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  1. Here is the issue we are running into: - We have a number that is meant to be specifically for faxes - but it gets voice calls on because it is listed in phone directories incorrectly Currently the fax number is programmed into a specific extension in the PBX (ie ext has account number 499 561-xxx-bbb) for the fax machine which is attached to the PBX using a cisco SPA112 gateway which works fine, the calls come in from the same sip provider (ie same trunk) as our main voice dedicated lines What we are trying to do is put an auto attendant on there so for people that call the line because of the bad listing they get a brief recording that briefly explains the listing error and asks them to press 1 to speak to a service rep (which then send them to the hunt group for the phones) and for the system to recognize faxes and send them to the ata instead of the hunt group Current call flow: - Voice Number is XXX-AAAA / Fax Number is XXX-BBBB - Call comes in on trunk is routed based on the incoming DID - for calls that are XXX-AAAA calls are routed to hunt group 300 - for calls that are XXX-BBBB calls are routed to fax ata 499 Proposed call flow: - Call comes in on trunk and is routed based on DID - for calls that are XXX-AAA calls are routed to hunt group 300 - for calls that are XXX-BBBB calls are routed to auto attendant 320 - Attendand plays recording offers to press 1 to speak to user - attendant menu structure: 1 = 300 / F = 499 We setup the proposed call flow and - if we call the fax number from say a cell phone or other land line we get the attendant if we press 1 we get the hunt group without issues - if we call into the system from a fax machine the blf light for for CO line lites up and starts blinking and the call log shows its hitting the attendant but the call is never routed to the ata. according to the sip router we have all the inbound calls routing through the call is being routed to our PBX and media type is t38 / PCMU8000, and the call just shows as ringing - if we disable t38 on the trunk at the provider and call in from a fax machine the attendant plays on the line for a second and as soon as a fax tone is heard it stops and routes the call to the ata, however because its not using t38 the faxes have a failure rate of about 8 out of 10. the provider we are using does t38 and they do detection on their end so the call coming into the PBX has the codec set to t38 already
  2. We are currently running Coma Berenicids Our extensions are in the 4xx range, our VM prefix is 3 In older versions (4.3) of the PBX we could setup BLF buttons on our phones that would have a value of say 3405 and if we were on a call and they wanted voicemail for extension 405 we could either hit transfer and hit the button that was set to 3405 or we could hit transfer, type 3405 and hit check, that worked without problems, we have also tried setting this as a "speeddial" button and it does similar to the blf option with this version (and i belive the version before it) with the same setup the system does this: = When using the transfer -> blf button approach, the system comes back after a few seconds on my end of the call and says "this feature is not available at this time", the call does however actually get the users voicemail box = When using the transfer -> 3405 Check approach, the phone (720) shows on the screen that the call is on hold, if you press the hold button it plays the message "this feature is not available at this time", the call does go to the voicemailbox = When using the transfer -> speeddial button approach, the system gives you a dialtone and if you hang up the handset or end speaker phone it comes back with the "this feature is not available" message, call does go to the voicemailbox The caller doesnt hear the "this feature is not available" audio they just get the normal voicemail prompt for the user Is there something were doing wrong setup wise or is this just going to be the way it works now?? we run a hosted pbx and we do have some customers using this type of functionality, right now they are running on the older version ( where this works without issue, but more of our customers are wanting 7xx series phones and we need to switch them over to the new server running the 4.5 versions for the 7xx phones to work. Basically the functionality we are looking for is: 1) I picked up a call that came into our office 2) Spoke with caller and found that they need to talk to PersonX (ext 405) 3) PersonX doesnt want to talk to them but wants them to leave VM, so we need to transfer caller DIRECTLY to PersonX's voicemail, so PersonX's phone isnt ringing for the next 30 seconds 4) We hit transfer button on phone, then press programmable button for PersonX VM and caller goes directly to PersonX's VM 5) hang up phone
  3. We have snom-one setup in key emulation with a bunch of 320 phones and an m9 cordless On the 320's we have the top 4 buttons programmed as shared lines on our trunk (co1 - co4) If we pickup a call on the m9 and put it on hold we can pick it back up from the m9 that put it on hold or from any of the 320's by hitting the appropriate line button However, if we put a call on hold from a 320 we cant pick it back up from the m9 Is there any sort of star code or magic account we can use to pick the call back up on the m9 that we can just program into the address book on the phone to allow for quick pickup of a line?
  4. Is this update compatible with a snom One Hosted license?? We have a hosting license and have been on 4.0 for quite a while and want to upgrade but havent heard anything on the hosted side of what used to be pbxnsip since snom took over....
  5. Hello, We are using the multi-tennat version of PBXnSIP ( (now snomone) We were having a number of users complain that they were too loud to the other party or were getting echos when calling PTSN numbers so we used the snom_320_custom.xml file and over-rode the hand-set mic gain on all of the phones attached to the system, that worked great and took care of the volume and echo issues we were having. Now we have a new customer using the system who is a very soft-spoken person (almost sounds like wispering even talking in-person) and we are getting complaints that people on the other end cant hear them, so we want to crank the gain up on her phone, but ONLY her phone. Because in the custom file we left the setting in RW mode we can log into the phone's web page and change it but it is lost as soon as the device reboots and we are back to square one... I cant for the life of me find a way to over-ride the custom file for a specific phone anywhere, with the loss of the old knowledge base and the wiki being next to useless and the search on the forums being about as good I cant find anything on how to do this. I tried manually editing the provisioning file in the generated folder for her extension but the software overwrote the file the next time a provision attempt was made. We dont want to upgrade to the newer snom-one software as of yet because of issues its having The phones are all using the HTTP provisioning method since the devices are not on the same network so any method of doing this needs to work with the HTTP provision system. So to summarize, what we are looking for is: - A way to specifically override a setting on just ONE phone using the http provision system - this setting is already being set by the snom_320_custom.xml file to a different value - the method must override the setting from the snom_320_custom.xml file
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