Carlos Chavez wrote: > > I have been struggling with an audio issue for a week now and have > not been able to solve it. > > We have an Asterisk server (now running 11.4 but started with 1.8) > with several sip phones on an internal network and a SIP trunk for > external calls. We recently put several phones in service that > connect via the Internet to the server. All NAT settings and port > configurations were done and all phones register. The problem we have > is that when external phones dial a pstn number they get no audio. We > found that if you dial and put the call on hold for a couple second > you then get audio on the call. > > I really do not know what else I can check in the configuration. Why > would putting the call on hold get the audio flowing? Any ideas or > recommendations?
Carlos, Please provide SIP traces of both call legs (external phone to Asterisk and Asterisk to SIP trunk) annotated to show when the audio starts as well as the CLI output of 'sip show settings', 'sip show peer <external phone>', and 'sip show peer <SIP trunk>'. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
