Hello,
I have installed and configured an Asterisk server but I can’t seem to get any audio when calling from my mobile.
I don’t have this problem when calling from a phone connected to the Asterisk server.
The phones are behind a firewall and we use a VPN to connect to the Asterisk server.
The Asterisk server is registered to a SIP trunk from a provider (OVH).
Here 
is a screenshot from Wireshark.
Where seems to be the problem?
91.121.129.23 - siptrunk.ovh.net
91.121.128.138 -don’t know exactly what this is. some IP from OVH’s network
10.107.0.51 Asterisk server
Thank you
Your firewall is probably turfing the incoming channel from your provider. So, your phone call is coming in from the PSTN, to your system, and gets to the firewall. The firewall says “I have no idea where to route this.” and it tanks.
Read up on qualify. This punches a persistent connection through to the PSTN gateway/VoIP provider and then the calls will know where to go.
Hopefully.
qualify dosen’t do anything with the RTP ports.
DERP. Yes, I totally mis-understood the question and after re-reading it, I see what the OP was asking for. Silly me. I
As far as the original question goes, are you able to hear audio at all on either the calling or called phone, if so which one? Furthermore can you please post a call log of a call in progress?
Hello and thank you for your time.
I have posted an answer but it doesn’t show up. Don’t know why.
Also, I have linked a photo in my first post but it vanished.
I can’t hear a thing it doesn’t matter who calls(mobile or VOIP Phone).
Here is a link with a screenshot from wireshark.
imageshare.web.id/images/x9ufkyd … igj1v9.png
Can you comment on it?
I was having the same problem a few days ago. Got it fixed with some help from this forum and my sip provider.
Heres what did it for me. In your sip.conf you have context with your sip provider info. There should be a line that starts with host; “host=sip.SomeProvider.com”, add “port=5160” just under it.
You should also have a register line with something like: register => username:password@sip.SomeProvider.com. Add port 5160 after so it looks like this: register => username:password@sip.SomeProvider.com:5160
Save and Reload Asterisk.
Worked for me.
All that does is explicitly set the default value. Unless you have set a non-default elsewhere, I don’t see how it helps. Also, it only affects the SIP protocol, not the RTP, so if the call is actually set up and doesn’t collapse after about 30 seconds, the bit that it affects is already working.
I was just saying I had the same problem that I couldn’t solve for a few days, I did what I posted above, and it was fixed. Probably won’t do anything but worth a try if nothing else is working.