That looks like the option that will help a lot. Thanks.
On 8 July 2010 23:21, Steve Edwards <[email protected]> wrote: >> [mailto:[email protected]] On Behalf Of Julian >> Lyndon-Smith >> >> We have had 20 calls over the last month where the SIP channel has not >> identified that the person on the receiving end has hung up. >> >> Is there a way of fixing this ? > > On Thu, 8 Jul 2010, Danny Nicholas wrote: > >> First thought is that you can put a timeout on your calls, but that is >> just a "band-aid". > > Also not fixing the source of the problem, but rtpholdtimeout and > rtptimeout may help. > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards [email protected] Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _____________________________________________________________________ > -- 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 > -- _____________________________________________________________________ -- 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
