I’m a newby with ip phones.
I’ve setup my freepbx 13.0.156 connected with iax trunks
All phones sip hardware and softphones.
Problem is with the sound quality being choppy.
I’ve set rtp 1000 to 1100 in fpbx and pfsense
When i speak i can no longer hear the other person.
After checking everywhere, i found out that it’s the duplex that should be the culprit, but i can’t find out how to fix this.
Followed what i found here http://wiki.freepbx.org/pages/viewpage.action?pageId=24051965 and it didn’t change anything.
Any help would be appreciated.
To what? What are you connecting your IAX2 trunks to?
Do you mean “I have a mix of SIP Hard phones and SIP Softphones” or are you trying to say something else.
So, you can hear the other person until you speak, and then you can’t anymore? I’m not clear on what you are saying. Also, where is “the other person”? Is their phone in your local network? Are you in your local network?
You’ve already tried the SIP one-way audio fixes, and that’s good, but those only work for SIP and you are using IAX2 in here as well. Have you set up pfsense to handle the IAX2 connections as well? Did you turn off the SIP-ALG module in pfsense?
Are you redirecting the ports in pfsense from and to the right interfaces?
You opened up the RTP ports and (I’m guessing) redirected them to your local machine, but that’s no clear from this problem definition.
We need more details before we’ll be able to help you solve this problem.
As always, one of the first places to check is the /var/log/asterisk/full log. In there, you will probably find clues about what is going wrong. If that doesn’t help, you might need to do a call trace to see what is getting passed back and forth between the systems.