Show Posts

This section allows you to view all posts made by this member. Note that you can only see posts made in areas you currently have access to.


Messages - zoo

Pages: [1] 2
1
Hope this documents will help..

BR
Zoran

2
Go to Terminals and check if any created terminals don't have assigned user - if found, delete terminal. This will release used license...If terminal is deleted from User screen - it's not really deleted - just not allocated to the user and will hold the license...

3
MiVoice 400 error list...https://1drv.ms/b/s!Aju8bIuuWmhWgYZybIDxWLLHWkljtg?e=4chScZ
Enjoy reading...

BR

5
Aastra - MiVoice Office 400 and MiVoice MX-One / Re: Room Charges
« on: June 03, 2019, 04:44:45 AM »
Hi bvtech
Haven't tried, but search for Virtual charges - basically, you created multiple outgoing routes with different pulse interval and then use LCR to route calls to different routes (according to destination number)...if using FIAS, you should be able to send final charges (in $) down to interface - or shown in own Hospitality manager...
Good Luck

BR

6
Try - terminal -> Multi lines -> select 3 (default 1), permission set -> busy on busy -> untick...
BR

7
PM sent

8
Aastra - MiVoice Office 400 and MiVoice MX-One / Re: Out of Service
« on: February 10, 2018, 10:23:28 PM »
Errr...yes

9
Aastra - MiVoice Office 400 and MiVoice MX-One / Re: Out of Service
« on: January 12, 2018, 04:14:03 PM »
Hi mate
First, check your license - how many Mitel SIP terminals are availble there? If 6, then go to Terminals and check if you have terminals created without users - if yes, delete those terminals...If unsure, post a screenshot here...
BR

10
Aastra - MiVoice Office 400 and MiVoice MX-One / Re: DSP/EIP MITEL
« on: December 20, 2017, 02:36:42 PM »
There is no simple answer to your question - again, will depends of your system config, network, SIP provider...
Eg. - you will using FXO - in order to communicate with IP (SIP) terminals, you will need Voice channels and DSP resource for transcoding. You will using SIP trunks - can you sucessfuly answer and transfer a call? If you have problem with transfer - then you need to route voice traffic thru the system - you need DSP. Can you make sucessful call and transfer between extensions on different nodes? If no - (sometime network config don't allow direct media routing between terminals) - you will need DSP...
In all my installation I have enough DSP's (or EIP) to satisfy all above - hard learned lesson on one of the first install...
But, it's your call...

BR

11
Hi fbp

1. No - 300Mb should be unlocked by applying Enterprise VM license - nothing to do with SWA (tried restart the system?)
2. Control outputs - never tried - but you will need FXO port -> change port type to Control output in    Configuration ➙ System ➙ Interfaces ➙ Analogue ➙ Analogue terminal interface (entry). Then, according to the document, you can open with *74 <Call number1)> or close with #74 <Call number1)>.
More info in document  syd-0570, page 474.

BR

12
Aastra - MiVoice Office 400 and MiVoice MX-One / Re: DSP/EIP MITEL
« on: December 19, 2017, 08:36:24 PM »
Hi Harrabi
Depends of quantity of those licenses (eg. number of voice channels, number of SIP access channels...,), your SIP provider config (do you need routing of media traffic thru system?), network settings Master - satellite (same as for SIP access channels),...
If you are configured A470 thru Mitel CPQ - number of required voice channels will be suggested in Result tab and necessary part (DSP or EIP) will be added to the config automatically...

BR

13
Aastra - MiVoice Office 400 and MiVoice MX-One / Re: SIP Trunk
« on: November 14, 2017, 01:56:46 PM »
Hi mate - yes, you need SIP trunk license for each trunks...depends of extensions used, maybe you need addititional DSP resources (and possible Voice channels license - again, depends of your system config)

BR

14
Another good program for testing...http://www.aggsoft.com/pbx-data-logger.htm
Its only demo, but good enough for testing...Also, upgrade 470 to the latest (or at least 4.1 HF2) to get enhanced SMDR report...
BR

15
This is, in my opinion, weakes point in 400...you have to use KT key function...if you have eg. 4 PSTN line, create 4 routes, 4 CDE and 4 KT keys. Only that way you can more-less see what is happening. But if any user use 0 (or 9) to dial out - you won't be able to see line activity (KT monitors CDE and with access code you bypassing CDE).
On 53xx work pretty stable but on 68xx expect problems with stability...
BR

Pages: [1] 2