Detlef

The VPN is not setup on the PBX itself. Its done with two identical DLink VPN router. The 2 PBXs computer have those DLink routers set as default gateway. So whatever goes from our IP space 192.168.104.x to 192.168.108.x or vise versa is send to the router. Same with the Grandstream Gateway, that one has also the DLink router set as default gateway. So I assume whatever happens to the VPN status does not affect the PBX or the GWX4108 gateway - those should just forward everything what goes out of our local IP space to the router. Detlef

Normally our VPN is pretty stable but still, it goes to Mexico so we have days where it goes up and down alot. Internal calls from one PBXnSIP to the other works fine back and forth. Never had a problem with Audio. The problem comes when I try to call in with my cell phone over a Grandstream GWX4108 gateway. I get the system menu which asks me if I want to place an outbound call - which I select. Entering the 3 digit extension of the PBXnSIP system in Mexico rings the correct phone over there but if someone picks up there is no audio in either way. I dont believe it is the PBXnSIP, may something with redirection that the Grandstream doesnt like? The PBXnSIP internally over the VPN always works. Detlef

The *90xxx works on my system too. I was just trying the Unicast SIP paging account and could not get any audio to play. The Aastra phones only show the caller ID on the display. Also, is it possible to somehow enable the *90xx inbetween two PBXnSIP installations? I have one in the US and one in Mexico both have different extension number spaces and are connected via a SIP Gateway Trunk and a dial plan entry. Would that work if I add a dial plan entry that gets the *90 & EXT over to the other system? Detlef

I am impressed how easy it is to interconnect two PBXnSIP servers via two SIP Gateway trunks and a simple dialplan entry which forwards extensions 1xx to the first system and 2xx to the 2nd system. Everything seems to work inbetween those two systems. Only problem I have is when you dial in with your cell phone over a PSTN gateway (Grandstream GXW410x) and select the outbound call option to dial an extension in the 2nd system it only rings but no audio is transfered either way. Those systems are connected via a simple VPN, no port blocking or anything - both systems can reach each other completely freely. So I am stuck on how I would troubleshoot this problem and what do I need to look for?

Currently using the 2.1.xx pbxnsip it works on my Aastra 9133i showing the busy status on a trunk. I think it did with earlier versions as well. I just configured co1, co2 and so on in pbxnsip trunks and the button on the Aastra as BLF with "co1" and so on as value. Just a very basic thing - line busy equals light on. I havent checked if it does any other states. I only wanted to see how busy my PSTN gateway would be. I cant confirm that the phones hang.

Has anyone got paging working with the Aastra phones? I am using Aastra 9133i with Firmware Version: 1.4.1.1077, Firmware Release Code: SIP, Boot Version: 1.1.0.10 All those phones do is display that it is paged but don?t play any audio.

Just installed the .2111 and still have so far only two smaller issues: - Music on hold is not working with the pre-configured moh.wav file. It's silent to the caller if a call is on hold or parked. - The auto attendant does not announce "please enter the extension" after the customer recording ends. Those worked before with 2.0.3.1715. The .2111 release was a fresh clean install from 2.1.0.2093 and upgraded to .2111 Detlef

You registered the Grandstream as extension? I would check what codec it is using, routing and connectivity between pbx and Grandstream, maybe a firewall inbetween, STUN settings... I had something similar with audio in only one direction (not with a Grandstream) and it was because of the low bandwidth codec g.729 (18) which has to be purchased separately in pbxnsip. Detlef

I assume the pin was accepted correctly because it would ask for the number you would like to call. If I entered a wrong pin on purpose it would tell me that the pin was invalid and ask for a different extension and pin I would like to use. But I think I solved the problem by upgrading to version 2.1.0.2097 (Win32). The version where I was unable to make outbound calls was version 2.0.3.1715 (Win32). So I think that was just a bug in the older pbxctrl.exe version because with the new one it started working today. Thanks for your input...

I experience some problems with the DISA system and placing outbound calls. PBXnSIP correctly recognizes the user?s stored cell phone and answers incoming calls with the DISA menu. After selecting option #1 for placing outbound calls it asks for the pin number. After entering the pin number it asks for the number to be dialed followed by the # key. So far it works but after entering the wanted outbound number the user keeps getting either way a message saying ?the number could not be found? or ?you are not authorized?. My question would be if there is any system configuration option that needs to be enabled to allow outbound calls or is it misunderstood that if you call in with your cell phone and selecting option #1 on the DISA menu to be able to call any outbound number that would be possible with the default dial plan?

Ahh never mind, found the appearance tab under system setting...

I just noticed that the notification emails pbxnsip sends out contain all external links to all the images. Currently one of those can not be found with the external link: http://www.pbxnsip.com/img/main_left2.gif and is displayed as this missing image. I also have some users that have internal email but no internet web access. For those the email notification emails look funny. Is there a way to replace all of those external links into internal and put the files on the pbxnsip server?? Det

Ah, now I feel stupid! I figured out my problem with thos failing incomming calls. It was not the router, it was a multiple use of the same callcentric account. While I was experimenting with pbxnsip I used the same callcentric account that was assigned to a Linksys PAP2T analogue adapter. The line itself was disconnected but it was still connected to the network and registering with callcentric and as they state in their support it will cause trouble with incoming calls using multiple registrations. Nevermind wasting your time...

I read somewhere on the internet that the DLINK DI804HV has a NAT table of 10. That doesn't sound much to me and might relate to the problem. The other option I have is an old Cisco 1700 series router that I briefly configured as gateway but it didnt work at all yet. I think as next test I disconnect all other stations temporarily that use this router's NAT in order to see if that makes a difference. What makes me wonder is that I have two Linksys PAP2T phone adapter running over the same DLink DI804HV and they never fail to accept an incomming call. Its only pbxnsip that accepts calls for aprx 60sec and then quits answering. Thanks for your input!!

I am using a DLink DI-804HV, it lists the pbxnsip workstation on the same NAT port as it registered with callcentric.com in the log file on the pbxnsip. So unless the DLink is lying it should still be good. In the active session status screen it lists the pbxnsip session to call centric.com with 300sec as time-out, which gets renewed when it reaches 240sec while the NAT port stays the same. Playing around with this problem it seems more like a timeout issue than number of calls. If I am quick after a restart calling and hanging up I get more than 2 calls answered by the auto attendant. Seems to me it?s somewhere in a 60 second +/- range when it quits. If I delete the keep-alive value it goes to the default 30sec minimum refresh. If I put for example 3600sec in that field to prevent pbxnsip from reregistering then the DLink router lists it with 300sec time-out but I still have the same problem. No answer from the auto attendant. Going further and letting the 300 sec expire the pbxnsip disappears of the active session lists, reregistering creates a new active session with new NAT port and still not answer for incoming calls. Weird problem? Thanks for all your input!

I am using Version 2.0.3.1715 (Win32) on a Windows XP Prof workstation. So far I figured everything out, just one problem I am having is incoming calls. I have to restart the pbxnsip service, and then I can receive incoming calls from "callcentric.com" for about 1 minute. After that the auto attendant doesn?t pick up any more till I restart the service again. I don't see anything incoming in the pbxnsip log (set to report everything). Somehow something seems to time out and callcentric doesn't find its way in any more. If I check my NAT router the pbxnsip workstation is still active on the NAT port it registered on. I set the trunk (no STUN) with keep-alive every 60 sec, which is does (seen in the log). Re-registering the trunk per web interface does not help either. Is there a way to figure this problem out easily? Outgoing calls always work, even if the incoming calls quit working.

I like the feature with the missed call emails and remote call initiation. Just one problem - if it was an external call and I use on the PBX for example a "9" to access external lines then the call fails because in the email notification link is no external line access code added to the listed phonenumber. Is it uncommon to use an external line access code on pbxnsip?