Hello ppl, Is it possible to include contexts in the RealTime scenario?? If not, wots the work around?? Thanks in advance. Ben.
>-----Original Message----- >From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >Benjamin Jacob >Sent: Monday, September 04, 2006 8:37 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [asterisk-users] includes in realtime ?? > >Hello ppl, >Is it possible to include contexts in the RealTime scenario?? >If not, wots the work around?? > >Thanks in advance. >Ben. >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersAmazing how the wiki has this vast amount of AT LEAST info to start your research on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
Rushowr wrote:>>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >>Benjamin Jacob >>Sent: Monday, September 04, 2006 8:37 AM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: [asterisk-users] includes in realtime ?? >> >>Hello ppl, >>Is it possible to include contexts in the RealTime scenario?? >>If not, wots the work around?? >> >>Thanks in advance. >>Ben. >>_______________________________________________ >>--Bandwidth and Colocation provided by Easynews.com -- >> >>asterisk-users mailing list >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > >Amazing how the wiki has this vast amount of AT LEAST info to start your >research on >http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions > > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >Sorry mate. Just slipped the eye. Now to another question, which I tried about. With the Realtime arch, can we change parameters of certain users, say sipusers, at runtime, for e.g. the codec and the change being reflected back immediately? The two SIP users I had, had allow set to "gsm;g729;ulaw;alaw", and the two Xlite phones have gsm,ulaw and alaw configured.Calls work fine . I changed the codec(set allow to have only g729). But still the calls go thru. I tried realtime load sipuser name <username>, to no effect. (anyway, realtime load is only for reading values, if i am not wrong). So is it possible to change user parameters at realtime? or am I missing something again? Thanks again. Ben.
If you want to use MWI, and I imagine most people would, you have to cache your realtime data, which means that changes to the tables do not become effective immediately. They become effective after you prune the entry in memory. Doug.> -----Original Message----- > From: RR [mailto:ranjtech@gmail.com] > Sent: Tuesday, September 05, 2006 12:26 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] includes in realtime ?? > > > Ben, > > The family name is not sipuser, its sipusers. So try this command > > "realtime load sipusers name <username>" and see if you get > nothing. What about? > > realtime load sipusers username <username> ? > > To answer your question, any change in the tables holding this sip > users information comes into affect immediately. That's the whole > point of realtime :) > > Cheers, > \R > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hello, The service I am using requires authentication. In sip.conf, setting: [authentication] auth=name:pwd@myprovider.com Gets the authentication working for the INVITES but when I try a transfer, I can see the REFER but then asterisk quickly says BYE. The provider sends back a 401 UNAUTHORIZED but asterisk never resends the REFER with the required authentication info. Is there a way that I can get Asterisk to authenticate on REFERs? Regards