Because one of our big partners has a lot of Ericsson setups, and you might know that Ericsson == Aastra these days, so the Aastra phones are compatible with Ericsson. If these phones work fine we can get SipX in those companies.
And because the hardware itself is pretty good and good looking, and not too expensive. And because it works just fine on Asterisk... ;) Regards, Niek On Sat, Sep 17, 2011 at 1:42 AM, Michael Picher <[email protected]> wrote: > Why the desire to use these phones so much from an unhelpful vendor? > > On Sep 16, 2011 3:30 PM, "Niek Vlessert" <[email protected]> wrote: >> Hello Tony, >> >> I couldn't agree more with your statement, but that doesn't get Call >> Pickup fixed on Aastra phones. And because Aastra is not doing anything, and >> we need this feature badly I'm asking for trickery. 2 options: remain >> stubborn and require full SIP compliancy or use tricks. I guess Aastra won't >> listen and not supporting this feature will not increase acceptance for >> SipX. >> >> We are fixing the BLF in a bad way for Aastra, but in the most elegant way >> this bad hack can be done. :) It's like problems with the Linux kernel in >> the past; you can fix hardware problems through drivers or tell the company >> to fix their hardware. Sometimes it's good to choose the first option. This >> BLF ticket is from 2008, and nothing happened. >> >> I agree with the Freeswitch thing. Most of the time we try to not involve >> Freeswitch, but it has more flexability because of all the applications and >> no recompiling when changes are made. If we can get grip on the call in SIP >> without using Freeswitch it's less ugly, but we have no idea how. >> >> When using Asterisk we could just listen on the AMI and then bridge the >> call to the phone doing call pickup without doing any RTP. Where do we >> inject like this in SipX? >> >> If you have some trick up you sleeve PLEASE tell me. :) >> >> Regards, >> >> Niek Vlessert >> Telecats >> >> >> Op 16 sep. 2011, om 21:13 heeft Tony Graziano het volgende geschreven: >> >>> Internal calls (where call pickup comes into play) is handled by the >>> proxy. It's always beena goal of the project to intentionally not use >>> a b2bua for every phone connection in order to achieve peer to peer >>> media and scalability. >>> >>> I would really not put FS in that role, it's intention is a media server. >>> On Fri, Sep 16, 2011 at 3:09 PM, Niek Vlessert <[email protected]> >>> wrote: >>>> Hello all, >>>> Some of you might know that Call Pickup and BLF with Aastra phones don't >>>> work on SipX because the Aastra firmware is not compatible. >>>> We already fixed the BLF issue >>>> >>>> (http://track.sipfoundry.org/browse/XTRN-113?focusedCommentId=55875#action_55875) >>>> but now we need Call Pickup. >>>> The problem is that the phone won't respond well to the call pickup SIP >>>> stuff. Is there a way to get control over the call in another way? >>>> Something like this; instead of dialing *78<extension> (which is call >>>> pickup) we dial *79<extension>. In Sipx, add a gateway to local port >>>> 15060, >>>> which is freeswitch, and a route to get the *79<extension> in >>>> Freeswitch. >>>> Freeswitch can execute any script. Is there a way to get to the SIP >>>> header >>>> and bridge the call to the phone which did the *79? I know, not >>>> beautiful at >>>> all, but it's a way… >>>> Some other direction we are thinking is that Freeswitch registers itself >>>> as >>>> a phone on the same extension as the Aastra phone, the dual forking >>>> feature >>>> in SipX. So if the number is dialed, the Freeswitch phone will also >>>> 'ring'. >>>> Maybe we can then bridge that call to the user who did the >>>> *79<extension>? >>>> But it we do that, then every Aastra phone needs a seperate SIP account >>>> in >>>> Freeswitch. Freeswitch can handle that, but that's even less beautiful, >>>> I'd >>>> say very ugly. ;) >>>> Anyone got a trick? >>>> Regards, >>>> Niek Vlessert >>>> Telecats >>>> _______________________________________________ >>>> sipx-dev mailing list >>>> [email protected] >>>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >>>> >>> >>> >>> >>> -- >>> ====================== >>> Tony Graziano, Manager >>> Telephone: 434.984.8430 >>> sip: [email protected] >>> Fax: 434.465.6833 >>> >>> Email: [email protected] >>> >>> LAN/Telephony/Security and Control Systems Helpdesk: >>> Telephone: 434.984.8426 >>> sip: [email protected] >>> >>> Helpdesk Contract Customers: >>> http://support.myitdepartment.net >>> Blog: >>> http://blog.myitdepartment.net >>> >>> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>> Ask about our Internet faxservices! >>> _______________________________________________ >>> sipx-dev mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >> >> _______________________________________________ >> sipx-dev mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > > _______________________________________________ > sipx-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev/
