.

Though it is always a possible Fongo issue, I would be more apt to believe your router is completely confused now, and should be powered down and start over. Same with the modem and ATA devices for good measure. I strongly recommend you set each ATA to use different RTP ports from each other. You have to set that up on the router and each SIP device. I would not worry about the sip port being the same, because that is essentially an outgoing port.

I am not trying to set up two freephoneline accounts on one ATA. When I set up SP1 with my inlaws account number and password I get the one way audio issue. I go back in and change the SP1 details to my account number and password it works without issue.

[Fongo Support]

If you are using multiple VOIP devices and multiple accounts, You have to make sure each VOIP device has it's own RTP port(s) You should use other unused ports other than the default, when using multiple SIP devices to minimize conflict, and flakey results.

Though it is always a possible Fongo issue, I would be more apt to believe your router is completely confused now, and should be powered down and start over. Same with the modem and ATA devices for good measure. I strongly recommend you set each ATA to use different RTP ports from each other. You have to set that up on the router and each SIP device. I would not worry about the sip port being the same, because that is essentially an outgoing port.

I am not trying to set up two freephoneline accounts on one ATA. When I set up SP1 with my inlaws account number and password I get the one way audio issue. I go back in and change the SP1 details to my account number and password it works without issue.

Hi there,

I apologize for the delay- please reboot OBI and try again now.

You might have to power off modem, wireless router and OBI for 20 minutes if there are any issues.

Please try and let me know.

Thanks,

[dreadnok]

Hi there,
I apologize for the delay- please reboot OBI and try again now.
You might have to power off modem, wireless router and OBI for 20 minutes if there are any issues.
Please try and let me know.
Thanks,

I rebooted the ATA and it is now working. Thank-you very much for your help.

[Rathwell]

I have the same issue. I also put in a request but got an email back saying "We do not provide support for freephoneline...." Request #28683 "New Unlock Key One Way audio"

I have tried powering everything down but that did not work.

Is there anyway you could help me.

Thanks

[Jake]

Rathwell, PM the ticket number to Fongo Support and they should look into it for you. (See Sig)

[red-i]

I'm having the same issue. if i dial out, i can't hear any incoming audio but they can hear me.

grandstream HT 702, firmware is 1.0.4.14

router is Cisco E3200, firmware 1.0.04

i've followed the configuration for the HT 701, however i have set random SIP to no and random RTP to no.

static IP on the grandstream and I have port forwarded UDP 5060 to 6060 to the ATA. I have also forwarded 5004 to it as well.

for NAT I have it set to no, but keep alive.

any help would be greatly appreciated

UPDATE: I got it working, but only by using STUN. is there a way i can get this working without a STUN server?

UPDATE 2: only outgoing is working. for incoming, the other user gets a busy signal after 1 ring, but my phone continues to ring

[bridonca]

I'm having the same issue. if i dial out, i can't hear any incoming audio but they can hear me.

grandstream HT 702, firmware is 1.0.4.14

router is Cisco E3200, firmware 1.0.04

i've followed the configuration for the HT 701, however i have set random SIP to no and random RTP to no.

static IP on the grandstream and I have port forwarded UDP 5060 to 6060 to the ATA. I have also forwarded 5004 to it as well.

for NAT I have it set to no, but keep alive.

any help would be greatly appreciated

UPDATE: I got it working, but only by using STUN. is there a way i can get this working without a STUN server?

UPDATE 2: only outgoing is working. for incoming, the other user gets a busy signal after 1 ring, but my phone continues to ring

It looks like you need to route a RTP port from the router to the ATA. Set up the ATA to choose an unused port, something like 12000 tends to never be used, and have the router forward that. Make sure you are forwarding a UDP port, After you save settings, you power down the modem, router and ATA, so the devices are certain to use the saved settings.

[red-i]

thanks for the reply. tried this morning:

changed NAT from stun to keep alive
changed RTP port to 12000, forwarded UDP 12000 port from router to ATA.
powered router off, ATA off, router on, ATA on.

still no luck, but i'll fiddle a bit more tonight and report back.

[red-i]

I got it working

I had the no key entry timeout defaulted to 2 seconds, I forgot to change it to 4.

so for me, this is what works

I followed the 701 documentation settings exactly, including keeping random RTP and SIP ports enabled.

I did NOT port forward anything from my router.

[905benchguy]

I too have been trying to assist my friend that I recommended buying only a VOIP key, as I had done ~ 1.5 years ago. I have had the exact same problem you expereinced. My key, entered into his new SPA112 works, while his new key does not. I hear no ringing making an outgoing call and I also cannot hear the person who picks the line, but they can hear me. I thought I read that there is NO support for those who just bought the VOIP key. Did Fongo fix the problem and if so was there a charge of any kind?

[red-i]

No fongo did not fix anything for me. I just followed the ht 701 documentation and realized I missed one setting.

[eeeee]

I just configured my new Cisco SPA112 ATA following the step-by-step guide provided in this forum and am able to make and receive calls, however, there's no inbound audio on my FPL home phone at all....

I power cycled both my router (Bell Connection Hub/Sagemcom F@st 2864) and ATA in sequence, waited for about 10 minutes before powering them on, yet it did not help.

Anyone with the same Internet connection (Bell DSL), the same router and ATA experienced this kind of problem? Do I need to forward ports on the router? Please let me know.

Thanks!

[Mango]

From within the SIP settings of your device, please try to set the following:

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

Let us know if this works.