Troubleshooting no sound call to specific numbers

[kariste]

Hello and happy new year

I've an fusionpbx 4.2 running on debian 8 server with a sip trunk for the outgoing calls. Being using it for a while, I didn't have any issues till recently that I noticed that calling a specific landline number, I get not sound, no dialtone but the call on the CDRs looks established. Can you point me to the right directions in order to troubleshoot this issue?

thank you

 

[Adrian Fretwell]

@kariste Happy new year to you too.
These type of problem can be difficult to find. It may be that the problem is not within your platform. Here in the UK, for example, we have experienced an issue with calls to one specific Other Licenced Operator (OLO) but not any of the others that we tested. We have also seen issues where the "landline" in question is routed through another SIP provider. It is not possible to guess what is happening in your case, but you asked about troubleshooting the issue.

The first thing I would do is set up a packet capture (Wireshark / tcpdump etc.) on the network between your telephone device and your server running fusion and (if they are different) on the network between your server and your SIP trunk. Examine the the packets that have SDP bodies, commonly INVITE and 200 OK, but there are situations where you will have an SDP in ACK. Look at the IP addresses and ports that the RTP media stream will use.

o=- 20009 20009 IN IP4 97.21.16.108.
s=SDP data.
c=IN IP4 97.21.16.108.
t=0 0.
m=audio 47638 RTP/AVP 18 8 101.

In the above example the target IP address for the media stream is 97.21.16.108 and the port is 47638. You will determine the endpoints for the media stream from INVITE and 200OK or 200OK and ACK.

Now you know the endpoints specified in the SIP signalling, check the packet capture to see if the RTP packets are actually being sent to/from the correct IPs and ports.

This is a good start for your troubleshooting, it may not get you all the way but I hope it helps. A.