Jump to content

David McGowan

Members
  • Posts

    18
  • Joined

  • Last visited

Everything posted by David McGowan

  1. I did find this download link... http://pbxnsip.com/download/pbxctrl-2.1.14.2498.exe Is that the latest and final 2.1.x release I should try? Also, I can include the extension in the Conference Name, which is the email subject, but the Tel Alias is what shows up in the location, so for example I have a tel alias like the following... tel:+14041234567 The Location of the Meeting is +14041234567, what I want to know is it possible to include the extension somehow in the Meeting Location, so they get the full number and extension for the Meeting Location, if not I will just always put it in the Name/Subject so they can see it... Glad it's working now, tested a recording and it's also working... Did notice though in 2.1.14.2498 if I call the phone from another phone and get the system, dial my extension, and during name announcement type in Pin Code, it no longer goes to voice mail system like it use too, did this change or break in this release? david
  2. I tried to upgrade to the latest 2.1.x version so I can test again, but cannot find the download information... Looks like the latest was 2.1.14 is there a exe download for the quick upgrade for the latest 2.1.x release? david
  3. I understand the Tel Alias, I will add the number back and then put the extension in the Description, that will work... so I understand how to enter the extension and/or email addres, do not use the quotes... all the emails in the domain work, sending voicemail attachments, sending missed calls, etc... So emailing works just fine, the only place email seems to not work is scheduled conference emails... we are running 2.1.5.2357 I noticed there are some newer 2.1.x versions, these should all work with the exe replacing upgrade method, I can backup current exe and try the newest one... I already have the latest voice packs installed... david
  4. We have a working system and do not need any features in 3.x version, and generally did not want to go through upgrading and have potential new issues for no new features... We do plan to update at some point, but the conference should be working in our version from what I understand... I entered them in the following ways, tried many different ones... 708;709 "708;709" email@address.com;email2@address.com "email@address.com;email2@address.com" 708;709;email@address.com;email2@address.com "708;709;email@address.com;email2@address.com" So I tried just the extensions, which all have a working email address, tried them with and without quotes, also tried just email addresses, and tried both extensions and email addresses... I entered the time for the meeting in military time, example 22:00 for 10:00PM, set start and end time for the conference... Tested setting the meeting from 5 to 20 minutes in advance to make sure it had time to send the email... The second part is after you schedule a meeting you can pull up the Meeting file, the one Outlook receives once the email is working... It has the proper schedule time, and in the body of the appointment it has the conference Access Code... But in the description it just had the extension if I left the Tel Alias blank, and if I had the Tel Alias added it just had the Phone number... I wanted the appointment to have both the phone number and the extension to dial so the Appointment has all the information the people need to call the conference... Which logs do I need to look into, and should I enable any special logging before I schedule the conference to make sure it writes what we are looking for? Thanks, David
  5. I have having issues with Scheduled Conferences... I am using version 2.1.5, and we have an ad-hoc conference working without any problems... I can setup the scheduled conference, but when a user logins and schedules a conference, no email is ever sent... The email works for all users, and the domain... Every users gets emails for their voice mail messages, missed calls, etc... and it all works... Is there anything else that has to be setup so that the scheduleded conference can send out the email? Second issue, our conference is an extension, is there anyway to have the email, once working, send out the phone number and extension? I can add a Tel Alias for the main Trunk, then it includes this in the mail and not the extension, if I leave the Tel Alias off it sends the Extension and not the phone number, I would like to do both... Last question, main reason to want to use scheduled conference is the recording feature, is there any way to start a recording on an ad-hoc conference... thanks David McGowan
  6. So if you record in prompt 0 of the Queue and there are still agents available, it will still play prompt 0 and then the music on hold until an agent actually picks up the handset? david
  7. Is there a way to have the music on hold play until an agent picks up and not hear the ringing? We have only 4 support phones registered and the default calls all 4 as soon as the extension is hit... this makes the user never hear the music on hold and hear the ringing as they wait for the user. We have the timeout set to awhile so we hear the phone ringing and someone is not at the desks and we need to go back to answer them, there is not always someone there, but we need them to ring. david
  8. Is there a manual update available for 2.1.1 david
  9. Hopefully they will get a firmware out, thanks for the response... We use to use some softphones, that might have been where I saw it working... david
  10. not sure if this is a 2.1 issue, but do not believe we had the issue before... I have an Agent group setup to call 3 extensions, when a user picks up on one of the extensions, the other two phones get a missed call... Should it not detect that the Agent group got an answer and not show a missed call? I read about this on here somewhere but cannot find it and didn't think it happened before 2.1... They are Polycom IP 430 SIP phones... I read this should work due to RFC 3326 headers or something, is there a setting in pbxnsip? did anything change with this in 2.1? david
  11. We have not had 2115 give the issue with the calls where you are in the middle of a conversation and the other end of the call just all of sudden goes to voice mail, but it did happen once last week with a good clean connection... still can't get click to dial in the email to work with 2115, it takes no username/password... This worked in all other versions... all and all very happy with 2115 and it's working great, hopefully the call issue with the user going to voicemail during a call will not come back... david
  12. I was using 2115 today as well and had the same thing happen... I was chatting away (~5 minutes) Call cut off (I didn't press anything!) The Caller was then immediately transferred into my voicemail!? Also noticed with 2115 the TAPI works great, but the missed call emails with the Click To Dial link no longer takes any username/password I give... The same extension and admin username/password work fine from the web interface... david
  13. Is 2115 the version you have released as the official 2.1? david
  14. I am now using 2115 and I'm not sure if this is an issue with the newer versions or not... I have an Agent group that calls for example 3 extensions... All of these extensions are set to redirect to a cell phone after a certain amount of time, which works if the extensions are called directly... But when the extension is called from the Agent group the cell phone does not ring... david
  15. I see that, nice... So the above sip.cfg is fine because it will sort everything out for me? I see the default is: 0 8 18 2 3 Which means: "0" (G.711 u-law), "8" (G.711 a-law), "18" (G.729), "2" (G.726) and "3" (GSM) If G711 is working then that is the best quality because it uses no compression right? thanks for all the help! david
  16. Just upgraded to 2114 and it fixed my email issues from 2108... I have a question about codecs and my sip.cfg... Here is the section in my current sip.cfg file... <codecs> <preferences voice.codecPref.G711Mu="1" voice.codecPref.G711A="2" voice.codecPref.G729AB="3" voice.codecPref.IP_300.G711Mu="1" voice.codecPref.IP_300.G711A="2" voice.codecPref.IP_300.G729AB="3" voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2" voice.codecPref.IP_4000.G729AB=""/> <profiles voice.audioProfile.G711Mu.payloadSize="20" voice.audioProfile.G711Mu.jitterBufferMin="40" voice.audioProfile.G711Mu.jitterBufferShrink="500" voice.audioProfile.G711Mu.jitterBufferMax="160" voice.audioProfile.G711A.payloadSize="20" voice.audioProfile.G711A.jitterBufferMin="40" voice.audioProfile.G711A.jitterBufferShrink="500" voice.audioProfile.G711A.jitterBufferMax="160" voice.audioProfile.G729AB.payloadSize="20" voice.audioProfile.G729AB.jitterBufferMin="40" voice.audioProfile.G729AB.jitterBufferShrink="500" voice.audioProfile.G729AB.jitterBufferMax="160" voice.audioProfile.Lin16.payloadSize="10" voice.audioProfile.Lin16.jitterBufferMin="20" voice.audioProfile.Lin16.jitterBufferShrink="500" voice.audioProfile.Lin16.jitterBufferMax="100" voice.audioProfile.Lin16.frequency="16000" voice.audioProfile.Lin16.payloadType="117"/> </codecs> I have a Polycom IP430... I want to use the best codec possible for sound, I have no preferences set on the Trunk.. Are the preferences in the above sip.cfg supposed to match with pbxnsip? If so what should they be set? What is a good default for trunk codec preferences? Thanks.. david
  17. It works now... I rebooted both the PC and the Server and it works... I feel stupid... thanks! david
  18. I am running 2.1.0.2108 and everything seems to be working great... One of the reasons I upgraded was to try out TAPI... I have downloaded pbxtsp10.exe from the site and have it installed and properly setup in Control Panel... When I go to Dial in Outlook it just stays at Dialing and nothing happens... The Click To Dial with the link works, but I'm trying to get the Outlook dialing working... Is there anything else I should setup? Are there any other ports that should be open in the firewall? I left all the ports default... thanks! david
×
×
  • Create New...