Jump to content

Lyle Chritton

Members
  • Posts

    27
  • Joined

  • Last visited

Everything posted by Lyle Chritton

  1. I've installed Vodia 5.4.1a and confirmed it's up and running from System Status Overview Software-Version: 5.4.1a (CentOS64) Build Date: Jun 9 2016 08:59:12 License Status: Vodia PBX Hosted Please use the information shown on this web page when you request help from the support team. Software-Version: 5.4.1a (CentOS64). Then I installed the latest Vodia Phone from the Google store which was v1.2 dated April 27 2016. Then in the Pbx Url: field I put in "https://servername.domainname.com" and clicked the "Connect" button. Nothing. Anybody get the Vodia Phone to work ?
  2. I stand corrected, I missed the part about it being Hunt groups. Is the * character ignored in Agent Groups as well ?
  3. We have been trying a lot of softphones - Ekiga, Bria 3. Bria 4, Zoiper,. Most are not fully compatible or not bug-free with the Vodia PBX in some way or another. We have issues with BLF, echo, call quality, call presentation and codec issues, and even calls that don't hang up properly at the far end. I have to go into the PBX and manually kill each call (and even sometimes the PBX won't let me kill the call, and I have to reboot the PBX. The softphone technical support ... isn't up to snuff. Having the ability to hookflash star codes (ie work directly with the PBX) helps troubleshoot whether the issue is with the softphone or the PBX. The ability to work with the phone calls directly with the PBX using star codes, rather than through the phone's 'buttons' (which are already programmed and can only be changed by the softphone manufacturer) is crucial for troubleshooting and streamling call flow. So back to my original topic: What is the star code combo to perform an attended transfer ?
  4. We have been trying a lot - Ekiga, Bria 3. Bria 4, Zoiper,. Most are not fully compatible with the Vodia PBX in some ways. The biggest disadvantage is
  5. I haven't tried *, but it should work. Try it. If I remember correctly, extensions are rung in order, lowest number to highest number. We've always specified the extensions because sometimes we need to change the order in which the extensions are rung, especially with staging groups of 4.... My question would be, "Can you combine declaring the ext specifically and use * So "100 101 200 201 105 205 110 210 * " This *should* ring 100 101 200 and 201 in the first staging group, then 105 205 110 and 210 in the 2nd staging group, and finally ring all extensions in the 3rd staging group. Can anyone verify this ?
  6. All I am using is a softphone - no hookflash button, no physical phone to hang up and then pick up again. How would you do an attended transfer with * codes ? (We've established how to do a blind transfer with *77). I can see how a conference call applies towards an attended transfer - how would you do it ? I can see how to put aside a call with Call Park and Call Retrieve, I don't see how you would do a conference call. First off, what's the answer to: " We still need to know how does the "We have something similar for the establishment of a conference call" work ? " and how would that work with star codes ?
  7. First of all, as far as I know, there is no hookflash symbol code available for use with star codes. In fact, there is no real description anywhere for DTMF codes for (Wait,Pause, Hookflash etc) But if there were a symbol code for hookflash, say R, you could, during a call do some interesting things. But hookflash would only work if you're still in the same call that you activated the hookflash. What happens if you hookflash call A, so you're now in the dial tone for Call B ( with call A on hold) and you dial a number in call B and after talking to callee B, call B hangs up. What happens with call A - does it ring back to you ? And rather than having to create a park orbit for every extension, which just may be necessary as an alternative plan to make this work ... We still need to know how does the "We have something similar for the establishment of a conference call" work ?
  8. how exactly does the "We have something similar for the establishment of a conference call" work ? That needs more explaination.
  9. I've tried this with the computers and networks I regularly use and it happens each time, just not at the same time each time (sometimes it's 2:45 and sometimes it 3:15 and sometimes it' 3:45). I am double-checking to see if other users have this issue and if this issue is reproducible from other machines on other networks. Also getting a PCAP of the user logging into the server and they playing back the audio recording is a great idea, but instructions or a how to would be better. It's not like I can turn on PCAP recording for that sort of thing from the PBX server. Thanks
  10. The reason for raising this question is because in the web interface for playing back the audio recordings (ie the audio media panel at the top of the list of recordings): 1) I can't play more than 3 or so minutes of recordings before the recordings just stop, even though there are still 1,2, 5 minutes left of recording to listen to. 2) I can't jump from one part of the audio recording (say the beginning) to the middle or toward the end of the recording. I can only play the recording from the beginning the first time. 3) If I try and re-play the recording after the first time using the audio media panel at the top of the list of recordings, it won't replay. I have to go down to the specific recording and click on the play button to the right of the recording.
  11. Being able to call from a cell or land-line in to a phone number on the PBX, simply entering a username, and then a password, and then be able to review Agent Group recordings. I've had requests from supervisors and clients wanting this feature so they can check how things are going from the road (hands-free or speakerphone) or when they're not in the office. (Internally from an extension, it would be great to be able to dial in to an Agent Group alias., entering a username and password, and then review the recordings.) (Webwise - it would be nice to be able to use WebRTC to check Agent Group recordings with a username and password) I've seen there have been other people asking for the same/similar feature. Lyle Recordings not available from phone Started by Kurt Harnish, Dec 06 2013 07:38 AM How to get recordings out to customers Started by sudo, Dec 05 2013 10:45 AM
  12. Turning PCAP on will get you the why it's happening, but not the quick fix. :-) We had similar problems as well with slow dial-outs and sometimes the following will help, but we, too, are *still* having this issue <sigh> which is, as Vodia Support pointed out, related to those damned carriers: 1st off: If you're in the US, always dial 1+area code + phone number. It never hurts, and sometimes works.. really!. 2nd: This may be DNS-related, and our 50% fix was to : Set "outbound via Proxy" and specify the FQDN and port for the VoIP server, e.g.: sip.domain.com:5060 Set in your topology (if you have it): - If using STUN or ICE, select "Discover public IP address (STUN)" and specify the FQDN and port of the STUN server address, e.g.: stun.stunserver.com:3478 -- if not using STUN/ICE, then just select "None (use local IP address)". Finally in your Transport section (if you have it): - Explicitly specify UDP as the Transport, or possibly TCP, but only if the server supports it. If you do find a fix, please post it. Thanks Lyle
  13. i have lots of recordings that are 2-4 minutes long, and lots of times the recordings will stop playing around the 3 minute mark +/- 30 seconds, even though there is more recording to hear. I have a lot of recordings that are in the 4-9 minute range which I can't finish at all because they stop playing after the 3rd minute. Anybody have a fix for this issue ?
  14. Can you right click on the play recording panel and save the call recording ? I've tried to do this in Chrome and Opera and I can NOT save the recording locally as I was once able to do.
  15. i have lots of recordings that are 2-4 minutes long, and lots of times the recordings will stop playing around the 3 minute mark +/- 30 seconds, even though there is more recording to hear. I have a lot of recordings that are in the 4-9 minute range which I can't finish at all because they stop playing after the 3rd minute. Anybody have a fix for this issue ?
  16. In order to do a blind (unattended) transfer, there is the *77 star code. However there is no star code for an attended transfer! I can't figure out how to do an attended transfer with the current star codes. Why isn't there a star code for attended transfers?
  17. So when blocking or unblocking your caller ID, it's a 3 step process involving: dialing *67 and hanging up to block caller ID, then dialing 19995551212 to dial a phone number with Caller ID blocked. Then you have to dial *68 again to unblock your caller ID. Just being able to dial in one step (as with any other cell or landline phone ) the string *6719995551212 to do both steps in one shot. Same 3-step scenario above applies to *68 when in a caller-ID blocked phone setup. There are other combinations as well... *91 whitelist and *92 blacklist Both of these should allow the addition of phone numbers at the end of the * code, so that numbers can be added directly from the phone. *91phonenumber would add to the whitelist in 1 step *92phonenumber would add to the blacklist in 1 step lastly, in order to use the ACD outbound-calling ANI feature with the *64 code, you have to dial 3 times: dial *68 to unblock caller ID dialing *64ACD_alias - to log in to the single ACD group which sets the out-bound ANI dialing 19995551212 to reach your party using that particular ANI being able to just put the 1st three steps together would be nice. You'd need a separator character (#) to let the system know when the *64 star code ends, but it would be something like *68*64acd_alias#19995551212 One step to rule them all....
  18. What does "REST" mean ? As far as moving to HTML5 and Javascript (which I LOVE!!!!) what does REST mean ? (And I love wallboards as well !!!)
  19. The people who want to call in to review agent group recordings by cell/land-line phone won't care about any hardware requirements, or that they have 1000 messages to review (newest first, obivously, just like any voicemail system) - they just want it to work. They are required (in some cases by law) to review all calls by their staff, regardless of how many voicemails there are. So what if there is 500 voicemails in the system - there are 2 supervisors, 1 manager and 1 owner who will all call in and check voice mails. I just need it to happen. I understand what I'm wanting isn't useful to 50% of the people out there - but I am one of the 50% for whom this issue is critical and necessary, regardless of any hardware overhead. I understand the resources needed to record all calls, all the time, etc. The original specs (well beyond snomONE recommendations) on my snomONE server were 2 Xeon dual cores with 8G of ram - I have no idea what it's been upgraded to now (2 Xeon Quads with 16G?), but I've never seen my media CPU usage graph ever go above ... 5% ? It should be well able to handle v4 or v5 with any weird stuff. If the agent group recordings are time-limited (ie they get erased after so much time) or space-limited (after a certain hard drive space/memory-usage limit has been reached), that's fine. Lyle
  20. On the website under Extensions, it says: Automatic call recording. Calls to the extension can be automatically recorded. The recorded messages are made available in the mailbox. Does this mean that for v5, any/all/selected calls made to an Extension can be recorded, and then are made available in the voice mailbox automatically ? What about having this done for an Agent Group? I know many managers, supervisors, etc who would love to be able to have the same functionality for an Agent Group. People would buy/need the automatic recording license, snom v5 saves any/all/selected calls made to the Agent Group as voice mailbox recordings to an Extension. Then manager, supervisors, etc would be able to call in via cell/landline and review call recordings.
  21. I know that the current version ...132 only support snomONE v5.0.3 and up. Is there an older version I could use ?
  22. Great. Thanks. I don't see anything on the a la carte menu. How much is paid support ?
  23. Great to know - it's not on the website under contacts. Sales hours 9-5 M-F EST is perfect. Thanks.
  24. When can I talk to Sales staff ? what time zone are they in ? When can I talk to Support staff? what time zone are they in ? I need to find out what version and a la carte features I need to upgrade from snomONE Blue v4 to the new v5. There is no way I saw to add more CO lines via the a la carte menu, which is hopefully an oversight and will be added shortly. A lot of features still aren't working properly with v4 and I want to make sure that they are fixed or will be fixed ASAP in v5!
×
×
  • Create New...