TimB

I didn't say when another call comes on.......When someone sends a PAGE through PBXNSIP it puts the caller on hold and the PAGE comes through the speakers of all the SNOM phones. This is bad if you are on the phone with a client and suddenly without warning they get put on hold.

so paging through the phones is useless.

Version: 1.5.1.1.a All Snom 300's Version: snom300-SIP 6.5.6 If you are in conversation and some does a Page to all of the phones it puts the caller on hold and the page comes through the speaker. Is there any fix for this that would not send a page to those phones that are offhook and only send the page to the idle phones? Thanks

version is 1.5.1.1a

The system I have at Big Brothers is having a problem with the 2 remote extensions. They are connected over a VPN thats working very good but inter office calls from office A to office B here a static sound about 6-7 minutes into the call then it drops. Same if a remote extension at office B grabs an outside line from the office A and is talking with a client 6-7 minutes into the call it drops. Any thoughts???? Many Thanks

35 views and 0 suggestion. I love this product!

In mid conversation you get a Static Interuption like TV static sound for about 10 seconds then the audio comes back to normal. Any ideas whats causing this?

I have a client who's voicemail stops working all the time. You restart the pbxnsip service and it starts working again. Any idea what causing this and is there a solution, people are getting pissed off at me a regular basis these days. Thanks ((BUMP)))

Have a page group 700 setup and when dial it does an all call page to all the phones. If your on the phone it puts the caller on hold and you hear the page. Is there a way to disable that so if your on the phone you wont get the page and can continue talking. All Snom 300 phones. Thanks

Yes, it worked great as soon as the VPN was up and running just had to change the the IP scheme from to 192.169.2.X so they were not both 192.168.1.X One thing went right today, I'm very happy!

Boy you nailed this one, documentation for this product is weak and relying on this forum for real support is a joke. It's nice when people reply but there really is not much from the pbxnsip admin. The big advantage with this product over Asterisk was SLA when selling a key system replacement. But now Asterisk 1.4.5 supports SLA and is way more flexible than pbxnsip. Also when you pay for a 25 user license it should be 25 extensions, counting trunks and AA as licenses is wrong. Just my opinion!

Thanks for the feedback, both routers will be Linksys/Cisco RV042. What do you mean by the type of VPN?

We are setting a VPN between two offices, office B will have 2 extensions of of the pbxnsip server at office A Once the VPN is up and configured will it be plug and pray (play) for these 2 remote extensions or will I still to open router ports? Many Thanks.

What's happening is when I set it to Gateway Mode the trunk does not register with the ISTP, only registers in Sip Registration Mode. Makes no sense to me becouse when the trunk is registered under Sip registration Mode under the outbound dial plan I can create 6* and from and phone dial 6 and the number to the remote Snom phone and it rings through.

I created an dial plan called Kennebunk. in the trunk drop down menu I select VoIP2 and hit save but it keeps defaulting back to AudioCodes. Any reason it's doing this?