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Call gets dropped in middle of conversation


ext101
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Can anyone please help?

 

Drop calls happen in the middle of conversation, sometimes as soon as 1 or 2 minutes in, and on some cases calls do last 8 to 10 minutes before it gets cut. We are just doing plain SIP to SIP calls, using a windows machine. Even with only 2 users on their calls this happens.

 

Thanks a lot!

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After doing test calls it looks like its not actually a dropped call, but instead the caller (pbxnsip) cant hear the called party --- like a one-way audio --- but the called party can still hear the caller.

 

Can anyone please help?

 

Drop calls happen in the middle of conversation, sometimes as soon as 1 or 2 minutes in, and on some cases calls do last 8 to 10 minutes before it gets cut. We are just doing plain SIP to SIP calls, using a windows machine. Even with only 2 users on their calls this happens.

 

Thanks a lot!

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Thanks for the link.

 

Would putting the pbx on the DMZ, or assigning it with an public IP, thus exposing it to the internet lessen this complication? Right now I only have port-forwarding rules on the firewall.

 

The best way to track this problem down is to use Wireshark and record a conversation. Typically, those kind of problems are because of NAT or firewalls. Also silence suppression can be a problem. A wireshark trace will show it.

 

More information can be found in http://wiki.pbxnsip.com/index.php/One-way_Audio.

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Would putting the pbx on the DMZ, or assigning it with an public IP, thus exposing it to the internet lessen this complication? Right now I only have port-forwarding rules on the firewall.

 

Check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses for thus ugly topic.

 

Can't wait until we have IPv6 rolled out, then there is no more need for this kind of address recycling!

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