I’m having a problem when forwarding external calls to other internal extension, with no audio to the person who is calling from outside.
I’ve a trixbox CE version 220.127.116.11 and Asterisk 1.6.0 with SIP Trunk to the operator.
So, the scenario is:
Someone calls me and we talk without problems, when I transfer the call using the *XXXX (indirect transfer) the person who is calling from the outside can’t hear nothing while we can hear.
If I transfer using #XXXX (direct transfer) it works just fine.
If using the xfer button option (cisco IP phone) or transfer button (Snom phones) it also works.
In the situation I was talking that doesn’t work, If the person from the outside press a button it will bit and the audio starts!
I don’t have any firewall or NAT, since it has a public IP and connects directly to the operator.
SIP Trunk Configs:
I’ve tried options that people suggest but without any luck, like:
Please help me with ideas or suggestions.
Thank you in advance.
Did the system worked before? How long you have this going on?
Yes, the system worked before and I can’t remember any modification that could make this problem.
This happened first time around 3 months ago, first time a user complained. I made a few tests back that days and didn’t found any problems (I used the ways it works for tests), recently more people complained and I found this situation.
About 6 months ago we changed the VoIP Operator. We used an analogic line before and with the change, we now use a SIP Trunk.
Can anyone help me out on this, please?
Would it be a huge deal for you to upgrade to FreePBX 2.11… It is free.
TrixBox is not a supported by anyone product, not anyone here really uses it.
That said, it looks like your problem is in RTP traffic rejection or timeout. If something worked before and absolutely nothing have changed, there is no reason why this would not work today. Something has changed, perhaps phone firmware is updated and some settings are changed. It looks like your expiry settings are pretty big (1800), usually these are 60, but if they worked before than they should work now, plus it would affect bunch of other things. Look into RTP timer settings on the phone sets. Set those to something small, like 60, set UAC setting on the phoneset. Your problem is in phoneset not in TrixBox. TrixBox has been dead for years and there is no way something got changed in it, but phone is a different story.