Greetings List!<div><br></div><div>I'm currently rolling out a new deployment of Asterisk 1.8 to replace existing 1.2 servers...and have run into an issue which could use your assistance!</div><div><br></div><div>For testing I have trunked (iax2) two of the servers - one running 1.8 and the other at 1.2. Calls placed from SIP --> SIP sound fantastic and crystal clear. However, when I place a echo test call (*43) from 1.8 to 1.2 the result is a pulsing echo and garbled audio. The same result is found when dialing into a meetme conference being held on 1.2 from the 1.8 server.</div>
<div><br></div><div>I know that the jitter buffer has gone through quite a bit of work since 1.2 but my results at this point seem to indicate 1.8 is actually worse than 1.2. </div><div>I've also trunked together two 1.8 boxes (physical locations are in different countries), to rule out if 1.2 was somehow causing the extra jitter. My results were identical - echo testing/meetme from 1.8 to 1.8 resulted in the same audio issues.</div>
<div><br></div><div>The new servers are in the same locations as our production units, using the same networks/etc... so I expected identical results between them, not for 1.8 to be substantially worse.</div><div><br></div>
<div>Any recommendations? Has anyone else experienced a similar issue and if so, perhaps you could share your experience.</div><div><br></div><div>Thanks!</div>