It is a very old problem, but still we are having this problem once in a while.
When an incoming call is answered, the caller can not hear the callee for a initial 2 a 3 second, after that all audio goes ok.
It does now always occur.
Also tested it with no firewall or nat.
When looking into documentation I found a setting in asterisk.conf: internal_timing = yes
I did not tested it yet, but could it be a solution?
I think it has something to do with rtp stream settings.
Does this problem occur to more people or is it a know problem? And is there a direction to look for in Asterisk or Freepbx to solve this or bypass this problem?
I hope someone can advice me or tell me some hints.
Many thanx in advance.
Important to be aware that call control (SIP) and call audio (RTP) are 2 different things. One is port 5060 UDP and the other is dynamic port 10000-20000 and usually UDP by default. You probably know that but important to point this out because therein is probably where your problem lies.
Also, if you have canreinvite=yes try change it back to canreinvite=no. Now called “directmedia” so look for either label.
I already know that rtp goes over udp 10000~20000.
Also I know that reinvite could be a problem so I always set this to No.
Thanx for your tips but in that direction I already looked.
I seems that RTP stream is trying to connect but the call is already setup.
Tomorrow another day.
You need to handle “early media” appropriately as yo how your VSP behaves.
Sorry Dicko, I don’t understand.
What do you mean with VSP?
The term early media is something I have seen in a Patton ISDN gateway.
Can I set it somewhere in asterisk or freepbx?
VSP = Voip Service Provider
"early media" is audio “rtp traffic” that is sent by the VSP before “answer” is given on that channel, apparently “. . for a initial 2 a 3 second” . . .
A quick workaround is to send the inbound call to an announcement that answers the call first with var/lib/asterisk/sounds/en/silence/1, then to your original destination.
Wow Dicko that suddenly makes sense!
With this customer we have 2 different VSPs so that explains why the problem is not alway’s there.
I could be a VSP setting.
The workaround sound realy realy good and simple. I will try this, but it all makes sense to me now.
Thank you very much!!
I apologize in your case use /var/lib/asterisk/sounds/nl/silence/1. it is similar to the english version but more gutteral