… but I’m receiving the other end just fine.
You see, every time I call or someone calls me over a sip trunk, I can hear them, but they cannot hear me.
But If they get to the voicemail, they can hear my prompts just fine.
http://www.maas-martin.nl/wiki/doku.php?id=projects:asterisk this is my sip.conf and a debug from the console
Perhaps someone has a suggestion?[/code]
Oh, and yes they can leave a message which I can then listen to.
Looks to be a NAT firewall issue. It seems that asterisk has a good audio path to both SIP devices. That being said, the easiest solution is to keep asterisk in the media path. Right now your UAs are issuing re-invite requests to take asterisk out and pass RTP directly between them. While this is very efficient it is also very tricky to configure properly with NAT environments.
I would add
reinvite=no
canreinvite=no
to your SIP config for the SIP extensions in question.
[quote=“g2010”]
reinvite=no
canreinvite=no[/quote]
I did not even realise those where two different options… I kind of mixed them up in my mind…
anyway, it did work, both parties can hear one another, but the sound is quite jerky and tends to fall out.
But that’s my problem to work with for a while.
Perhaps you know a good link about the correct NAT setup to get reinvites working etc.
Thanks,
MeneM
I’m guessing it just needed some time to settle or something, because the sound is perfect now.
Thank you for your suggestion!
There are some tricks you can do to try and get direct audio path from UA to UA without passing through asterisk. In my opinion it’s probably a waste of time and will cause more headaches than it’s worth.
Assuming you aren’t trying to support tons of concurrent calls on your server you should be fine allowing asterisk to stay in the media path. You actually gain a lot of functionality for very little downside. It certainly makes things easier from a management point of view.