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don't sweat it, all questions are good questions here ... in the newer 3.0 versions you can go to the System Level settings , go to settings -> PNP -> look for "snom_3xx_phone.xml/transport =" and change it from tls to udp ... if it is a less recent version , you have to dump the PNP and go with hand registration , yori

I run into this all the time , in my opinion there is a bug with SRTP and hunt groups, my guess if you have the same issue I run into all the time , the phones are encrypting SRTP , when this happens to me I switch from TCP to UDP either with PNP or manually and the problem goes away , hope this helps .. yori

I recently setup the programmable buttons on my snom phones with 7.1.35 and the most recent version of 3.0 .. I compared using the buttons programmed by hand through the GUI as extensions, PnP with Buttons set as "monitor extension" and PnP with Buttons set as "speed dial Discovery results determined that if you do them by hand you have full functionality . if you use the "buttons" feature , you can use 1 of 2 options and each option has 1/2 the functionality , Manual Program BUTTONS with Monitor BUTTONS with Speed dial Can Intercept another Ringing extension with BLF Button YES YES NO (if another phone is ringing and it being monitored , you can press the button and intercept the call avoiding the need for *87) Can Blind Transfer a call using Xfer + BLF Button YES NO YES (If you want to transfer you can press the transfer button then the BLF button rather than i.e. transfer + 207) Can call another extension by pressing the BLF button YES NO YES (basically allows the BLF to work as a mutlifunction button for monitoring and speed dial) Can call intercept a call on hold YES YES NO (if someone puts a call on hold , you can monitor that extension and see the BLF light slow blink) if you want you can simply press that person's line to take their call off hold and recieve it) I find this a problem since PBXNSIP advertises and actively encourages using Plug and Play for button assignments , but the results you get are absolutely not the same as programming the the buttons by hand , doing them by hand is clearly the more powerful but most labor intensive option , what can we do to get the "Buttons" feature with the PBX to better mimic the encumbent process, since customers switching to PnP and Buttons are clearly not getting the full functionality , y

I recently installed a system with Polycom 550's and want to take advantage of the 3 softkeys I have on those phones. I tried to think of a good way to emulate a key system since I was taking the customer away from an old Avaya system , what I cam up with was this ... I created 3 park orbits 801, 802 and 803 and add them to be monitored extension on all the extensions. This made them populate to the lines 2,3 and 4 respectively on the phones , items I was happy to discover , 1- I could make a quasi line 1, line2, line 3 etc for them in sort of a key system style , using co1 co2 etc wouldn't work since I could only monitor 3 lines and they had around 30 phones . 2- I could use the *85 and it would automatically read me the extension the call got parked to 3- the BLF lights indicated on all phones a call was parked on an orbit items I think we could improve to increase sales .. 1- I would like to be able to push the corresponding lit button to pickup the call rather than dial *86801 ... this would be a cleaner approach and would be nice since the customer's first instinct was to push the button to retrieve the call .. which leads me to problem 2 2- if you push the button it calls the extension you used for the park orbit , if you forget to turn off voicemail it asks you to leave a VM for extension 801 ... this seems an unprofessional and makes the feature seem as a work around ... so in a nutshell , can we create a raw extension type that for Park Orbits and can we make a way to allow those parked calls to be picked up by dialing them directly or using the buttons on the phone to retrieve them , it seems dialing extra digits is simple but customers coming away from key systems would be very happy to use this , especially considering the functionality is almost 80% there .. y

We also have systems worldwide and specialize in remote installs , I find if you partner with either a wiring/data guy or local PC tech , it goes quite well .. I would be interested in discussing with you , yori

Yes , I have done a bunch of these and they will do PnP fine , if not it is a configuration issue not the phone yori

as long as the pbx shows the registration for extension 100 in the accounts section , it will follow whatever dial plan you assign that extension , just * with no replacement will work , I have setup a bunch of these ... if it doesn't work with that , something is not configured correctly . here is a sample ini file ;************** ;** Ini File ** ;************** ;Board: MP-114 FXS ;Serial Number: 712095 ;Slot Number: 1 ;Software Version: 4.80A.014.006 ;Board IP Address: 10.0.0.250 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 10.0.0.1 ;Ram size: 32M Flash size: 8M ;Num DSPs: 1 Num DSP channels: 4 ;Profile: NONE ;------------------------------ [sYSTEM Params] SyslogServerIP = 10.1.1.89 [bSP Params] PCMLawSelect = 3 LocalOAMIPAddress = 10.0.0.250 RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 [ATM Params] [Analog Params] MinFlashHookTime = 100 FXSLoopCharacteristicsFilename = 'MP11x-02-1-FXS_16KHZ.dat' CallProgressTonesFilename = 'usa_tones_12.dat' [ControlProtocols Params] MGCPDTMFDetectionPoint = 0 [MGCP Params] [MEGACO Params] EP_Num_0 = 0 EP_Num_1 = 1 EP_Num_2 = 0 EP_Num_3 = 0 EP_Num_4 = 0 [sS7 Params] [Voice Engine Params] IdlePCMPattern = 85 VoiceVolume = 1 RFC2833PayloadType = 101 [WEB Params] LogoWidth = '339' [sIP Params] ENABLECALLERID = 1 MAXDIGITS = 11 REGISTRATIONTIME = 3600 ISPROXYUSED = 1 ISREGISTERNEEDED = 1 GWDEBUGLEVEL = 5 ENABLEEARLYMEDIA = 1 ISUSERPHONE = 0 PROXYNAME = '10.0.0.254' REGISTRARIP = '10.0.0.254' SIPGATEWAYNAME = '10.0.0.250' CNONCE = '0a123bcf' PASSWORD = '787899' ISFAXUSED = 1 REGISTRARNAME = '10.0.0.254' CODERNAME = g711Ulaw64k,20,0,0,0 PREFIX = *,10.0.0.254,*,0,255 TRUNKGROUP = 1-1,20,0 TRUNKGROUP = 2-2,777,0 TRUNKGROUP = 3-3,22,0 TRUNKGROUP = 4-4,21,0 PROXYIP = 10.0.0.254 AUTHENTICATION_0 = 20,#*0000 AUTHENTICATION_1 = 777,#*0000 AUTHENTICATION_2 = 22,#*0000 AUTHENTICATION_3 = 21,#*0000 TXDTMFOPTION = 4 [VXML Params] [iPsec Params] [Audio Staging Params] [PSTN-SDH Params]

Running on my Mac Book Pro with OS x 10.5.4 runs like a champ , finally !!! I am happy to see this add ...

I think you may be stuck then , with limited bandwidth and that kind of latency there isn't anything the pbx can do to speed up the invites and SIP messaging while the call sets up , other than sticking with UDP and avoiding encryption overheads ...y

Hey Carl , long time ! I usually check the registration tab in the pbx and see what is has registered , either the 192.168.x.x or say 74.12.234.x .... the SBC should take care of it unless the router is sending something it doesn't like .. then it's ugly stuff like STUN which may work but hate to use it long term .. yori \

I come across this alot , and you will always when going from the TDM world and proprietary products , since PBXnSIP is standards based it basically won't work , you would have to use a console and speed dial ALL users as *90+extension, which may satisfy a front desk person , but not if they want this on all phones like an avaya or comdial etc .. some systems like Allworx etc that are willing to stray from RFC standards and create their own phones have some ability for this , but it would be impossible for pbxnsip to modify the SIP package and then still be able to support the features on Linksys, SNom, Polycom, Aastra etc phones without issue .. and deal with all the bugs it can create cross platform .. so it becomes a sales game make the other features the you can offer with pbxnsip outweight the customers need to avoid dialing *90 to page another extension , If you make laminated cards to give the customer at install and to show them pre-sales how easy you can make this transition for them is your best bet , probably not the answer you want:) but it is basically what you're going to have to work with ..yori

the 1 in 10 calls having delays may be the provider causing more delay , alot of the 2nd and 3rd tier providers tap into Level3 and Global Crossing, your timing may be affected by their least cost routing or moving you to another switch due to high traffic on their end , I would try a 2nd provider as well and compare results , if this delay is strictly handset to hendset registered to the pbx, then that rules out my theory since there is no provider variable at that point .. yori

some of the white boxes came with a WAN port but the system didn't support it , the new black colored boxes are the only ones that actually can use the 2nd NIC

If you are using the WAN and the LAN port of the CS-410, make sure only 1 has a default gateway assigned , if both NIC's have a defasult gateway it tends to confuse the OS for routing in and out between the ITSP and the phones ,.. yori

That is the problem .... I can't update , it crashes the system using the GUI and I can't SSH in for some reason ,

if you are forwarding all calls to a cell phone , can we push the initial caller ID ? or just the trunk ID ? I think this was an old one we could not do , but hoping there was a way maybe now , they are using callcentric if that gives any leg up on the possibility of it ,

I have a newer 2 NIC black CS-410 ... running 3.0.0.2933 , it updates with the MSP update fine , but all others updates crash the system and has to be rebooted , also odd that the system will not respond to SSH requests either so I can't get in to do it by hand either ... working remotely and would like to avoid a service call that would end up being warranty work anyway .. I tried logging the SSH in the PBX but nothing shows up in the system log , not sure if the pbx can dig into the system far enough to get ssh requests into the log files any ideas ? could this be hardware failure .... obviously user error is entirely out of the question

I spoke to Polycom and they confirmed this is lacking feature on their part . They advised all interested parties submit a feature request on the Polycom Resource Center for Polycom Certified VARS , I am doing so and any others reading this please do as well if you have access to the PRC .. Yori

How can we setup the Polycom sidecars to to start monitoring users just on the sidecar and not starting on the phone itself , there must be a way in the config file to do this , it just would look cleaner if all the monitoring started and finished in one place ..

I have used these and didn't have to change any dial plan entries , the FXS would just time out and dial through the system on the FXO ports , send you .ini files if you like , yori

I guess if I post enough i'll answer my own questions eventually , with callcentric .. 9* and sip:\*82\1@\r;user=phone as the Replacement worked fine , y

sip:\*82\1@\r;user=phone this actually works with * and no replacement , but now I want to do 9* and get a 9 in the number everytime ,

I tried and get the same thing as Bill with Callcentric ,. it puts the *82 at the end of the number and not at the beginning .. y

their is a loose file system in that the files starting with attxxx.wav are the auto attendants, acd0_xxx.wav is the ACD greeting , acd1_xxx.wav are all the ACD first messages , name1.wav = the recorded name (option3) and personalx.wav is the greeting you have mentioned ... unfortunately, that's it and there's really no way to see what recording goes to what extension I have ever found , it may be buried in the XML tables somewhere ... I would like to see the ability to upload a .wav in the future for all extensions and accounts, just like we can for Auto Attendants .. that would make personal recorded messages more useful .. if we can upload a picture , why not the greetings ...

|^1([0-9]{10})$|sip:+1\1@\d;phone=yes|d I got this off the Snom site , looking to allow the Snom phones to dial after a 3 digit extension or an 11 digit OB call, using Identity 1 -> SIP -> Dial plan string , since it relates to pbxnsip and benefits pbxnsip users I figured I would post it here .. any help on this ?