Use a better phone. The m3 is a consumer grade device. I have no issues with timing on 4960's when configured properly. On Jun 12, 2012 2:44 AM, "milosz" <[email protected]> wrote:
> > It is probably not getting or missing an ack to indicate the call > established. This would indicate a carrier or gateway config issue. > > the m3 seems to be blanking the route header field. take a look at the > trace. > > > I am curious what this actually has to do with sipX? > > answer: something. > > > Adtran's SIP stack, while it has gotten MUCH better, is still garbage > (when > > used as a gateway). It was designed with Asterisk in mind and is pretty > much > > limited to routing by numbers only (which makes a lot of alpha-numeric > stuff > > sipX does not work properly). > > i haven't had too many issues, but i've only used it for very basic > deployments. it is cheaper and less horrible to configure than the > audiocodes and is cheaper than and has a superior timing source to the > basic patton 4960. the stack on the patton is the best but it's often > difficult to justify spending 4x the money. > > the call disconnect behavior is not limited to the adtran, i had it > happen with the patton as well. > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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