ONE WAY VOICE... No one can hear me

[amado_1]

I am experiencing a ONE WAY VOICE and i would like to get some help from you…
I am using a Linksys SAP 1001 with freephoneline and my ISP is Teksavvy, my modem is a ZyXEL VSG1432 VDSL2 it runs off the Bell network.
Basically this is the problem. I have dial tone, I can make calls and receive them without problem but after one minute approximately I still can hear but the other end cannot hear me at all, then I can make or receive calls over and over again but the problem continue…
I had the opportunity to experiment with the settings of the ATA:

SIP Tab >> SIP T1: 1
Line Tab >> NAT Mapping: Enabled/Yes
Line Tab >> NAT Keep Alive: Enabled/Yes
Line Tab >> NAT Keep Alive Msg: KeepAlive
Line Tab >> Register Expires: 180 (I tried 300, 60 and nothing happened)
Line Tab >> SIP Port: 5060
Line Tab >> Preferred Codec: G729a (I tried G711a no dial tone and G711u same like G729a)

Line Tab >> Proxy: voip.freephoneline.ca
Line Tab >> Outbound Proxy: voip.freephoneline.ca

Primary DNS: 8.8.4.4
Secondary DNS: 208.67.222.222
DNS Server Order: Manual, DHCP
DNS Query Mode: Parallel

So far, the Line 1 is lit up and I have been making and receiving phone calls but the problem still happen…

[Fongo Support]

I am experiencing a ONE WAY VOICE and i would like to get some help from you…
I am using a Linksys SAP 1001 with freephoneline and my ISP is Teksavvy, my modem is a ZyXEL VSG1432 VDSL2 it runs off the Bell network.
Basically this is the problem. I have dial tone, I can make calls and receive them without problem but after one minute approximately I still can hear but the other end cannot hear me at all, then I can make or receive calls over and over again but the problem continue…
I had the opportunity to experiment with the settings of the ATA:

SIP Tab >> SIP T1: 1
Line Tab >> NAT Mapping: Enabled/Yes
Line Tab >> NAT Keep Alive: Enabled/Yes
Line Tab >> NAT Keep Alive Msg: KeepAlive
Line Tab >> Register Expires: 180 (I tried 300, 60 and nothing happened)
Line Tab >> SIP Port: 5060
Line Tab >> Preferred Codec: G729a (I tried G711a no dial tone and G711u same like G729a)

Line Tab >> Proxy: voip.freephoneline.ca
Line Tab >> Outbound Proxy: voip.freephoneline.ca

Primary DNS: 8.8.4.4
Secondary DNS: 208.67.222.222
DNS Server Order: Manual, DHCP
DNS Query Mode: Parallel

So far, the Line 1 is lit up and I have been making and receiving phone calls but the problem still happen…

Hi there,

Please verify your settings against the below:

http://forum.fongo.com/viewtopic.php?f=15&t=310

Also, please ensure that you have the correct ports fwd'ed in your router.

For RTP- in your SPA says UDP- 16384-16482

Also, if you have static IP setup on the SPA, please try to change back to DHCP unless you are very confident with networking.

Please try the suggestions above and let me know.

Thanks,

[amado_1]

Hi there,
Firstable i want to thank you for you help...
I already verified the setting and:
1. For tha ATA i switch everything exactly like is in the document, except Dial Plan category because the line that you wrote does not match with the picture below, so i pasted this :(911|[2-9]xxxxxxxxx|1xxxxxxxxxx|011xxxxxxxxxxxx.|98*|[6-7]x*xxxxxxxxxxx.)
2. RTP values in the ATA are OK...
3. I am very confident with networking, however i tried DHCP wich is the current configuration.
After that i Reboot the ATA.

About the Router: The theory says that Port Forwarding creates a bridge in order to let a party from the outside contact a specific user inside the network, The problem is from inside to outside, in other words i always can hear the other person perfectly so i am not expert but i think that the router lets come in the voice without restrictions but after 1 minute or so the other person cannot hear me any more...

So the problem still happen...

[Fongo Support]

Hi there,
Firstable i want to thank you for you help...
I already verified the setting and:
1. For tha ATA i switch everything exactly like is in the document, except Dial Plan category because the line that you wrote does not match with the picture below, so i pasted this :(911|[2-9]xxxxxxxxx|1xxxxxxxxxx|011xxxxxxxxxxxx.|98*|[6-7]x*xxxxxxxxxxx.)
2. RTP values in the ATA are OK...
3. I am very confident with networking, however i tried DHCP wich is the current configuration.
After that i Reboot the ATA.

About the Router: The theory says that Port Forwarding creates a bridge in order to let a party from the outside contact a specific user inside the network, The problem is from inside to outside, in other words i always can hear the other person perfectly so i am not expert but i think that the router lets come in the voice without restrictions but after 1 minute or so the other person cannot hear me any more...

So the problem still happen...

Hi there,

What are you describing here is caused by a NAT issue (symmetric NAT). Can you check your ATA and ensure that there are no STUN servers defined.

Thanks,

[amado_1]

I checked it and there is nothing in there...

FPL_NAT.jpg

[Fongo Support]

I checked it and there is nothing in there...

FPL_NAT.jpg

Hi there,

Do you have the ports 16384-16482 open in your router for the IP of your ATA?
Also, if your router has a sip alg function- please disable it.

Thanks,

[amado_1]

Hi there,

Firstable I want to really thank you, I appreciate all your effort for help me...

Ok to recap... I verified and switched the setting like you told me, i used to have static IP setup on the ATA, now i switched to DHCP. The SIP-ALG function was already disable but the problem still happened, As you suggetedto me i checked the ports in my router wich are not open, so i added the new rule in order to open them for the IP of the ATA and... BINGO..!!! I made a call and it kept on for more than 20 min, i am going to keep watching a mean doing tests and i will keep you informed...

Again Thank you very much...

[amado_1]

The day after the set up the phone did not have dial tone so i reboot the ATA and one week later so far so good... No problem AT ALL, Everything is working properly...