Maybe you have a Codec issue? On Tue, Oct 28, 2008 at 11:20 PM, Benny Amorsen <[EMAIL PROTECTED]<[EMAIL PROTECTED]> > wrote:
> Lincoln King-Cliby <[EMAIL PROTECTED]> writes: > > > Periodically I'm seeing calls placed from the 7961s through anything > > on the PBX that requires digit entry (the Auto Attendant, Voicemail, > > etc.) 'randomly' drop; extension-to-extension calls > > extension-to-PSTN, and PSTN-to-extension calls never have any issues > > whatsoever. Nor have I been able to duplicate the issues hopping > > around auto attendants on an inbound PSTN call. > > I am not sure this is relevant in the 1.4.x versions, but here goes > anyway: > > In Asterisk 1.2.x it could sometimes happen that Asterisk believed the > path to a server was so good, that it would only allow 1 ms for > answers to be received. It would do all its retransmissions in less > than 200ms, and then it would complain about no reply to critical > packet. > > Anyway, you can adjust the minimum timer with the configuration option > t1min in sip.conf. I would recommend setting it to at least 100 (it is > in ms) and perhaps 500 would help for you. > > It is also highly possible that your issue is completely different. > > > /Benny > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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