Inspire

Same issue here. Axom and IVM. Caller comes in. Have an OGM saying for John 101, sue 102 etc. Caller dials ext and does not transfer. Then tried setting Opt1 for john, opt2 for sue, etc and it will tranfer but will not roll to voicemail or alternate ext if not answered. This is pretty basic for a phone system but I cannot seem to get it to work.

I got mine running after I realized that the documentation was not correct. The Axom MUST call to the Sipura in the external lines config (where they say to leave it blank. Not right, must put IP of Sipura in there. I included the port number as well. Then told my Sipura to dial if not registered to yes.

Try setting the remote gateway option to the Axom server. BTW is your VPN allowing Split tunneling?

I agree this is a real bummer. Kind like to use my soundcard for other purposes only need it to stream not play out the speakers.

With MOH server Shutdown click on start, go to run, paste this in : C:\Program Files\NCH Swift Sound\IMS double click on default.nch. Should reload it and play.

I do not have first hand experience, MOH with this device is via a TFTP server. I read it somewhere amongst the scraps of notes they call a manual. You can download a TFTP server from Solar winds or from Cisco.com

First of all, how does the server act as a firewall if your IP's are on the same subnet, there is no way it can route like that. Did you mean one IP is 192.168.1.1 and the other is 192.168.2.1? I know what you mean about it IP hopping, I wish it did not do that. Remember you can assign multiple IP's to a single NIC.

That would be done in the network adapter settings in Device manager. Problem is if only one network card installed on server all packets will have the priority bit not just the VoIP ones. If no settings for that then you can look at installing two network cards in server, put one card on a voice vlan, set QOS on that whole vlan. GL

I believe CME 3.3 uses Skinny protocol not SIP. Newer versions of Callmanager 4.0 and up support SIP if I recall but not the express ver. You would have to run a H.323 gw or Plair.

I noticed quite a bit of lag but it from my Cisco ATA 186. It is such a dawg.

You need to read and read. I think to keep costs low they limit their support. Since support takes a long time is probably the reason for them limiting their access. I actually appreciate them keeping the cost low by doing this. Why not throw your questions out here to the group, explain what it is you need or trying to do and I think you will get some feedback. Also, when I get emails from them on my hotmail account I find it in the Junk Mail folder, be sure to check there and flag them as not junk mail.

Mine does the same thing, will not transfer. Using a voicemodem i am able to get it to hookflash and transfer my call to my cellphone which is pretty cool but no luck getting it to tranfer to a VOIP extension. I can call IVM from an IP phone on my network press an option and have it transfer me to another extension When I try an incoming call via my fxo it seems like it does a hookflash or something does not tranfer correctly. Found the Fix, in Sipura go to PSTN and use outbound proxy set it to the Axom server IP also set use outbound proxy and dialog both to yes. Works now.

On mine will not stream unless it is set to play using the sound card as well. Pull down menu, must have sound card selected or will not stream.

To do it legally you need to use royalty free music, or you can do like I did and get permission from a friend who is a band!

I have mine runing on SBS2003 with only one problem, when I try and get it to only stream without using the sound card (deselect the sound card) the audio stops. I have to let the music play though the sound card all the time. That doesn't make much sense since I only need it to stream over an internet port. But yes it plays, I keep loosing the voice messages if I keep playing with it and have to load them back in but once set it will stay.