<div dir="ltr"><div>Hello,</div>
<div> </div>
<div> Sorry for the delay - was out of office. I also cross-posting it to OpenSIPS list.</div>
<div> </div>
<div>I have a small pilot (20-30 phones) which also does some sort of SIP to PRI transcode for our old PBX. The pilot is base on Asterisk and mostly Polycom-501 phones. It works quite well, but I have a few minor/missing issues:</div>
<div>- I have the RPID patch, and unattended transfers fails with it.</div>
<div>- No SLA, only BLF. I know there is SLA, but it is cumbersome to deploy.</div>
<div>- Confference is limited to 3 participants. I guess I can do more with external server but didn't</div>
<div> manage yet to make it working.</div>
<div>- No "busy dial again" which is required by our users.</div>
<div> </div>
<div>Now, to the original issue: I tried adding 1000 extensions to the SIP database, and then use SIPP to send one REGISTER for each extension. After doing so Asterisk still worked, but it was continously accessing the database for all these extensions, just polling them. This raised a red flag to me, and I decided to check the following config: OpenSIPS/Kamailo/etc. as registrar and "SIP switch" for the phones, while using Asterisk only for media related issues (which is the common suggestion here). Now, I have new problems:</div>
<div> </div>
<div>- SLA works, but very "fragile".</div>
<div>- Not BLF, although I think it will be solve with the dialog handling on OpenSIPS 1.5</div>
<div>- Same confference and "busy dial" problem.</div>
<div> </div>
<div>Next week our management is going to decide (I hope...) how to proceed: Do nothing (stay with the Nortel as we are tight on budget), go to open source or to a commercial solution.</div>
<div> </div>
<div>Although a commercial solution allows me so sleep well at night, I am going to recommend the open source direction. If accepted, then I will continue the development and you'll hear me quite a lot here asking hard questions :-)</div>
<div> </div>
<div>BTW, If I didn't say so far: we have around 8,000 extensions on 4 Notel PBX'es, using around 10 PRI's to the world.</div>
<div> </div>
<div> Regards, __Yehavi:<br><br></div>
<div class="gmail_quote">2009/3/17 Vincent Li <span dir="ltr"><<a href="http://vincent.mc.li/" target="_blank">vincent.mc.li</a>@<a href="http://gmail.com/" target="_blank">gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div><br><br>On Tue, 17 Mar 2009, Yehavi Bourvine wrote:<br><br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hello'<br><br> I am at the same situation as you. I also work at a university and we have<br>over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot.<br>
<br> I am using a realtime users database and the main problem is that Aaterisk<br>does too mcuh database access to inquire for the currently registered users.<br>(I am using direct RTP path between the phones so this is not a limiting<br>
issue here).<br><br> I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS<br>will serve the phones and Asterisk the more complicate things (voicemail,<br>transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they<br>
are being worked on.<br><br> Regards, __Yehavi:<br><br></blockquote><br></div>Hi Yehavi,<br><br>Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time.<br>
<br>Regards,<br><br><br></blockquote></div></div>