Outgoing calls work, Incoming calls ring but both parties cannot hear

[Cedarboy]

Tried searching on the forum for similar problem but could not find a solution.

My configuration is:
Cable modem -> Linksys WRT54G -> OBi110 -> Phone

SIP service is callwithus.com

I am able to make outgoing calls and works perfectly. When anyone calls me on the SIP number, the phone rings but when I pick up both parties cannot hear each other.

I have tried placing the OBi between the cable modem and WRT54G and it still has the same issue.

I have port forwarding set for 5060,5061 in WRT54G and 16600 to 16798 as well.

STUN is disabled on OBi.

What can be the issue here? Please help.

Thanks.

[Cedarboy]

An update to this:

I have tried the following settings and corresponding results.
1. With STUN Disabled:
    a) Directly connect OBI 110 to ADSL Modem: Outgoing works, Incoming - phone rings but no voice on either end
    b) Connect OBI 110 to Linksys WRT54G (which is connected to ADSL Modem):
Outgoing works, Incoming - phone rings but no voice on either end

2. With STUN Enabled:
    a) Directly connect OBI 110 to ADSL Modem: Outgoing works, Incoming - phone rings but no voice on either end
    b) Connect OBI 110 to Linksys WRT54G (which is connected to ADSL Modem):
Outgoing works, Incoming - phone rings but no voice on either end

My SIP service is callwithus.com.
They shared the following log. According to them SIP ALG could be the issue as the 3 ports highlighted in bold should be the same.

<<START LOG>>
Mar 11 12:51:41 west /usr/local/sbin/kamailio[744]: INFO: <script>: SIP message from udp:59.184.181.179:50683

REGISTER sip:sip.callwithus.com:5060 SIP/2.0

Call-ID: bf364af9@192.168.1.5

Content-Length: 0

CSeq: 34323 REGISTER

From: <sip:accountno@sip.callwithus.com>;tag=SP13d8c94261e5303e5

Max-Forwards: 69

To: <sip:accountno@sip.callwithus.com>

Via: SIP/2.0/UDP 59.184.181.179:50684;branch=z9hG4bK-fc7a511;rport

Authorization: DIGEST algorithm=MD5,nonce="UT4L8lE+CsaTYiYXTkPhYHbmcijiIkX1",realm="sip.callwithus.com",response="a67fab8769a

a09c836ee80be94c59395",uri="sip:sip.callwithus.com:5060",username="accountno"

User-Agent: OBIHAI/OBi110-1.3.0.2776

Contact: <sip:accountno@59.184.181.179:50691>;expires=60;+sip.instance="<urn:uuid:00000000-0000-0000-0000-9cadef00f002>"

Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER

Supported: replaces
<<END LOG>>

Strange thing is I had a Grandstream HT486 before and using the same configuration and it was working great when I disabled NAT Traversal and STUN on the HT486. I replaced the Grandstream HT486 with OBi110 and now it does not work. I remember having the same issue of not hearing incoming calls on HT486 and then disabling STUN and NAT traversal to make it work.
Is there a specific setting on OBi110 to disable NAT Traversal?

Experts, please help.

[Cedarboy]

It is really suprising and sad that no one has bothered to respond or suggest any remedies.

I see a lot of expert comments on other threads...is it a case of no-one having any ideas or just that the answer is hidden somewhere and nobody wants to help find it?

Expected much more from OBI110, the community and the support team.

[QBZappy]

Cedarboy,

Let's have a go at this if you like.

If you tried a STUN server without any success, then disable any STUN server settings on the OBi for this exercise.

Try the following to disable NAT. You may need to port forward the RTP ports.

ITSP Profile X
    General
    SIP
X_DiscoverPublicAddress   = Disable
X_PublicIPAddress = Your fixed ip or dyndns

Can you get the OBi echo test?

[Cedarboy]

Thanks a lot for responding QBZappy.

X_PublicIPAddress = your fixed ip or dyndns

Would the fixed ip in this case be the WAN IP Address for the ADSL Modem to which the OBI is connected?

Thanks

[QBZappy]

Yes.

[Cedarboy]

I did the settings as per your suggestion.

It still shows up as NAT under SIP Registration with Callwithus.

[QBZappy]

Did yo disable SIP ALG?
Have you tried putting the OBi in the DMZ? If this works it is most likely a firewall issue.

[Cedarboy]

The ADSL Modem does not have SIP ALG setting. It used to work fine with the previous ATA adaptor (Grandstream HT486).

I added the IP to the DMZ on the ADSL Modem in addition to the previous settings you suggested.

Rebooted both the ADSL Modem and OBI. Still shows as NAT.

Question:
1. Sometimes the OBI shows up twice under SIP registration at callwithus. After a few min, the second entry goes away. Is this normal?
2. The settings that you mentioned in the first email were only under ITSP -> SIP. There was nothing to be done under ITSP -> General. Is that correct?

[QBZappy]

The ADSL Modem does not have SIP ALG setting. It used to work fine with the previous ATA adaptor (Grandstream HT486).

Did you disable SIP ALG is on the router (not the modem).

The ADSL Modem does not have SIP ALG setting. It used to work fine with the previous ATA adaptor
Question:
1. Sometimes the OBI shows up twice under SIP registration at callwithus. After a few min, the second entry goes away. Is this normal?

My Grandstream PBX allows up to 3 different end points to register. Commercial sip providers may or may not allow more than one end point to register at the same time. Perhaps yours does. Test it with a softphone and the OBi at the same time if you want to find out.

The ADSL Modem does not have SIP ALG setting. It used to work fine with the previous ATA adaptor
Question:
2. The settings that you mentioned in the first email were only under ITSP -> SIP. There was nothing to be done under ITSP -> General. Is that correct?

Correct. That was just a roadmap to get you there.

[Cedarboy]

OBI is directly connected to the ADSL Modem which is connected to the internet. There is no router in between.

[QBZappy]

The ADSL Modem does not have SIP ALG setting. It used to work fine with the previous ATA adaptor (Grandstream HT486).

Just for the record the modem may indeed have an ALG setting. To make it more difficult to fix, I have read a few people needing to telnet into it because it is not displayed in the GUI. It may have worked with the HT486. I have an OBi110/202 and a Grandstream HT-502 ATA all happily working behind a a Tomato moded router.

Can you compare the setting on both atas and notice anything that is different.

BTW what model modem?

[QBZappy]

More desperate measures to try:

1) Try port forwarding these 50683/50684/50691 to the OBi (according to the log). Or the range 50683 to 50691.
2) ITSP Profile A->General->X_SymmetricRTPEnable = Put a check mark (Enable it)

[QBZappy]

Another thing to try:
Enable UPnP in the modem if it is supported.

[Cedarboy]

QBZappy:

"Can you compare the setting on both atas and notice anything that is different." - The Grandstream broke down recently and hence I replaced it with OBI, maybe should have stuck to the same model :-(.  So will not be able to check the settings. I have asked a friend of mine to share his settings that work as he is also using the Grandstream.

The Modem model is UTStar UT-304R2U. I also found that lot of folks telnet to disable NAT. I did telnet into the modem but could not find any setting or command that would disable SIP ALG.

On Port Forwarding, I have a few questions
1. Should the port forwarding from 50683 to 50691 be for UDP, TCP or both?
2. The Modem has two tabs - One for Virtual Servers and the other for Filters
    Virtual Server says "Virtual Server is used to allow Internet users access to LAN services. When the default value of Local Port = 0 , it means Any Port."
    Filters page says "This Page is used to View and Modify IP Filter Global and Rule Configuration." It has options for Inbound and Outbound filters.
Which one of these pages should be used for Port Forwarding? I guess the Virtual Server page but not 100% sure.

Should UPnP enabling be used ONLY if the Port Forwarding and SymmetricRTPEnable does not work or should it be enabled along with port forwarding and symmetricRTPEnable?

[Cedarboy]

I checked and UPnP was always enabled on the Modem all this while.

[Cedarboy]

QBZappy:

Any thoughts on my questions?

I went ahead and setup google voice as SP1 and moved callwithus to SP2.

Strangely enough,GV works perfectly both incoming and outgoing calls.

So the question is why would callwithus not work?

[Anders]

After more than a year of flawless operation and "no" configuration changes I started to experience exactly the same problem as Cedarboy about two weeks ago. GV working w/o problem, SIP "Localphone" Outgoing calls no problem, incoming calls ring, but not connected when phone taken off hook. However I have a CISCO (cable) modem, a Netgear WGR614 router and an Obi100.

What I found was that the router had lost part of its configuration and among other things SIP ALG was no longer disabled. After disabling SIP ALG in the router incoming calls now work as expected.

I have no idea how the router spontaneously could have lost its configuration, however about two years ago I experienced something similar when my cable operator forced an unannounced FW upgrade on the modem. When the modem was "upgrade" the mode of operation was changed from "gateway" to "router". This effectively caused me to have two routers in series before the Obi100, something the SIP implementation in Obi apparently can't handle (incoming calls ring but can't be answered, no audio on outgoing calls), however the GV protocol seem to be able to handle this. Since all other applications could handle this configuration it took me some time to understand what the problem was. The solution at that time was to reconfigure the modem back to "gateway" mode.

I suggest you review both your modem and router configurations for any unexpected configuration changes. They do happen... I would also try to use a PC SIP phone like Twinkle. It is my experience that Twinkle works with SIP ALG enabled in the router, while Obi100 doesn't.