jlumby

I noticed the fine tuning where you can have the mailbox act as a park orbit, however the devil is really in the details. You cannot transfer into the orbit without a star code dialing the orbit number will not pick up the call On the up side, the BLF does light when there is a call parked there. So it is not nearly as fully fine tuned, and polished as a dedicated park orbit could be

After other posts about the ability to use CO lines in a multi domain mode with a shared trunk not being successfully answered, I got to think bout what could be the simplest ways to pickup a call that is on hold (technically in my senario, it would be parked), that unlike buttons is completely sip compliant, and there would be no need to remember star codes. The simple thing that popped in my mind would be to create a different type of account that is actually a park orbit. As opposed to a standard extension, you could set it up so if a call was transfered to the park zone number, it would park it in that zone. If a call was parked there, it would light the BLF of any extension monitoring it, and if it was dialed, it would retrieve the call from the zone. This would solve other manufacturers problems since no one other than snom can do "buttons" The final outcome for the user is that any call that is parked there looks like, and can be picked up just like a call on an old key system, and it is fully SIP compliant.

Is there a way to use CO lines in a multi domain environment where all domains are shareing one trunk, however I do not want the CO lines to light up unless the call is actually in their own domain?

I would be happy to build a config file, however I do not have a phone to try to build it with.

Do you know what version you did the fix to? I am having a problem with .3160 and I could have sworn i had tested it on another system running .3177 and there was no issue.

I believe the issue is then that the Cisco is not subscribing. Is there a way to make it work along side a device that is subscribing?

I am doing it very successfully. I would say the number one problem is people trying to use junk routers that do not fully know how to statefully packet inspect/translate SIP as it does the NAT for it. I would highly suggest doing your NAT on a Cisco ASA5505 with Sip inspection turned on. I have had wonderfull success with this unit. It is fully featured with all of the DHCP, AND QoS features you would ever want.

I am trying to find a way to light a MWI on all phones that are registered as the same extension. I do not have any issues if all of the phones are of the same manufacturer (all Cisco), however if I have 1 snom, and 1 cisco registered as the same extension number, then the MWI will only light on the phone that registered first. I have the same issue between Cisco 7960 and an Audiocodes FXS gateway.

I had the same symptoms when someone decided to hookup a remote sip phone through a router that was not sip aware. It crushed the PBX and almost no bandwidth was used. I feel that we really do need some dynamic blacklisting for addresses that are causing a certain number of failures per minute.

I have a CS425 running 3.3.1.3177 and if I change the network parameters from the gui, and click on save, they are not saved across a reboot. Any help would be appreciated.

I have never used the nat setting on a xml based cisco phone. I have only used it on a 7960 and only when it was behind a crappy router that was not sip aware. Do you happen to have a copy of your config you could post? Is your phone connecting directly to the PBXnSIP (No Nat) Are you seeing it loose it's registration on PBXnSIP, on the phone, or both?

I am running this same phone with PBXnSIP without a problem. The only difference i see is you have some real old firmware. The oldest I have tried is 8.4.2 and I am currently running 8.4.4 I do have my registration timer set to 3600

Is there a way to set a permission on an extension so it trusts a specific IP address to make outbound calls? I noticed the trusted IP field but I am assuming that is for provisioning. I am interested in not requiring digest authentication for outbound calls from the specific extension if the IP address is trusted. Most ITSPs can do this, can PBXnSIP?

They look like a nice service, however I would like the PBX to work with the way my existing service provider wants to handle 911

I have tried, and it seems very intermittent. Several versions ago it stopped working alltogether, and I have pretty much given up on the address book for changing caller ID name.

Any plans to add the copy feature in upcoming versions? The time it is taking me is killing me as I am growing.

Does anyone know how this feature might apply to Nexvortex? They had never heard of it.

Then you cannot differentiate the domain is you are only sending the IP

Be very careful putting a FQDN in the proxy field. I have found it causes the phones to reboot at least every 24 hours, and they do not care if you are in the middle of a conversation or not. I have found if you just put the host name in the proxy field, and let the phone pull the rest of the domain name from the DHCP server, the phone will be stable, and you can successfully run it in multi domain mode.

This is nice, however how does it give them my E911 address since it is not sending the remote party ID that they are looking for, or does whatever I type in here override the normal remote party ID that I would be sending?

This problem has not disappeared in the multiple versions that have come out since. Any updates? I am currently at 3.3.0.3165 Win32

I am trying to find a way to send a different outbound caller ID only when someone dials 911. According to Nexvortex, the way that they do E911 when you have multiple E911 locations with them is based upon the ANI you send out. The problem is since they require a SIP registration, the only way I can change the ANI is to send it out a different trunk, however I only have one trunk. The other thing I had thought about is to have it go through another PBXnSIP machine, however I would prefer not to need an additional machine to manage, and have found that as you get multiple PBXnSIP registrations to eachother, performance tends to degrade quickly, so I want to avoid that at all costs. What is the best way to acomplish this?

I did the packet capture, and the PBX is only sending the notify message when a new voice mail comes in, or when one gets deleted, and not when one moves out of new and into saved status. I am running 3.3.0.3165 for Windows

Well, I am successfully faxing across it (as well as using a postage machine across it) (5 fax machines, and 1 postage machine). You just need to set it to disable echo cancellation when it detects a fax tone. As for Caller ID name, it is currently not supported, however I have been told by a Sangoma tech that it is in the works, and will probably be out in a few months. I have not tried it in a 64 bit environment. Does PBXnSIP even officially support 64 bit?

I am running 3.3.0.3165 for windows, and I had hosting customers complain that if they get one voicemail, listen to it, and keep it as a saved message, their MWI does not turn off. However if they delete the message, or get a second message, and delete that one, then the MWI turns off. Is there a way to set it so the MWI turns off when there are no longer any new messages. I verified that the web interface only shows old messages, and the MWI stays on until one of them is deleted.