Isaac Schneider

I am doing an initial test using FreeBSD 7.1-STABLE and 8.0-RELEASE-200902 kernels. I was looking to see if anyone has implemented PBXnSIP/FreeBSD on a virtualization platform like XenServer? Now that the enterprise edition of XenServer is free I was looking to do baremetal with Xen & BSD. I seem to recall durring training a provider using PBXnSIP/FreeBSD kernel parameter kern.hz=1000 (under VMware?). Currently I am using HP ML350 G5 w/E5300 series Quad core Xeon, Local SAS Storage, and GigE i/o with XenServer 5 The VM slice is configured with 1 cpu core no CPU cap,256mb ram. The kernel parameter is set kern.hz=100 and have not noticed any side effects with initial basic testing. Thanks, Isaac

MichaelW, A visio style overview would be great best would be drag-n-drop visual call flow programming. Adtran does visio style display on their routers and multifunction units in a limited fashion no drag n drop though. 3com has a non graphical hierarchy that uses html pulldown boxes that turn into links after applying them within an auto attendant menu. I know that you are able to replace some of the html coding within pbxnsip. Durring training the head guy at PBXnSIP Christian was showing a ton of customization flexibility. I do not know how limited the functionality is, but maybe it could be possible to add some javascript magic to show this information. I have already asked my contacts at pbxnsip for more detailed examples. -Isaac

I just got back from the PBXnSIP training and it was mentioned that the 3.x branch supports video in a limited mannor. I was trying the Counterpath XPro client but am not having luck establishing call or adding video to existing call. From my other vendor trainings this is a design problem of the PBX. When using a B2BUA architecture the phone system needs to know what each option is in the SIP and RTP streams. If it does not know it drops the info from the relayed stream. Is anyone playing with this? Thanks, Isaac

Rix, I found early on as a Zultys reseller, their phones default to 10ms packetization with whatever codec in the earlier releases of firmware. The newer releases I believe default to 20ms packetization. I never thought that would be an issue except for higher packet header overhead. I had the same issue where if one side (20ms SIP UA) initiated the call (to a Zip) it would work but if switched (Zip to 20ms SIP UA) it would not. You can set this via the ZIP phone's web gui of the TFTPconfig file. Most other vendors phones do not have a good dsp/codec that can handle the higher io rates at 10ms. Can you believe that the Zip 4x4 and 4x5 had a powerpc cpu (full, not ip core) in each unit! Talk about horsepower... no wonder they went bankrupt. At the engineering level the Zip 4x5 were way ahead of their time. VxWorks RTOS, IPsec VPN, Dynamic Call Encryption, POTS ATA, Bluetooth, etc etc... I loved the PCB engineering and sheet metal design of the MX1200, MX250 chassis. However the industrial designer of the ZIP series responsible for the shell and layout should have been shot. The phones all felt cheap (nice way of putting it). Did not hold up well under high volume use. The phones were a hard sell when viewed after a Aastra, Snom or Cisco IP Telephone. I still use the 4x5 internally and still love them even with their flaws. Getting harder to keep the ZIPs arround when phones like the Snom 820 & 870 are becoming available. 8-) -Isaac