Ryan Posted January 17, 2008 Report Share Posted January 17, 2008 Hello, After setting up a grandstream phone, It seems that some routers work and others don't. If I open port 5060 on the router, I can hear both ways, yet if not specifically opened, the phone will only ring, and when picked up, you cannot hear either way. What could possibly cause this? Thanks Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 17, 2008 Report Share Posted January 17, 2008 Do NOT use STUN. Do NOT use STUN. Do NOT use STUN. Do NOT use STUN. Do NOT use STUN. Do NOT use STUN. Quote Link to comment Share on other sites More sharing options...
Ryan Posted January 21, 2008 Author Report Share Posted January 21, 2008 I had heard not to use STUN before, and we haven't been. It's been off. ALSO:: I tried testing a SNOM 105 phone and had the same problem. So can anyone else tell me why the phone will ring, but I can't hear on either end? Besides a firewall issue? Thanks. Do NOT use STUN.Do NOT use STUN. Do NOT use STUN. Do NOT use STUN. Do NOT use STUN. Do NOT use STUN. Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted January 21, 2008 Report Share Posted January 21, 2008 So can anyone else tell me why the phone will ring, but I can't hear on either end? Besides a firewall issue?Thanks. Do you want to hear that it isn't the firewall, when all evidence says it is? We specify Xyzel routers and we have yet to have a 1-way audio problem. Zywall Plus+2 makes a fine choice. RINGING occurs as a result of a TCP message using SIP Talking occurs on a UDP Stream once handed off between the two endpoints talking SIP to each other. The firewall of choice is clobbering the UDP streams but passing the SIP messages as you already know since you have 5060 forwarded. PLease post you firewall choices so others will know...(perhaps a firmware upgrade from vendor will help?) Quote Link to comment Share on other sites More sharing options...
Ryan Posted January 21, 2008 Author Report Share Posted January 21, 2008 I'm just looking for all possible options. We have an Asterisk server we set up a while back and with a phone on the same router / setup, it works fine back and forth. And we all know Asterisk uses port 5060 by default, just like pbxnsip. Which is why I am asking about other reasons beside the firewall. I don't want my clients when we go live to have to figure out how to port forward on their routers unless absolutely necessary. Do you want to hear that it isn't the firewall, when all evidence says it is? We specify Xyzel routers and we have yet to have a 1-way audio problem. Zywall Plus+2 makes a fine choice. RINGING occurs as a result of a TCP message using SIP Talking occurs on a UDP Stream once handed off between the two endpoints talking SIP to each other. The firewall of choice is clobbering the UDP streams but passing the SIP messages as you already know since you have 5060 forwarded. PLease post you firewall choices so others will know...(perhaps a firmware upgrade from vendor will help?) Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted January 21, 2008 Report Share Posted January 21, 2008 I'm just looking for all possible options. We have an Asterisk server we set up a while back and with a phone on the same router / setup, it works fine back and forth. And we all know Asterisk uses port 5060 by default, just like pbxnsip. Which is why I am asking about other reasons beside the firewall. I don't want my clients when we go live to have to figure out how to port forward on their routers unless absolutely necessary. We too have supported Asterisk in the past. However we seldom are forced to use a NATTED solutions but when forced we have used Zyxel. In all installations where we control everything we specify a Public IP dedicated to PBXnSIP for both Linux and Windows Installations. This avoids in most cases where doublenatting with home phones for the boss on some cheap cablemodem router barfing at us. I'd highly advise a Public IP in all cases. I''ve been advised by a good friend that Asterisk (next major release) will be more sip compliant and they too will begin to experience problems common to SIP compliant solutions. Time will tell though. Quote Link to comment Share on other sites More sharing options...
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