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Calls drop after approximately 30 seconds

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Billy Issac

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I would appreciate it if anyone could point me in the right direction.

1 - When outgoing calls are made from our “in premise” Cisco 7960 phones - on every occasion the call drops after approximately 30 / 32 seconds. Incoming calls from all sources are fine.

2 - Whenever one of our Cisco 7960 phones ends a call and the other caller does not hang up, I can see from the Myphone interface that while the Cisco phone has dropped the call, the call persists in a connected state within the PBX, until the other caller terminates the call on their phone..

Background

We had been running a 3cx ver 11 PBX on Windows Server in our Portsmouth Office for a couple of years without issues. The system comprised of a Windows 2008 Server, 8 port PoE Switch, Cisco EA500 fiber router and 6 Cisco 7960 handsets. We have a small number of field workers who tunnel into the PBX using the 3CX Softphone. A number of our users both in the premises and remotely use the desktop 3cx Myphone in addition to their hard or soft phones.

As mentioned previously this system operated reliably for a couple of years

Recently we decided to host our PBX on an AWS EC2 M1 Small instance which has been launched into a VPC (Virtual Private Cloud), with both a static public ip and static private ip address.

In our new hosted setup the “in premise” Cisco phones are connected to the AWS hosted PBX through the same switch and Cisco EA4500 Router (as previously connected to the local PBX) using the 3CX Sip Proxy Manager. The phones pull their configuration files from a TFTP server sitting on the “in premise” Network and the individual phones successfully register as extensions within the AWS hosted 3CX Management Console.

Generally the new setup for the phone system works well apart from the issues described above.

Calls made to both the “in premise” Cisco phones and “off premise” softphones from external numbers work perfectly and as expected.

Calls made from softphone to softphone attached on a PBX extension, and to external numbers work perfectly and as expected.

Any advice would be appreciated.
 
I'd look into Duplexing issues along the network chain, i had this almost exact problem when a new firewall was put in with NICs that didn't duplex properly with the switch attached to it. Sometimes a call would drop as you picked up (rare) but most of the time it was exactly 25-30 seconds into a conversation, even if they were on hold.
 
Another very good question would be, what do the server logs say about the BYE from; on the 3cx? is it the PBX / phone hanging up or the destination/originating call address/phone number?
 
Were any changes made to the network. Hardware, software, firmware updates?
How long ago did this start? What changed about that time?

As mentioned, check the 3CX server log. 30ish seconds is about the time before 3CX will give up waiting for a SIP reply and put an ACK not received in the log. Which means that the device either did not receive the original message, it never never sent the ACK, it was sent to the wrong IP, or it was blocked by the network.

The use of Wireshark may help to figure out which of these it is, if all else fails.
 
Thanks Guys - I can see it is an "ACK not received " from the server log - going to dig a bit deeper.

The only change made to the system was that instead of the phones connecting to an "in premises" server we connected to a hosted AWS EC2 instance using the 3cx sip proxy manager. We set this up with the same version of 3cx and used the same configuration file we had been using and only changed the external and internal ip to reflect the ip values of the AWS server.

I'm not a real techie person but, because calls from 3cx softphones connected to the system via their own tunnel are not affected I reckon it must be something to do with the sip proxy manager which is used to connect the cisco phones to the hosted server.
 
I would start at the phone end, using Wireshark, to see exactly what the set is receiving, and, if it is sending the ACK message, and to where. If it is going where you think it should be, then you may have to look into the settings of the "destination".
 
Sounds to me Leejor is 100% correct, what I would try would be to go to VoIP providers tab--->advance---> I'd would then select specified IP as opposed to stun and put in your WAN IP. I woould then go to settings--->network settings----> turn off stun request and input your WAN IP. Make test call at that time... Also enable pbx delivers audio for the direct remote extensions.
 
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