Nick
>From my perspective the issue is that both parties, the sipXecs project and the phone vendor, have to cooperate to make this happen. This is important not only to get an initial version to work, but also in order to be able to support a solution going forward. Aastra would have to agree that sipXecs is a platform they support and therefore accept support tickets from companies or users of their phones in conjunction with sipXecs. Therefore, in my view this is not primarily a technical problem. We need a relationship with Aastra and therefore someone should take the lead, contact them, and see what they would be willing to do. --martin From: sipx-dev-boun...@list.sipfoundry.org [mailto:sipx-dev-boun...@list.sipfoundry.org] On Behalf Of Niek Vlessert Sent: Sunday, September 18, 2011 5:41 AM To: sipXecs developer discussions Subject: Re: [sipx-dev] Aastra Call Pickup Hello Todd, Thank you for this very informative and interesting answer. It's a very good thing that SipXecs is trying hard to remain in the boundaries of open standards. Doing this will make sure a huge amount of other problems will never happen. But I'm not sure that I'm not following open standards when doing things like this. I agree, a phone should just follow a few RFC's concerning call pickups, and then all will work. Brands should just do that! But what if a phone does not support Call Pickup? Then there's 2 options; wait for the phone to support it, or create a helper thing around it. In Asterisk of FreeSwitch land I can easily create a helper without having any RTP through the proxy. So I'm not breaking the thought of letting the proxy be a proxy and I'm also not breaking any RFC. I'm simply watching channels coming in and then bridging them based on a user dialing some number. One can look at it as an add-on instead of breaking a philosophy. The only ugly-ness in it is that I'm taking system resources to fix a problem a phone should fix. Advantage of this is that ALL phones will have call pickup support on SipX. Something like this: *78 means RFC based pickup, *79 means channel bridge based pickup. I guess it's a matter of opinion about strictness. I might oversee things like HA & Fail Over (what if a Call Pickup comes in on server B for phones on server A?), don't know enough yet about it. One major difference between Asterisk & SipX is that in SipX land the phone should take care of a lot of stuff, in Asterisk it's the other way around. When doing things like this, it's true that I'm breaking this philosophy. But what is the difference between this fix and the MOH on Polycom? Polycom will redirect the audio to the media server, so MOH is on the server. Same issue with the philosophy. Come to think of it, there's a software called Voice Operator Panel which bridges calls all the time on SipX, so I will have a look at that, maybe that will provide an answer on how to get in the call in a similar way as AMI. I will also look into your suggestion about other people trying to fix SIP incompatibilities. If we succeed I will still supply the patch. ;) Regards, Niek On Sat, Sep 17, 2011 at 9:48 PM, Todd Hodgen <thod...@frontier.com> wrote: > I'm not the owner of Sipfoundry, or a member of the developers team, so I > can't speak with authority to the subject. But from what I have seen in the > history of this mailing list for the past few years, the fixes need to be > within the confines of the appropriate RFC. This project uses open > standards, and as such, fixes to make Aastra work with sipXecs should be > based on using only those open standards. > > Possibly, you need to look at development of your own gateway to sit between > sipXecs and the Aastra phones to provide sip truing, if that is even > possible. If the opportunity for your firm is great, you may want to > investigate the business case for development of some custom software to > work between the applications. > > Citel is an example of a company that builds devices that work between a sip > server and proprietary phones - like Nortel, Avaya, Panasonic, NEC, Toshiba, > etc. They also make gateways that allow the old legacy PBX products to be > extended over IP networks. Many of the SBC's on the market do truing > between disparate SIP implementations, so there is no lack of demonstration > of products that merge incompatible SIP implementations, or similar. > > A couple days legwork looking at what options might exist may be time well > spent for your business opportunity. Some of the developers might have some > suggested avenues to try. > > > -----Original Message----- > From: sipx-dev-boun...@list.sipfoundry.org > [mailto:sipx-dev-boun...@list.sipfoundry.org] On Behalf Of Niek Vlessert > Sent: Saturday, September 17, 2011 12:21 PM > To: sipXecs developer discussions > Subject: Re: [sipx-dev] Aastra Call Pickup > > Hello Todd, > > The problem is obvious in the Aastra firmware. We tried to get Aastra to fix > the problems, but they don't seem to care; it works on Aastra PBX's, > Ericsson PBX's and Asterisk. We are still pushing Aastra, but it's a > difficult process, and other tried and have not succeeded. > > A fix to the firmware would definitely benefit the community. But won't a > completely functional work around as well? > > Regards, > > Niek > > On Sat, Sep 17, 2011 at 8:50 PM, Todd Hodgen <thod...@frontier.com> wrote: >> It seems for you, that it would be prudent to document the issues with >> the Aastra phone, determine if they are SIP standards issues, or >> issues in the template of sipXecs, and develop a corrected template >> for them, and work with Aastra to fix their deficiences. You can find >> cost effective firms out there to develop software for this open >> source project. Fixing the issues that are keeping you from >> deploying to your customer, will not only fix that for you and create >> sales for your firm, but benefit the community as a whole >> >> -----Original Message----- >> From: sipx-dev-boun...@list.sipfoundry.org >> [mailto:sipx-dev-boun...@list.sipfoundry.org] On Behalf Of Niek >> Vlessert >> Sent: Friday, September 16, 2011 11:38 PM >> To: sipXecs developer discussions >> Subject: Re: [sipx-dev] Aastra Call Pickup >> >> 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 <mpic...@ezuce.com> wrote: >>> Why the desire to use these phones so much from an unhelpful vendor? >>> >>> On Sep 16, 2011 3:30 PM, "Niek Vlessert" <niekvless...@gmail.com> 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 >>>>> <niekvless...@gmail.com> >>>>> 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=5587 >>>>>> 5 >>>>>> #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 >>>>>> sipx-dev@list.sipfoundry.org >>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> ====================== >>>>> Tony Graziano, Manager >>>>> Telephone: <file:///C:\Program%20Files%20(x86)\eZuce\openUC%20Outlook%20Add-in\StatusIm ages\phone.png> 434.984.8430 >>>>> sip: tgrazi...@voice.myitdepartment.net >>>>> Fax: 434.465.6833 >>>>> >>>>> Email: tgrazi...@myitdepartment.net >>>>> >>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>> Telephone: <file:///C:\Program%20Files%20(x86)\eZuce\openUC%20Outlook%20Add-in\StatusIm ages\phone.png> 434.984.8426 >>>>> sip: helpd...@voice.myitdepartment.net >>>>> >>>>> 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 >>>>> sipx-dev@list.sipfoundry.org >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >>>> >>>> _______________________________________________ >>>> sipx-dev mailing list >>>> sipx-dev@list.sipfoundry.org >>>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >>> >>> _______________________________________________ >>> sipx-dev mailing list >>> sipx-dev@list.sipfoundry.org >>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >>> >> _______________________________________________ >> sipx-dev mailing list >> sipx-dev@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >> >> _______________________________________________ >> sipx-dev mailing list >> sipx-dev@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >> > _______________________________________________ > sipx-dev mailing list > sipx-dev@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > > _______________________________________________ > sipx-dev mailing list > sipx-dev@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > _______________________________________________ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
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