I have posted a similar problem earlier on this mailing list with my Asterisk-system + TDM410 + Grandstream telephones. But there has not yet been a response to this.
My client is also experiencing a 'simplex' conversation. There seems that audio can only flow 1 one way at the same time. What I have tried is change the codec on the internal SIP-network from alaw to gsm (so more compression, less bandwidth needed) but problem not yet resolved. Also I don't know where to begin to look for the problem... So, I'm curious for the solution. Greetingz, Jonas. On Sat, 2009-05-30 at 14:35 -0400, Nathanial A. Byrnes wrote: > Hello, > I am working on a trixbox based system with a TDM410P connected to 3 > phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN > with some polycom and Aastra SIP phones. In general everything works. > the problem I am trying to solve is that if both parties to a call speak > at the same time one of the voices gets cut out such that the talker A > cannot hear what talker B is saying. When talker A stops talking, he/she > can then hear what talker B says. This issue occurs across all the > different phones we have set up. I have played with the OSLEC settings > in the thoughts that the echo cancellation was being a bit ambitious, to > no avail. Any recommendations on how to best troubleshoot / correct this > issue? > > Thanks and Regards, > Nate
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