Mitel Forums - The Unofficial Source
Mitel Forums => Aastra - MiVoice Office 400 and MiVoice MX-One => Topic started by: gpais70 on November 14, 2017, 09:08:37 PM
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Hi. I am strugling finishing the last configurations on this model. I am lost to config income and outogoing calls. Please help!
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What type of trunks do you have?
Incoming calls are routed via CDE's
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Like sarond mentioned, you use the CDE.
When configuring, just follow the call flow as below. Use the graphical view under Routing to see what is not working the way it should.
Incoming call:
Network Interface -> Trunk Group -> DDI (if available) -> CDE -> Destination (user, UG, KT, ACD, etc)
Outgoing call:
User -> Route -> Trunk Group -> Network Interface
What exactly are you struggling with?
BR/Bojo
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Like sarond mentioned, you use the CDE.
When configuring, just follow the call flow as below. Use the graphical view under Routing to see what is not working the way it should.
Incoming call:
Network Interface -> Trunk Group -> DDI (if available) -> CDE -> Destination (user, UG, KT, ACD, etc)
Outgoing call:
User -> Route -> Trunk Group -> Network Interface
What exactly are you struggling with?
BR/Bojo
Hi Bojo
Thank you for you help.
Just resume:
The media resource only support G.711.
No DSP card.
The controller is already in function.
The users are all registered
Landline is analogic
I do call internally (users)
Receptionist(Astra 5380) is connected to controller.
Your unswer is that problem.
I am from Moçambique. The country,code is +258.
I just followed the graphical view but i stacked there.
If you send me the samples will succeed on this matter.
I hade a basic training on this model.
You help will very appreciat.
Email: gpais70@gmail.com
Kind regards
Thank you
The
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Like sarond mentioned, you use the CDE.
When configuring, just follow the call flow as below. Use the graphical view under Routing to see what is not working the way it should.
Incoming call:
Network Interface -> Trunk Group -> DDI (if available) -> CDE -> Destination (user, UG, KT, ACD, etc)
Outgoing call:
User -> Route -> Trunk Group -> Network Interface
What exactly are you struggling with?
BR/Bojo
Hi Bojo
Thank you for you help.
Just resume:
The media resource only support G.711.
No DSP card.
The controller is already in function.
The users are all registered
Landline is analogic
I do call internally (users)
Receptionist(Astra 5380) is connected to controller.
Your unswer is that problem.
I am from Moçambique. The country,code is +258.
I just followed the graphical view but i stacked there.
If you send me the samples will succeed on this matter.
I hade a basic training on this model.
You help will very appreciat.
Email: gpais70@gmail.com
Kind regards
Thank you
The
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Hi
i rechecked the configurations but the device still not make calls (wrong number) apear on the screen.
calling but not generate ring.
The digits are not working
Please help
Regards
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What type of trunks do you have?
Incoming calls are routed via CDE's
Yes
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In the Routing/Graphical View click on outgoing (top menu)
Right click user and click path, does the arrow point to the Route->Trunk Group->Network Interface
Also check that you're dialling the exchange access before the number. (you can find this in the number plan)
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Can you share what you have here?
Sample incoming routing:
(https://i.imgur.com/MfgGBhO.png)
Sample outgoing routing:
(https://i.imgur.com/7yJEcgL.png)
BR/Bojo
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Hi
The calls are coming in to console. But after 10 seconds without attending the call goes to another user rotating.
istill not call out, no permition.
Please assist.
Regards
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Please share your call routing screen shot.
Then user has a default route for external calls when your exchange access digit is dialled, is that working?
Does dialing the route call number (example: 170 in my screen shot) directly plus external number work??
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Hi
See attach.
Please correct using arrows
GP(https://uploads.tapatalk-cdn.com/20171122/bad00e826ab70a581e58c34db499e360.jpg)(https://uploads.tapatalk-cdn.com/20171122/03bc2b3a1484fa3d0028dc00dd729a84.jpg)
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Hi
Please correct with demonstration.
Regards(https://uploads.tapatalk-cdn.com/20171122/d923effad6d7f2622b0b8c8625df4945.jpg)(https://uploads.tapatalk-cdn.com/20171122/d5ed7047e5303509e73becb0fe3b744a.jpg)
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Please share your call routing screen shot.
Then user has a default route for external calls when your exchange access digit is dialled, is that working?
Does dialing the route call number (example: 170 in my screen shot) directly plus external number work??
Hi
Now is working, i corrected all as you mentioned and i successfully.
Just short the answering call by other user when the receptionist is not pick up?
How to direct it?
Regards
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You could add users to the delayed subgroup (in User groups)
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Thanks
Is the some process on mitel 3300?
Kind regards
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No, not the same as 3300.
In your user group you can have a delayed subgroup, those phone will ring after the first group.
(https://preview.ibb.co/gp0HYR/Mi_VO400_User_Group.png)
This is the easiest way, there are other ways as well but will require using CDE's.
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Fantastic
is possible to programe the MVO430 timing wich after work time is switched off till next day?
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You could set Time Controlled Functions
Configuration>Services>Time Controlled Functions
e.g
Day could have function *85011 Monday to Friday at 8am (This would activate Switch Group 1 Pos 1)
Night could be *85012 Mon-Fri at 5pm (This would activate Switch Group 1 Pos 2)
So the codes to change switch groups is *85<switch grp><pos>
Most likely you are using switch group 01 and you have 3 positions 1-3 which are programmed via the CDE.
Program your CDE for incoming calls to go to voicemail or similar with Switch Pos 2
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Please share your call routing screen shot.
Then user has a default route for external calls when your exchange access digit is dialled, is that working?
Does dialing the route call number (example: 170 in my screen shot) directly plus external number work??
Hi
Now is working, i corrected all as you mentioned and i successfully.
Just short the answering call by other user when the receptionist is not pick up?
How to direct it?
Regards
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Like sarond said, you could use delayed subgroup or CDE if no answer/busy in a scenario where the destination did not pick-up.
(https://i.imgur.com/N4xTWHS.png)
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Please share your call routing screen shot.
Then user has a default route for external calls when your exchange access digit is dialled, is that working?
Does dialing the route call number (example: 170 in my screen shot) directly plus external number work??
Hi
Now is working, i corrected all as you mentioned and i successfully.
Just short the answering call by other user when the receptionist is not pick up?
How to direct it?
Regards
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Like sarond said, you could use delayed subgroup or CDE if no answer/busy in a scenario where the destination did not pick-up.
(https://i.imgur.com/N4xTWHS.png)
Hi
I did but the phone wich i preprogrammed is ringing at some time with the receptionist also i need to know to give permition to user to make international calls.
Please assist.
Regards
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Check Permission Set and Digit Barring.
You need to configure your digit barring tables accordingly. Then under permission set, assign the barring tables you need. Assign the relevant permission set to users. For example, those who need to dial IDD need a permission set where the external digit barring table allows them to dial 0+country code. (The easiest way is to set all free in the digit barring).
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Check Permission Set and Digit Barring.
You need to configure your digit barring tables accordingly. Then under permission set, assign the barring tables you need. Assign the relevant permission set to users. For example, those who need to dial IDD need a permission set where the external digit barring table allows them to dial 0+country code. (The easiest way is to set all free in the digit barring).
Thank you.
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Check Permission Set and Digit Barring.
You need to configure your digit barring tables accordingly. Then under permission set, assign the barring tables you need. Assign the relevant permission set to users. For example, those who need to dial IDD need a permission set where the external digit barring table allows them to dial 0+country code. (The easiest way is to set all free in the digit barring).
Thank you.
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Hi
On mitel 3300 or 430 series how can transfer a call from land line to Cellphone?
Regards
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Check Permission Set and Digit Barring.
You need to configure your digit barring tables accordingly. Then under permission set, assign the barring tables you need. Assign the relevant permission set to users. For example, those who need to dial IDD need a permission set where the external digit barring table allows them to dial 0+country code. (The easiest way is to set all free in the digit barring).
Thank you.
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Hi
On mitel 3300 or 430 series how can transfer a call from land line to Cellphone?
Regards
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New licence
Hi
I renew the licence for MVO430 but on SWA State shows the information belllow: licence server is not reachable. Please check your internet connection.
Please help.
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So have you checked your internet connection?
Are your IP settings all okay? DNS?
Is your 430 on a network that has access to the internet?
If you telnet to your 430, can you do a dnsquery successfully? ping an internet URL or IP?
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So have you checked your internet connection?
Are your IP settings all okay? DNS?
Is your 430 on a network that has access to the internet?
If you telnet to your 430, can you do a dnsquery successfully? ping an internet URL or IP?
Hi
Thanks is sorted.
Now on the DSP resources the VoIP and FoIP are not available after load new licence. The receptionist cant transfer calls to the users.
Its sound like caused after new licence renewal.
Regars(https://uploads.tapatalk-cdn.com/20180223/559410d34d1c7ecbd1cf26e4ae4d37ba.jpg)
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Have you sorted the DSP?
Enable VoIP, re-assign resources (i.e. DECT) to VOIP.
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Have you sorted the DSP?
Enable VoIP, re-assign resources (i.e. DECT) to VOIP.
Hi
No DSP card available on this MVO.
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No on board DSP on the 430?
I can see you have resources allocated to DECT and Audio in your screen capture.
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430 should have 2 built in DSP's I believe.
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430 should have 2 built in DSP's I believe.
Hi
Please see the info bellow:
Regards(https://uploads.tapatalk-cdn.com/20180226/12beaa92e8b98ff4b52d126039d50b0f.jpg)(https://uploads.tapatalk-cdn.com/20180226/c90f0e9587b0ea43e42253236100b591.jpg)(https://uploads.tapatalk-cdn.com/20180226/fa43fc01fd9142b3e6c78d081a84b9eb.jpg)
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No on board DSP on the 430?
I can see you have resources allocated to DECT and Audio in your screen capture.
On the 430
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No on board DSP on the 430?
I can see you have resources allocated to DECT and Audio in your screen capture.
On the 430
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Hi
No DSP card on the 430. The licence in use as 729.G codec.
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No on board DSP on the 430?
I can see you have resources allocated to DECT and Audio in your screen capture.
On the 430
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Hi
No DSP card on the 430. The licence in use as 729.G codec.
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Aastra 5380 internal call (Circuit overload). How to sort?
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(https://img3.picload.org/image/dalgagla/lizen.jpg)
Sorry für German Docs
You need no Voip From DSI/FXS/BRI/Voicemail vice versa.
From IP(SIP)/ VOIP Phones 68xx/53xxIP to DSI/FXS/BRI/Voicemail you need a voip channel.
If you don´t use System Dect you can configure it as :
(https://img2.picload.org/image/dalglaai/dsp.jpg)
Use g.711 only so you can get more channels on onboard DSP. This setting does not affect the Voicemail codings.
you should add a dspx1 modul for more resources since the build in is quite limited.
greetings Jan
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(https://img3.picload.org/image/dalgagla/lizen.jpg)
Sorry für German Docs
You need no Voip From DSI/FXS/BRI/Voicemail vice versa.
From IP(SIP)/ VOIP Phones 68xx/53xxIP to DSI/FXS/BRI/Voicemail you need a voip channel.
If you don´t use System Dect you can configure it as :
(https://img2.picload.org/image/dalglaai/dsp.jpg)
Use g.711 only so you can get more channels on onboard DSP. This setting does not affect the Voicemail codings.
you should add a dspx1 modul for more resources since the build in is quite limited.
greetings Jan
Hi Jan
Thank you for the help.
See the the picture from my MVO after renewal of the license.
The scenario is:
Income/autogoing calls only the receptionist allowed.
Users only internaly calls
I am not sure if is my mistake doing configurations but was working before new license.
Did you have any demo of procedures for calls:
Germano(https://uploads.tapatalk-cdn.com/20180322/0fda5235547bc6bae09832104833ed8a.jpg)
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Click on down arrow for DECT, choose blank. Afterward, see if you can then assign any to VoIP. If you don't need GSM, do the same. Click on down arrow, change the value to blank and assign to VoIP. (you need to restart after changing DSP settings)
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Click on down arrow for DECT, choose blank. Afterward, see if you can then assign any to VoIP. If you don't need GSM, do the same. Click on down arrow, change the value to blank and assign to VoIP. (you need to restart after changing DSP settings)
Hi
I did the procedures it still not working.
Is this MVO can work without licence?
This device not have dsp module inside.
I don't understand why not working with new licence. It was "active".
See the picture.
Germano(https://uploads.tapatalk-cdn.com/20180326/6f62b058d4331fd76ff905509df1c4a7.jpg)
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That message means what you have licensed is more than what is configured and/or available. You still did not assign any channels to VoIP after removing them from the DECT and GSM. You added more to Audio instead. Please assign channels under VoIP.
(https://i.imgur.com/0fMhfW3.png)
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That message means what you have licensed is more than what is configured and/or available. You still did not assign any channels to VoIP after removing them from the DECT and GSM. You added more to Audio instead. Please assign channels under VoIP.
(https://i.imgur.com/0fMhfW3.png)
Hi
Please see the picture bellow.
I assigned to VoIp already. Now the the calls not coming in or outgoing.
I am not sure is something have to change in routing?
Regards
(https://uploads.tapatalk-cdn.com/20180327/a72ebd5884078617837f4d4093771d0c.jpg)
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That message means what you have licensed is more than what is configured and/or available. You still did not assign any channels to VoIP after removing them from the DECT and GSM. You added more to Audio instead. Please assign channels under VoIP.
(https://i.imgur.com/0fMhfW3.png)
Hi
Please see the picture bellow.
I assigned to VoIp already. Now the the calls not coming in or outgoing.
I am not sure is something have to change in routing?
Regards
(https://uploads.tapatalk-cdn.com/20180327/a72ebd5884078617837f4d4093771d0c.jpg)
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Hi Bojo
I would like to reset all configurations on this device than upgrade to new version. Is correct idea? consequences? The version available is 4.
Let me know the procedures-for it.
Regards
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You do not need to reset the configuration to upgrade. (Note that you need a valid license number or license file depending on the version you are upgrading to.)
There is no issue doing a reset or first start... provided, you are trained and know what to do. If you plan to configure the system from scratch again, I do not think it is practical to get the guidance you need purely via the forum if you do not have the foundational knowledge of the product yet. The forum can help, but should not be your only source of instruction. This is just my personal opinion.
For your issues, you can check the event log under system events... what alarms are there? You can also use the built-in system traces to further troubleshoot what is happening.
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You do not need to reset the configuration to upgrade. (Note that you need a valid license number or license file depending on the version you are upgrading to.)
There is no issue doing a reset or first start... provided, you are trained and know what to do. If you plan to configure the system from scratch again, I do not think it is practical to get the guidance you need purely via the forum if you do not have the foundational knowledge of the product yet. The forum can help, but should not be your only source of instruction. This is just my personal opinion.
For your issues, you can check the event log under system events... what alarms are there? You can also use the built-in system traces to further troubleshoot what is happening.
Hi Bojo
Please the messages and revert.(https://uploads.tapatalk-cdn.com/20180331/f1cdf912bcb289adf2b31db59a1467b8.jpg)
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What is the existing problem you are having?
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What is the existing problem you are having?
Hi
Calls problems, no calls in or out.
The routing.
I not sure if is correct, but before the new licence it was working.
The line is connected in port 1.1
(https://uploads.tapatalk-cdn.com/20180403/8492f74f976ab7fd02ba478c4d0ff3f2.jpg)
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Not sure about the routing but your line seems to be down. Your sync state is "hold". Is the line working? By default, any incoming call will reach user group 16 unless that routing was removed.
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Not sure about the routing but your line seems to be down. Your sync state is "hold". Is the line working? By default, any incoming call will reach user group 16 unless that routing was removed.
Hi
See the picture below the state changes dinamicaly to hold after a lapse of time.
On you route and call distribution wich is sequence as cde, swt po1, etc?
Regards(https://uploads.tapatalk-cdn.com/20180404/b48a0bf265d9c526b5a174923ad2d50c.jpg)
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Could you share a screenshot of your incoming and outgoing routing with the arrows to and from your trunk? You could black out any confidential numbers.
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Could you share a screenshot of your incoming and outgoing routing with the arrows to and from your trunk? You could black out any confidential numbers.
Hi
Please see the pictures.
Let me know step by step what to do on route.
Regards(https://uploads.tapatalk-cdn.com/20180404/d600497049ffdc511ab98368c51ce748.jpg)(https://uploads.tapatalk-cdn.com/20180404/c6c927e25c5457edd4f1f376afadd387.jpg)
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Could you share a screenshot of your incoming and outgoing routing with the arrows to and from your trunk? You could black out any confidential numbers.
Hi
Please see the pictures.
Let me know step by step what to do on route.
Regards(https://uploads.tapatalk-cdn.com/20180404/d600497049ffdc511ab98368c51ce748.jpg)(https://uploads.tapatalk-cdn.com/20180404/c6c927e25c5457edd4f1f376afadd387.jpg)
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New config.
Please check outogoing calls and what is wrong.
Regards(https://uploads.tapatalk-cdn.com/20180405/11f74d6e4cfb942c67316a9e541457b0.jpg)(https://uploads.tapatalk-cdn.com/20180405/c325b14bce948ba0bd62a9b15b9c055a.jpg)
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1. As I understand, your exchange line is physically connected to port 1.1
2. I assume your default outgoing route for users is route 1 based on your photo.
Your latest photo shows there is no allocated trunk group for route 1. You should allocate trunk group 1 (ADMIN) which are your BRI ports.
When your users dial the exchange access, it will go via route 1--> trunk group 1--> 1.2 then 1.1.
If you are not using 1.2, double-click on it under Network Interfaces and tick outgoing barred... so it is not included in the hunt for an outside line.
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1. As I understand, your exchange line is physically connected to port 1.1
2. I assume your default outgoing route for users is route 1 based on your photo.
Your latest photo shows there is no allocated trunk group for route 1. You should allocate trunk group 1 (ADMIN) which are your BRI ports.
When your users dial the exchange access, it will go via route 1--> trunk group 1--> 1.2 then 1.1.
If you are not using 1.2, double-click on it under Network Interfaces and tick outgoing barred... so it is not included in the hunt for an outside line.
Hi Bojo
Now i can call out but the sinchronization still on hold.
See the pictures below.
Regards(https://uploads.tapatalk-cdn.com/20180410/653bea751e77c164d3cde3cecf73c458.jpg)(https://uploads.tapatalk-cdn.com/20180410/1e34aaf83dcecadb1884c4b6fd55632a.jpg)(https://uploads.tapatalk-cdn.com/20180410/ca63c3dd8c260a9785a4a05921bfd1f7.jpg)
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Speed Dialing Aastra 430
Hi
Would like to know how can proceed with programing speed dialing only for internally calls on phones 6863i 58xx.
Our extensions starts with 1000-1007.
Best regards
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