Inbound calls drop after 30 seconds

[Daniel3]

Hello,
new installion of FusionPBX on a raspberry pi with 3 Polycom extensions and Twilio for sip provider. Raspberry pi is on internal network behind pfsense firewall.
Extension to extension and outbound calls work great. Incoming calls drop after 30 seconds. I have all Twilio addresses in NAT and the firewall does not log any blocking of Twilio addresses.
Tried changing ext-rtp-ip to various setting found here but nothing helped.
I am very new to this so any help is greatly appreciated, can upload any logs you need.

Thank you,

Daniel

 

[fusionpbxnoaudio]

SOLVED - 32 second call disconnection

I know there are many posts with this issue, I have tried all that is possible but it is still disconnecting. Let me explain my setup. I have a simple full cone NAT on a router which has an external IP and the internal LAN is 192.xx.xx.xx Fusion PBX resides on a computer inside the LAN which...

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Check this link - if it still doesn't work post your external profile settings page

 

[Daniel3]

I have tried various settings from the link you provided and reset the box after but the inbound calls still disconnect. I have attached the external sip profile. The IP address in the ext-rtp-ip and ext-sip-ip fields is the external IP address of the firewall. I also tried the pbx IP address (10.5.1.50) without success

Let me know if you need to see any other settings or files.

Thank you.

 

[DigitalDaz]

Post a sip trace of a call please

 

[fusionpbxnoaudio]

you need to set like this, don't use autonat in both places

ext-rtp-ip	autonat:externalip	True
ext-sip-ip	externalip	True

 

[Daniel3]

Still dropping the calls after changing ext-rtp-ip to autonat:externalip and ext-sip-ip to externalip.

Attached is the logs from a call.

Thank you

 

[fusionpbxnoaudio]

Still dropping the calls after changing ext-rtp-ip to autonat:externalip and ext-sip-ip to externalip.

Attached is the logs from a call.

Thank you

If the log file is correct, then, you have incorrectly set the inbound route, there is no "external" profile being used as per the log file. Your inbound call incorrectly hits the internal profile.

SOLVED - Inbound call hits incorrect dialplan

Please see attached log, I have set an inbound route using Dialplan->Destinations However when an incoming call is received from the gateway, it hits the Dialplans in the Dialplan Manager obviously my expected dialplan is not in that list as it is under Inbound routes. How do I make the inbound...

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[fusionpbxnoaudio]

Next time capture the logs after executing "sofia global siptrace on"

 

[Daniel3]

Hello,

it has been a while since trying to get this working. Looking at the logs, I see a strange IP address, 172.18.86.223:

Via: SIP/2.0/UDP 54.244.51.0:5060;branch=z9hG4bK4535.698f77dd8e05bf72c92efa85cd3d49db.0

Via: SIP/2.0/UDP 172.18.86.223:5060;rport=5060;branch=z9hG4bK113f5355-d510-43dd-8c6f-b511b42e9db0_6772d868_437-3897283728293853576


There is nothing on the network with 172.18.86.223 so I don’t know why that IP address shows there. In fact the last two octets change from call to call.

Twilio is seeing that 172.18.86.223 is requesting the BYE:

1536.373350149.28.158.9954.244.51.0SIP784 Request: BYE sip:Anonymous@172.18.86.223:5060;transport=udp |


Any suggestions?

Thank you,
Daniel

 

[fusionpbxnoaudio]

The call origin party in your case, I assume it is another SIP server, is not receiving the SIP OK message which you are sending after the call is established. The BYE message is from the originator. The internal ip which you are receiving is from the other SIP server which is connecting to yours.

Your freeswitch is still sending internal ip in the SDP header - this is a separate issue.

 

[Daniel3]

Thank you! That helped to resolve the problem. For others dealing with this issue, here is what worked:

External SIP Profile:
ext-rtp-ip: $${external_rtp_ip}
ext-sip-ip: $${external_rtp_ip}

Internal SIP Profile:
ext-rtp-ip: Public IP Address
ext-sip-ip: Public IP Address