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Darcy Primrose

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  1. Ok, I did that and I get the following <cell>6139876541</cell> <cell_acd /> <cell_conn /> <cell_dis>6139876541</cell_dis> Which one should I use. It appears that on some sytems it uses <cell> and others it uses <cell_dis> Please advise
  2. Using SOAP to add and modify extensions, we would like to input the cell number, but the paramter "cell" does not add the cell phone number. Also a description of what the fields need would be useful, ie, cell_mwi should it be yes or on or whatever. Thanks
  3. Currently using (Linux) We use the simple cdr format to send billing info to the call tracker. The string we request is as follows: <cdr_format>Domain{$m}Day{$b}Hour{$B}Date{$w}Direction{$o}type{$v}origTrunk{$x}destTrunk{$y}CLID{$c}FromExt{$F}ToExt{$T}Account{$R}Redirected{$r}ToHeader{$t}FromHeader{$f}Connected{$S}Answered{$C}Ringing{$A}CallId{$i}ivr{$W}hold{$E}</cdr_format> The "type{$v}" parameter is sending type "attendant" for a call that routes directly to an extension when the call is presented to the pbxnsip as a DID. The CDR is as follows:{voip.voipsub.com}Day{20081123}Hour{075324}Date{20081123075324}Direction{I }type{attendant}origTrunk{pstn}destTrunk{}CLID{anonymous}FromExt{}ToExt{61352933 0 4@voip.voipsub.com}Account{}Redirected{}ToHeader{"Test Phone" <sip:6135293304@voip.voipsub.com>}FromHeader{"Anonymous " <sip:anonymous@voip.voipsub.com:5060;user=phone>}Connected{1227444804}Answered{3}Ringing{1}CallId{6135293304.1227458822.116@M LCProxyServer-}ivr{}hold{} The type should be "extcall" or perhaps a new parameter "calltoext" since that is what it actually is versus an extension making a call. Here is the list of accounts in the domain 211;Snom 321;Extension;00041328704A 212;second 2;Extension;0004132C0165 c01;;CO-Line; c02;;CO-Line; c03;;CO-Line; c04;;CO-Line; 6135290000;Darcy One;Extension;000E08ECB460 6135290001;Marnie Two;Extension;000E08ECB460 6135293304;Test Phone;Extension;
  4. Yes I realize that, it then takes all the paramters from this file. But I loose the unique items per gateway that I require, such as the account, name etc that I get when the system uses the mac address to identify the spa2102 from the generated file. Is it possible to get the unique parameters from the generated file and the common from the html directory, I have not been able to do that, with with the spa 2102 or the snom phones.
  5. We currently are doing an off board solution which provisions all the parameters we require, however, it is off board and is not a clean process. Can the attached template be modified by integrators such as The VoIP Highway. If parameters are added to the PnP file, does this require an upgrade?? Since we are a reseller of pbxnsip, we like to customize each application to suit the user.
  6. While auto provisioning Linksys products, a file "spa_phone.txt" is generated in the Generated/{mac} directory. If we place this file in the html directory, the phone will receive this file but it does not get any of the parameters required for provisioning. How can this be accomplished with the linksys product or better yet, how can we affect change on the parameters generated in the "generated/{mac}/spa_phone.txt" file, that would be the ideal scenario. Specifically, we are looking at passwords and dial plans.
  7. we are running (Linux), which is the latest and greatest on the web site
  8. Using auto provisioning with the snom 320,360 and 370, we have good success. However, when we try to auto provision snom300 we do not get the buttons assigned, In the generated directory, the file "snom_300_fkeys.xml" has a length of '0', so naturally it won't work. Any suggestions why the system is not generating this file. We are seeing this on all systems we have or support.
  9. On standard key telephone systems, which we emulate with the pbxnsip, the DND key usually causes any blf assigned to an extension to be luminated when the dnd key is activated on the associated phone. This should probably be optional on a per domain basis to make it backwards compatible. We are getting several complaints on this from Telephone interconnects that are deploying this platform
  10. I notice in one of the latest releases they added creating accounts from CSV files. I cannot find any documentation on how this works, where do we find it.
  11. If a call is routed to voice mail and the voice mail is set to not receive any messages and the call goes to maximum duration, then disconnects, there is no cdr written. If the caller disconnects while listening to the message, a cdr is written. Might seem unimportant, but we have a customers looking for this and did complain about it. Part of our call tracking program.
  12. We use the simple cdr format to get the info for billing and call rating. PBXNSIP does not send the Request-URI so I cannot do what you suggest. If there is a way to get it, ie, sustitute the the to message with the request uri, that would do it. I do not have a concern with the people using the *00 codes seeing their destinations since this is a billing related issue and the only person to see it is the domain administrator unless they choose to pass it along.
  13. When you make an outbound call using speed dial (ie. *20) or redial ( ie. *66 ), the cdr fields you can collect the destination number from have the *xx code versus the actual destination. Makes it alwfully hard to bill the call. Immediately below is the cdr_format string we are using and three examples, you will notice the toHeader, which is the only place I can see to retrieve the destination number, has *20 and *66, I included one call which actually has the number I manualy dialed, 7451576. Any chance of getting a fix for this?? <cdr_format>Domain{$m}Day{$b}Hour{$B}Date{$w}Direction{$o}type{$v}origTrunk{$x}destTrunk{$y}CLID{$c}FromExt{$F}ToExt{$T}Account{$R}Redirected{$r}ToHeader{$t}FromHeader{$f}Connected{$S}Disconnected{$E}Answered{$C}</cdr_format>{voiphighwayoffice}Day{20080723}Hour{120304}Date{20080723120304}Direction{O}type{extcall}origTrunk{}destTrunk{6134233300}CLID{*20}FromExt{311@voiphighwayoffice}ToExt{}Account{}Redirected{}ToHeader{<sip:*20@voiphighwayoffice;user=phone>}FromHeader{"Darcy" <sip:311@voiphighwayoffice>;tag=fjokizv0ap}Connected{1216828984}Disconnected{1216828991}Answered{}{voiphighwayoffice}Day{20080723}Hour{115800}Date{20080723115800}Direction{O}type{extcall}origTrunk{}destTrunk{6134233300}CLID{*66}FromExt{311@voiphighwayoffice}ToExt{}Account{}Redirected{}ToHeader{<sip:*66@voiphighwayoffice;user=phone>}FromHeader{"Darcy" <sip:311@voiphighwayoffice>;tag=een157a894}Connected{1216828680}Disconnected{1216828692}Answered{1216828690}{voiphighwayoffice}Day{20080723}Hour{115937}Date{20080723115937}Direction{O}type{extcall}origTrunk{}destTrunk{6134233300}CLID{7451576}FromExt{311@voiphighwayoffice}ToExt{}Account{}Redirected{}ToHeader{<sip:7451576@voiphighwayoffice;user=phone>}FromHeader{"Darcy" <sip:311@voiphighwayoffice>;tag=z80h9gdbl9}Connected{1216828777}Disconnected{1216828793}Answered{1216828790}
  14. We installed a PROXY server between ourselves and various carriers and this completely solved the problem for us with the line parameter. You can contact Kevin Moroz re availability of this product.
  15. Hi John We run a hosted Voice over IP company in Canada and use the PBXNSIP product for our hosted server. We have coupled it with our own sip proxy, so we have the added bonus of being able to deploy cs410s and multiple PBXNSIP servers on a single network. We use a Linux platform and such as the other comments, we see the product as very stable. We use the SNOM phones exclusively and I must say the product is pretty impressive when used in conjunction with the Snom phones fully supporting plug and play and Key Telephone System features. We actually market our product as a virtual KSU.. We currently have about 50 domains on a single server and I am not seeing any slow down or performance issues. Be sure to take special care in Creating Trunk Groups, we have found that there can be an issue if you mistakenly add more Trunks than what you really need, pbxnsip is addressing this. We also have one domain we created on our server that we call "home.com" that provides a residential service for your business clients employees. A real nice touch. The PBXNSIP product works extremely well with the Linksys SPA2102 both POTS and Fax. Using the autoprovision feature of the SPA2102 it takes about 90 seconds to program a gateway, about the same time as a Snom320. I would not hesitate to recomment the PBXSNIP product as a hosted server. The feature set is equal to most of the large players. You can start small and grow it and you can give your clients some features they would normally only expect from the larger players. The single largest advantage I have found, you train yourself and your staff to manage the Hosted Server, than you can deploy the CS410 ip-pbx or the Officexx products and have the identical interface and feature sets, no retraining. It also lets you go to a lower level and thru xml programming, you can setup some very nice options such as IVR and Billing solutions. Do not be afraid to experiment, this product will do far more than they advertise it does. Darcy
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