Hi Michal, That sounds great. Let me know of the progress. Maintaining one box is way much better than maintaining two applications. But does this means that what I am trying to accomplish is not doable using asterisk?
Regards, Mac On Mon, Feb 7, 2011 at 11:24 PM, Michal Bielicki < [email protected]> wrote: > You could use freeswitch. > Actualy I am working on having the freeswitch in sipx come with wanpipe, > openr2 and libsng_isdn. > If that is enough for you let me know, I can prioritise this a bit more :) > > Am 07.02.2011 um 16:02 schrieb Leonimar Cape: > > Hi Tony, > > Thanks for the reply. Its sipx version 4.2.1-018971. Asterisk is connected > to pstn via e1 isdn pri and mfcr2. I could not remove asterisk since sipxec > doesn't support pstn connection unless I will use different gateway such as > audiocodes. > > Call is originating from the pstn passing thru the asterisk and into one of > the extension in sipxecs and the one who will initiate the callpickup is > also another extension in sipxecs. > > Any suggestion? > > Regards, > > Mac > > On Mon, Feb 7, 2011 at 6:27 PM, Tony Graziano < > [email protected]> wrote: > >> what version of sipx? what kind of pstn connection is it? it seems it >> would be far simpler to remove asterisk (IMO). two different systems to do >> "one" thing is really making it more complex than it needs to be. >> >> On Mon, Feb 7, 2011 at 4:59 AM, Leonimar Cape <[email protected]>wrote: >> >>> Hi Guys, >>> >>> Hope you can shed me some light on whether what I am trying to accomplish >>> is possible and what is the right approach or what I am doing wrong in my >>> current setup. >>> >>> Currently I have setup an Asterisk which acts as a media gateway and >>> handle all pstn calls to voip and a Sipxecs in distributed mode where all >>> the extensions/locals are registered. Asterisk and Sipxecs are connected >>> using the unmanaged gateway configuration. Everything works fine until I was >>> asked to make the call pickup works. >>> >>> I have checked that yealink phones do support the rfc4235 features which >>> needed to make this work. Asterisk 1.6.2 states that it is also supported. >>> But I can't make it work. >>> tshark trace shows that the SUBSCRIBE message is being send to the >>> primary and distributed sipxecs but it seems that I can't find the asterisk >>> connection and returning back a "404 not found" response. >>> >>> Any feedback would be much appreciated. >>> >>> Regards, >>> >>> Mac >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: [email protected] >> Fax: 434.326.5325 >> >> Email: [email protected] >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: [email protected] >> >> Helpdesk Contract Customers: >> http://support.myitdepartment.net >> >> <http://support.myitdepartment.net/>Blog: >> http://blog.myitdepartment.net >> >> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > *Michal Bielicki* > Geschäftsführer / CEO > > *Seventh Signal Ltd. & Co. KG* > Weigandufer 45, Büro 115, D-12059 Berlin > Voice: +49 30 60988730 > > Amtsgericht Charlottenburg HRA 44413 B > Ust.-ID: DE266981999 > Geschäftsführer: Michal Bielicki > Persönlich Haftende Gesellschafterin: > Seventh Signal Ltd, 69 Great Hampton St. Birmingham, > B18 6EW, GB, Company Nr.: 06889439 > WWW.: http://www.seventhsignal.de > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >
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