Could Problem number two be caused by incorrect Refresher, or timer settings? If so, what should they be?
On the gateway: *Session Expiration: * (in seconds. default 180 seconds) * Min-SE: * (in seconds. default and minimum 90 seconds) * Caller Request Timer: * Yes No (Request for timer when making outbound calls) *Callee Request Timer: * Yes No (When caller supports timer but did not request one) * Force Timer: * Yes No (Use timer even when remote party does not support) *UAC Specify Refresher: * UAC UAS Omit (Recommended) * UAS Specify Refresher: * UAC UAS (When UAC did not specify refresher tag) -Bryan Anderson On Wed, Apr 18, 2012 at 10:37 AM, Bryan Anderson <branderso...@msn.com>wrote: > I have been having issues with a new Grandstream GXW4104 fxo gateway and > was wondering if anyone could help. > > We have 4 pstn lines from qwest going into the gateway. All calls go to > an Auto Attendant when answered. > > the two problems we have experienced are: > > 1) After about 1-1.5 hours the call hit the Auto Attendant but wont > transfer out. Some dials and extension they just get dead air. (this is > fixed by rebooting the gateway.) > > 2) The external uses (either some one who called it, or some one we have > called) stop hearing audio, but we can still here them. This happens > anywhere from 1-10 minutes into the call. > > sipXecs (4.4.0- 2011-08-14EDT19:47:12 domU-12-31-39-04-D4-A1) > > Grandstream firmware: Program--1.3.4.13 Loader--1.1.3.4 Boot--1.1.3.2 > > The phones are 1 Polycom IP 550 and 7 Polycom SoundPoint IP 331 > > -Bryan Anderson > > > > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >
_______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
