Call audio & Connection issues

I have searched the internet for hours to try and figure out a solution to this problem with no avail. Hopefully somebody on here is able to point me in the right direction as to how to fix this issue

The problem is that we do not have 2-way audio when making SIP trunk calls. If the calls are made internally or via our PSTN line then the audio is perfect both ways.

Only audio originating from the PBX is sent to the external phone, no audio from the external phone can be heard on the internal phone. So far our pbx has rtp ports 10000 - 20000 selected and forwarded and i have manually entered our external and internal IP sets into the SIP settings. I have confirmed that the correct codecs are being selected for calls and i have selected yes to NAT in the sip settings as the PBX is behind a Sophos UTM.

I cannot see any further config that should be needed to make the SIP trunk calls work properly so I would appreciate any suggestions

What about the Nat mode setting in the extension itself. Extension->Advanced->Nat mode->Yes

NAT Mode was off for the extension and i put it in Yes mode but still didnt seem to make a difference

There are lots of other NAT related configuration items, including in the SIP Settings under the Advanced tab. You also need to make sure that your “external” (firewall) address is set correctly in the SIP settings. Remember that PJ-SIP has the same settings and needs to be configured if you are using PJ-SIP instead of Chan-SIP.

Are you talking about the advanced tab under extensions or under Chan SIP settings?

Our external firewall address is defiantly set correctly, we have a static IP from our ISP which is set statically in the SIP settings, and under the firewall settings in the PBX we have our ‘internet classed LAN’ address set correctly. I haven’t configured any settings for pj-sip as we are currently only using Chan-SIP for all communication

Under the Advanced Tab, there are some SIP settings that you need to verify.

Do you mean these advanced settings?

Or the advanced settings tab i see within an extension settings menu?

Yes. There should be some SIP Settings in there.

These are the only settings i have found there and they all look fine to me, I see that SIP nat is set to yes so these settings shouldn’t really be causing a problem?

Also a bit further down i can see the channel driver is set to chan_sip which is also correct

Bumping this issue to see if anyone can suggest a solution

I wanted to follow up on this as I see other people are having similar issues as I am. I’m told a fix is to roll back to an earlier version of asterisk with yum downgrade asterisk13*-13.14.0

I am going to try this myself tonight after hours. I think the affected version is 13.15.0

Following up to say that running the command yum downgrade asterisk*-13.14.0 after restarting asterisk resolved the issue.

I was then able to run all other upgrades and reboot the server.