Running * 1.6 on Ubuntu 8.04
I’m having an issue when a external call is placed the person answering cannot hear the caller for about 3 seconds. This isn’t on every call. Any ideas?
could be a number of issues.
1 - network latency in relaying the SIP 200 OK message.
2 - firewall issue. If you aren’t allowing data to come in on the RTP ports, and you are relying on your firewalls OUTBOUND/RELATED rules you may have some delay there. By default, Asterisk uses UDP 10000-20000 for RTP. You can try opening up the firewall to allow UDP on those ports to see if it fixes things.
Thanks very much for the replies. I think i’m starting to tie it to latency. It seems to happen most often when I get notifications that the cicso router is maxed… It’s also happening on all 24 channels too.
The asterisk server has it’s own T1 for voice so I don’t think is the firewall blocking any data. It connects to to comm box.
i also just assumed you were using SIP… if so, latency in call setup from the SIP messaging can certainly cause a delay.
Internally we are using SIP. Externally, we are PSTN using DTMF wink on all channels.