<div>Hi everybody,</div>
<div> </div>
<div>I finally want to get rid of 1-way audio problem. Please help me here.</div>
<div> </div>
<div>I have 3 scenarios.</div>
<div> </div>
<div>1. Audio is always one way. Caller who dialed can't listen the called party but called party can listen him. In this scenatio Asterisk is on dynamic IP with dyndns FQDN. sip.conf has externip = <a href="http://abc.dyndns.org">
abc.dyndns.org</a> and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is the voice getting lost from the called party? NAT is there but Asterisk is in DMZ.</div>
<div> </div>
<div>2. Conversation is going fine when all of a sudden you realize that other parth has started saying 'hello, hello' because they can't hear you. But you are hearing them loud and clear. Now you are on static IP with dyndns FQDN. externip and localnet settings in
sip.conf (do we need them for static IP?). After about 15-20 seconds, again 2-way converstaion is established again. IAX trunk, SIP extension, no NAT.</div>
<div> </div>
<div>3. Conversation goes one way for 15-20 sec during the most important part of the conversation (Murphy's Law). You are on a static IP with no dyndns enrty. Trunk is ZAP on PRI, extensions SIP. NAT present but router properly configures for port forwarding. externip and localnet settings present in
sip.conf</div>
<div> </div>
<div>Is think may be due to some reason RTP stream gets lost, routed to wrong IP. But why would this happen during a call and how to stop it from happening. Or is there some other reason behind this? Does dyndns setting have to do anything with this problem? How can I overcome this problem once and forever.
</div>
<div><br>-- <br>Zeeshan A Zakaria </div>