No Audio Remote Peer Asterisk 11.1.2

Asterisk Asterisk Support
1

Hey Everyone,

I’ve been using Asterisk as part of the PBXIAF setup now for about 2 years and its been awesome. Earlier last year I upgraded my install from 1.8 to 10.5 and it went as smooth as can be. Yesterday I decided to upgrade from 10.5 to 11.1.2 mainly because it is now the next LTS version of Asterisk. Everything with my setup works as expected except for my softphone at work. At work I’m using Linphone on my Ubuntu machine and it connects to my asterisk server at home with no issues however whenever I attempt to make a call the call connects but no audio is being delivered back to my client. All of my phones in my house work just fine. To rule out if it was something with my setup at work I rolled back to a snapshot of 10.5 and everything worked just fine on the 10.5. When I did my upgrade to 11 I followed the same process as I did with the 10.5 and built locally from source so I cannot figure out what is going on here. Nothing is standing out to me in the logs or the debug.

My setup is a CentOS 6.2 VM 32Bit running on ESXi 4.1
I have firewall rules in place to allow my work computer to remotely connect to my Asterisk server at home
Asterisk 11.1.2
FreePBX 2.10
YATE for GVoice (although I’ve heard that in version 11 GVoice support has improved so might move back)

Let me know what else I can provide to you. Thank you.

2

So I just installed an IAX client and the audio works just fine with that so I think that I might use the IAX client for now but I would like to still troubleshoot what the issue is with the SIP client. Both my IP phones and my Linksys device at home are sip devices and they connect to the Asterisk box just fine so the issue lies only with the remote client. Thanks.

3

Chek your nat settings for SIP and forwarded ports for SIP & RTP

4

If you have properly forwarded the SIP ports and RTP ports for audio…you might want to look into your edge device. There are quite a few known devices that are not SIP friendly.

One of the reasons I have noticed people like IAX is it is generally easier to get to work in a ‘NAT transitioning’ type situation…

5

Can you please do “sip set debug on” in the Asterisk console and copy/paste the output here? Also please explain which device is on which IP …

6

Thank you all for your responses. I will get you the debug output later on this afternoon when I have a second to get it turned on. Thanks.