Rob Lloyd

I have a couple clients with Polycom IP550 phones. The incoming calls go to a hunt group. When a phone is answered the BLF lights do not show up on other peoples phones. Outgoing calls this works just fine. Is this a problem that can be fixed easily? These clients desperatly wants to be able to do this. Even the PAC doesn't show the person on the phone. Thanks,

I'd like to request a way to restart the pbx service from the webpage. Sometimes I don't have full access to the OS, just the PBX Admin page. Having a way to restart the service from there would be helpful. Thanks, Rob

I was getting this today. I had the polycom firmware files in the tftp folder. The phone would download and install the firmware. Once it finished that it would restart and try to get the config file. Get the error message. I renamed the tftp folder so it wouldn't look at it anymore and the phone provisions fine now. Not sure what's going on but that was my bandaid fix.

No issues, I've got it running on several clients with SBS2003 and some new SBS2008 installs. No issues once you change port #s for HTTP and HTTPS. File, Print, Exchange all running fine.

You can stop the WWW service, install PBXnSIP, let the service start on port 80. Then go into the web console and change the ports to 81, 444 or whatever you like. Once you change the ports and save it, you can restart the pBX service. Then restart the WWW service and all will be well. I had to do that because the service didn't start and the install would stop and remove everything.

I have a client that has everything working almost perfectly. The biggest issue for them right now is transferring to VM. I tell them to dial 8ext#. When I do I get the vm message of the ext right away, then I hang up to get the caller to the VM and it continues where I heard it last. It should wait to start playing the message until the transfer is complete. I called and hear it on both sides - on the transfer side - "thanks for calling"... then I hang up... the caller then continues with the rest "I'm away from my desk...." This is with the latest version.

I'm using the new one. I'll check out the old one thanks for checking.

One the extention settings I put in 76 (the service flag #) and it doesn't show on the phone. If I use an * I see all the accounts, just not the flag. Might have to xml this one with phone config files?

I wish the buttons feature worked on Polycoms. Any idea if that will ever happen? I will try to monitor it and see what happens. I tried it on my system and it doesn't show in the PAC. I didn't have time to try it on the phone itself, but I'll do that now.

That's the idea but I'm not sure to pass the info back to the patton device to use the PSTN lines. If it was a seperate box, that would be easy. I need a way to tell the pbx to have it go to a different hunt group. Can you do sub-interfaces on the patton? Hunt group 1 goes to an ip of 4.2 and hunt group 2 has an ip of 4.3. Then I could create 2 trunks in pbx to seperate IPs. I have a support request in with Patton, but since they aren't familiar with PBXnSIP I was hoping someone here may have done something like this.

How do I setup a day/night service flag as a button on a 650 w/ sidecar? And on a 550. Ideally they would like it to light red on night mode and off in day mode. Thanks, Rob

I have a client with 1 voip trunk, (2) cable voip lines on a Patton SN4114 ports 1&2, (2) analog PSTN lines on ports 3&4. They only use the analog lines for incoming and DR of the voip trunk(s) - no outbound calls unless everything else goes down. Ideally they would like calls to go out the cable voip lines first, then if those are filled use the voip trunk. If that is down then go back to the patton and use the pstn lines. Is this possible using the same patton gateway? I currently have a hunt group setup to use just ports 1&2 for outbound. Incoming on all 4 ports works just fine. Thanks, Rob Lloyd

There is another PBX system that creates folders for each extension. Within each extension there are 2 folders - Voicemail and IVR. In the IVR folder there is greeting.wav or something like that. Just need to replace that wav file for that ext. and you're done. Very easy. Also easy to get to a users VM folder for backup or recovery. Or even to off-load it to a different location for archival purposes. Just a thought.

This is exactly what we are looking for. A service-flag that could play a greeting before the regular auto-attendant greeting. The above example is perfect - Good morning being played and set with service flag times. My client is looking to just play "hi today the office is closed due to snow and all classes are cancelled". Then play the standard greeting which they don't want to change since it has their company jingle in the background. He wants to call the system, record the new days event and turn it on. Sounds like a useful feature. Even "HI thanks for calling, ask your sales rep about month end specials".... or "Hi thanks for calling we are out of stock on that special game you are looking for, try back in a couple weeks". Sorry it's been a long day

What does the Conference timeout setting do? If they can program their devices to send keep alives that might work. Thanks, Rob

Client said that the calls cannot be dropped so they are increasing the call duration to infinite. Is there any way to have the calls detected and dropped?

Is there a way to have it play the standard greeting after the service flag greeting is played?

Where is this "hidden" setting? Sounds like there is nothing to worry about then. This will make for a happy client! Thanks!

I have a client using that has a unique application. They are using a push button/speaker/mic device to automatically connect to pbxnsip and connect to a conference room. That works fine. All the devices are in the same conference room and this lets everyone communicate like an intercom system. If the devices reboot there is no bye packet being sent. When they boot back up and automatically dial the conference room again the call status shows that the same extension now has 2 calls in there. Will this cause problems after a couple days or weeks or months? The extension is only showing 1 registration since it is from the same IP. They are just concerned with the extra, no longer connected calls. Will this eventually crash the system as the CPU starts going up and up from all these simulaneous calls? I don't think it matters but they are using the 410 and 425 appliances but it also does this on windows and linux. They have also tried 3CX and had the same results so it's not PBXnSIP specific. What can be done to prevent problems or is this a non-issue? Thanks.

Is there a way to setup a pre-AA greeting/message w/ a service flag before the normal AA greeting is played? Client wants to close the office for bad weather. He calls in (from an external extension, cell phone or regular phone), records the new message "Hi today is monday December 21st and we are closed for snow". Set a service flag from the phone to play the newly recorded greeting before the regular greeting. Right now there is a dummy extension setup - for weather press 4 and he can record that with the *98 codes. This works ok but not the ultimate goal. Thanks, Rob

Thanks! that was it. I didn't think renaming that would be a problem. Being a windows guy i was thinking windows domains.

I just checked incoming calls again. They re-route to my cell phone and voicemails work fine and get emailed correctly. Just no phone registrations. This is on an SBS2003 server so the domain has not changed - I'm registering to an IP and not a name anyway.

netstat shows 5060 being used by pbxctrl.exe so I'm not sure what is happening. I can port ping 5060 to the server IP and that works. xlite and my phones still show 404 not found and why would the pbxnsip log files show the same? What can't it find? I can receive calls so I know the service is working.

I had a working system yesterday. Today when I went to check it all my extensions are unregistered and I'm getting 404 not found errors. I can verify that port 5060 is open, my polycom phones are downloading the config files from the tftp directory. XLite is also showing 404. Here is the log file from a phone: [7] 2008/08/18 20:46:47: SIP Rx udp:192.168.2.53:5060: REGISTER sip:192.168.2.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.53;branch=z9hG4bK722d0ef046C1FE17 From: "500" <sip:500@192.168.2.9>;tag=EA77FDDB-92205626 To: <sip:500@192.168.2.9> CSeq: 1 REGISTER Call-ID: ea1787b2-dadb9cb1-d1523344@192.168.2.53 Contact: <sip:500@192.168.2.53>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.0.3.0401 Max-Forwards: 70 Expires: 3600 Content-Length: 0 [7] 2008/08/18 20:46:47: SIP Tx udp:192.168.2.53:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.2.53;branch=z9hG4bK722d0ef046C1FE17 From: "500" <sip:500@192.168.2.9>;tag=EA77FDDB-92205626 To: <sip:500@192.168.2.9>;tag=6d75a189fc Call-ID: ea1787b2-dadb9cb1-d1523344@192.168.2.53 CSeq: 1 REGISTER Content-Length: 0 Any idea what is going on? I have restarted the service and also the server with the same results. I was using the latest 2.0 version and swtiched to version 3. Replacing the 2.0 version back yields the same results. Thanks, Rob

Is there a way to start recording from the beginning of the call? Like a Tivo? Cisco has this ability and sounds like a nice feature to sell. I know this would mean that you have to record all calls temporarily. I am thinking my legal clients - they are in a call that was not supposed to be important but then the conversation turns to something that is. If you can record from the start there would be no issues of somone saying the call was out of context.