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
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