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I have an existing asterisk-voicemail server, that works stable. It is a asterisk server with a SIP-trunc connection to a sip-Server. As soon as a call arrives at the trunk, the asterisk Voicemail (VM) takes the call after a few rings, plays an recorded intro, than plays a sound "beep" and waits for a message of the calling party. That message is recorded and than send via email to me.
That all works well and scope of this project is _NOT_ to change anything here.
However, during the implementation of that system (done by a very expert asterisk expert) we noticed, that the recorded-intro had bad quality, whereas the recorded message was good quality. We were able to identify, that there were NO network issues and finally the consultant was able to do a small configuration of the server machine (not within the asterisk setup) or lauched a service or something like that, that instantaneously solved the issue. From than on we had very good quality of all messages, also the recorded-...
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