[asterisk-users] No audio until you put call on hold...

[Carlos Chavez]

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	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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