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Everything posted by brandywinetech.com

  1. In version Windows , I have setup alll incoming calls to a hunt group , I setup the default recording option to NOT record HUNBT group calls , I left the Hunt group to DEFAULT and on the users I have recording ON , problem is , Domain = YES record HG + HG = DEFAULT + user = NO then the domain overrides and it records the call anyway , the NO does not override the domain choice if Domain = NO record HG + HG = DEFAULT + user = YES, The opposite happens and it will not record the incoming calls regardless , yori
  2. I am using the Park and Pickup reminder and it works nicely , but the customers want to be able to tell between a new incoming call and a Pickup reminder without having to rely on caller ID on the phone , would it be possible to add another option for different ring type or something to indicate when park reminders are being issued , .. mainly looking for Snom or Polycom usage , Since park and pickup is getting more sophisticated a new topic may be in order , yori
  3. I figured it out , if you don't press "clear" after each file you upload , it over writes the last one ...
  4. I have a customer with CS-410 (Linux) If I go to main administrative section -> MOH , I can create a file other than moh.wav . say xmas.wav... it appears that you should be able to have 3 or 4 choices here , but instead , it only saves the last option when i go to localhost -> MOH dropdown I read the WIKI and it's not clear , is the intent that you get (1) additional choice for MOH per Domain , or should I be able to upload a few files i.e. Xmas , halloween etc for retail stores and let them change the music on hold to reflect the current season .. this seems to be the better option ... so in a nutshell , is this a feature that we could expend on and make it better , or did we already make it better and a bug in there is only allowing 1 additional file at a time , yk
  5. try using 11 insteasd of * ... i.e 1167 ... most of the analog carriers still respect the old rotary codes , the pbx still tries to process the star internally .. if it's a SIP trunk have fun , I generally let customers know up fron in my sales discovery that outoging *69 type codes can be troublesome and try to deal with it upfront , can't be all things to all people .. yori
  6. if I change the name of a trunk, my trunk goes 401 unauthorized and I have to re-enter the password , small item but it screwed me for an hour while I had to go find the password , seems like a bug , the trunk name shouldn't be tied to the password , y
  7. actually I think you are correct here and the old way was by extension and now it is concurrent calls , sorry if I posted some slightly incorrect information ,
  8. You are missing my Point , the REAL point here is that if I do the Snom buttons by hand EVERYTHING works fine , if I do plug and play for the "buttons" , they don't work exactly the same , basically doing PNP with the buttons is less feature rich as doing them by hand , but we are telling all customers how great PnP is for buttons etc and telling them to do it this way, and in essence they are NOT getting exactly the same results , PnP is good beacuse it is easier right now and has some usable features , but until it can support the full feature set as hand entered buttons , it is lacking in it's usability and in my opinion a "half job" that is out there lingering waiting to create customer complaints and kill customer references , bottom line is , we can provide the highest level of secure voice encryption , support wideband voice protocols , allow easy redundancy for hot swap servers and monitor calls on our desktops , but if the stupid buttons don't work , they are going to throw the thing out the window , yori
  9. The first thing they will tell you is upgrade to 2.2.2 on the Polycoms and try it , start there , I see this alot , I see alot of posts (and my own customers) experiencing something to do with muteed calls not sending RTP packets and then conference bridges will boot them alot , if you can upload an ethereal trace for support of an exapmle call being disco'd you will get better results , yori
  10. this eats up licenses, but probably the best way is to make extensions for all say 3 users, then forward all their calls to the accounting phone , then they would normally say call me at extension 202 , but they could have 210 as a dummy extension to allow them to be found in the DBN directory with calls forwarded to 202 , or register all 3 users to the same phone if it supports it , still eats licenses , or put in a feature request to have the ability to link more than 1 name field to an extension , if there is enough need for it in the market , it may happen , chances are , you will need to use a "work around" . .. yori
  11. are you using both NICS? and if so does ONLY the WAN port have a default gateway ? yori
  12. PBXNSIP supports t.38 pass through , which means you can support faxing to an analog ATA device via SIP trunks (if you are stubborn enough to do so) , there is no fax server built in to it though , so you cannot without a 2nd party application have DID's work for user calls and for their faxing , I know this does exist in some higher end enterprise systems as they have the software built in .. there is a company faxback.com that specializes in VoIP Faxing and fax servers , they are also a PBXNSIP user as well, and would be worth contacting , since they will have excellent insights as to fax server opportunities and how to integrate them with pbxnsip .. otherwise try Efax or callcentric.com now has a similar efax hosted product you could use in conjunction to give each user a DID for voice and a second DID for fax , regards, yori
  13. cost is based on the number of users and number of calls , there are 2 versions , 1 has a billing and reseller component built in for you and starts at about $15,000 to $20,000 Solutions11 , the other is basically the same PBXNSIP version as the commercial shipping product , but the licensing will accommodate more domains and better scaled pricing as you build up , I would guess if you want to start small your initial investment will be around 5K USD , obviously these are real rough numbers and we will have to look at your exact needs to price it out for you , your best bet is to contact the sales office and have them get a reseller in your area, if there is one , there are Italian prompts available for you , try contacting Paul Jamieson at PJ@pbxnsip.com and he'll get you to the appropriate reseller for you , if you have no luck PM me and i'll help you myself .. kind regards, yori
  14. http://provisioning.snom.com/update6to7/update_once.php I found this in snom_3xx_phone.xml , where can I change this ?
  15. I am running windows system 3..0.1.3017.exe snom 7.1.35 phones with Plug and play , after the phone boots , I set the U.S. date format (mm/dd): on off edit to ON , after every reboot it resets to OFF ,. I RESET the phone to factory 5 times , deleted the Generated and cleared the TFTP directory , I can't see what is doing this , yori
  16. same here , I had to go to every polycom_phone.xml file and change the "1" to "0" to disable missed calls tracking , is there a better way to do this ? the hunt group calls fill the display and on Polycom 320/330 phones it's like 10 keystrokes to clear the missed call list , since they are on a hunt group it constantly clutters the screen with the missed calls <missedCallTracking call.missedCallTracking.1.enabled="0" call.missedCallTracking.2.enabled="0" call.missedCallTracking.3.enabled="0" call.missedCallTracking.4.enabled="0" call.missedCallTracking.5.enabled="0" call.missedCallTracking.6.enabled="0"/> y
  17. I am adding <volume voice.volume.persist.handset="1" to all the <macaddress>.xml files in the generalted file for the Polycom phones , obviously alot of work , how can I do this for all phones without editing every file , shouldn't the there be a default file, like 000000000000.cfg file somewhere , or a master XML ? y
  18. I have found in a few 3.0.0.x and 3.0.1.x installs that hunt groups are displaying the following issues , Hunt group 600 stage 1 extensions 200 201 202 203 204 final destination 601 (auto Attend) set the ring duration to 0-29 seconds , works fine ,. set the ring duration for Stage 1 30+ seconds and it hangs up or I get callcentric number cannot be found , I am attaching 2 SIP traces , sorry could not get a Pcap , working remote on a CS410 yori bad.rtf good.rtf
  19. it would be nice to be able to have the option to upload professionally recorded voicemail greetings like we can with an AA , hunting and pecking through the file system to replace the .wav files is a pain for those who want to have them pre-recorded .. y
  20. it would be nice to be able to use a * wildcard for allowing access to a group mailbox , having to type in 15-30 users for the default company box takes a while ., yori and yes I did RTFM If the mailbox should be shared with other extensions, those extensions can be listed (separated by space) in the setting "Share the mailbox with the following extensions". If the mailbox does not require a PIN code, those extensions can directly dial into the mailbox and listen to messages. Those extensions will also receive the message waiting indication, if the sending has been activated and the phones register for MWI events.
  21. I see the system will send Vlan tags via plug and play , unfortunately comes a little too late in the setup as we really need to get DHCP to hand out the vlan tag to the phone, more of a snom questions but relating to installing pbxnsip , how can we get the snom phone to accept a vlan tag via DHCP even if we have to stray from RFC standards like I believe Polycom has..
  22. can you disable one of the CPU's in the bios , I know there were some old issues with dual CPU , maybe a shot , i'm around tomorrow , do you have remote access ? PM if you want .. yori
  23. You'd have to have alot of phones for a 100mbs network to bog down , I would include a packet trace to the thread to be looked at , I'm still not convinced there aren't issues with the HG coding .. sorry my advice made it worse , it's what generally does the trick for me ,
  24. let me know, i'd be curious since I have run into this alot , if you look at the status , you should see cpu spikes in the call history in the pbxnsip GUI if my hunch is correct , then they will go away if you switch to udp and the calls will not lose audio when the lines ring in , BTW what version are you running ?
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