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?
Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología
+52-55-91169161 ext 2001
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