Question

I have a freeswitch based PBX that has been working fine. I was using Skype connect as a SIP provider and I have had no difficulty making and receiving calls using this. Also, no difficulty with internal local-local calls.

I have just changed my sip trunk provider to voip-unlimited (based in the UK) and updated my sip profile accordingly. I can receive calls fine with the new provider, but when I make a call, the other party can hear me, but I cannot hear them. I do not get a ringing tone when I dial out (the remote party's phone rings, he answers the call, he hears me, but I cannot hear him).

I have ports 5060 and 5080 open to both UDP and TCP traffic and the router also supports PnP. I am uncertain if it is a firewall issue but certainly no problems were experienced with Skype connect previously.

Was it helpful?

Solution

the best thing would be to run a packet sniffer (tcpdump or wireshark) and see what's going on when the call is set up.

It might be:

  1. codec negotiation problem
  2. firewall settings problem
  3. NAT traversal problem

OTHER TIPS

Ok, got it sorted.

I set the PBX back to using Skype Connect. I ran wireshark and could see the connection getting established over TCP and the RTP packets flowing to and from the PBX using UDP.

I then switched over to the new SIP trunk provider. I again ran wireshark, could see the connection getting established over TCP, but this time incoming RTP packets were not present.

I checked the router's firewall and all seemed fine. Nothing in the log files etc. I still suspected the router however. Upon googling for my router model (a Netgear WNR2200) I came across a setting to disable the SIP ALG (Application Level Gateway). I did this (disabled) it and problem solved. By the looks of things, the SIP ALG feature of the router was interfering with and breaking SIP. It is supposed to solve some NAT problems, but in this case its use was undesirable.

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