Hi guys, I’m using FreePBX distro, with Asterisk 13.
I have a Sip trunk service using ChanSIP and have no problems receiving inbound calls. I can also make calls between handsets internally without any problems.
I am having problems making outbound calls.
If I make an outbound call, the external party receiving the call can hear me, but I can’t hear them.
HOWEVER, if I transfer a call to an external number no problems.
If I setup auto call forwarding from an extension to an external number no problems.
Incoming calls: OK
Internal calls between extensions: OK
Auto call forwarding from extension to external: OK
Transfer to external: OK
Call external number: Can’t hear recipient, recipient can hear me.
This is my sip peer settings (I have confirmed the with my Sip trunk provider the codecs they support).
What is your Asterisk version? Is it 13.13.1 ?
FreePBX version is10.13.66
Asterisk version 13.12.1
NAT=NO from your trunk setup and check
@dcitelecom . Thank you for your advice that fixed the problem.
@dcitelecom I’ve come into the office to test (as previous test was from a softphone outside the office) and I have found that NAT=YES has not resolved the problem when the outgoing call originates from a Yealink T-23G and Yealink T-46G handset.
I stumbled on a strange bug on the last versions of Asterisk and Yealink phones.
This bug was already reported but not resolved yet.
Try to remove all the codecs from your ip phones except one codec (alaw) and retest.
You are not providing enough info.
Is the extension NAT mode "YES force rport comedia)?
Is the firewall on your router disabled?
Is SIP ALG disabled on your router?
Problems solved by changing extensions from PJSIP to CHAN_SIP.
I doubt it was was a firewall/router related problem given that outgoing calls worked if they were forwarded/transferred out, and weren’t working when they were directly directed from the extension.
Thanks for your message, the problem was resolved by changing extensions from PJSIP to CHAN_SIP. When I visit the office next, I’ll revert an extension back to PJSIP and try changing the codec to just alaw and report back whether it was a bug with yealink and codecs.
It is good to hear that you solved your problem. I would stick to the chan_sip if you want peaceful life with your blf buttons and the presence.
In any case, the bug of the Asterisk that I mentioned is on 13.13.1 version.