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Messages - saravanapandian

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1
MiVoice Office 250/Mitel 5000 / Re: SIP REFER
« on: February 18, 2019, 06:38:15 AM »
SIP REFER based transfer is requirement for our Carrier if PBX supports.Hope this is not supported hence i dont find any configuration

2
MiVoice Office 250/Mitel 5000 / SIP REFER
« on: February 11, 2019, 12:58:58 AM »
Do Mitel 250 supports SIP REFER transfer on SIP trunk?

3
MiVoice Office 250/Mitel 5000 / Re: SIP Trunk Fail Over Support
« on: January 23, 2019, 12:45:18 AM »

I have two trunk Groups under SIP peer.Trunk 1 name is 92002 and Trunk 2 name 92003.

I enabled registeration on both Trunk Groups

Please find the setting as part of registeration

Enable Registeration 5
Initial Retry Interval 50
Registeration Refresh Interval 300
Registeration IP Port 5100

Registerar IP will Diffrent for Both the Trunk Groups.I used MBG as SBC function since i used Route set IP is My LAN side IP (10.64.3.16)of SBC for Both the route sets.

Phones I configured
Outgoing Extension as 92002 and Emergency Extension 92003


I see 92002 is "TG not available" displayed on Phone as soon as dial 8 after putting 92002 as out of service, I assumed the call should route via 92003 but not happening



4
Hi Tech,

I dont have the explanation for the settin.what i observed is i dia 8*77 and wait for2 to 3 seconds then the call is been send out with Out 8.

Please find the below sample SIP request URI :

INVITE sip:*77@21x.20x.6x.xx:5060 SIP/2.0 

5
The issue has been resolved with the following setting. Now we can send start codes via SIP


set Spare Word Data 2 to value 4 (OLM view/System Information)

 8) :) :) :)

6
MiVoice Office 250/Mitel 5000 / SIP Trunk Fail Over Support
« on: January 14, 2019, 04:42:40 AM »
PBX: MiVoice Office 250 Version: 6.3 SP2

I have the Mitel 250 configured and two registered SIP trunk and both trunks are working fine.

Can someone confirm the SIP configuration option for Failover trunk when the First trunk does not respond due to some reason?

I want the call should be routed to  trunk2 when trunk1 failed to respond.

Please respond if this feasible and revert if need more details required.


7
Hi Tech,

Thanks for your respond, Unfortunetly I need to send * dialling in the initial request as per our service provider request.

Our SIP SP is being capable of activating some feature in there with respect to the star code, For example, the below scenario where the user doesn’t want to receive the call from Anonymous user from PSTN.so the anonymous call is being blocked in SP level. 

“PBX User1 dials *77 and hears msg "Your anonymous call rejection service has been activated successfully".”

The below are the Codes needs to be sent by PBX to activate a certain feature

These are all some sample needs to send in Request SIP URI


•   Anonymous Call Rejection Activation (*77)
•   Anonymous Call Rejection Deactivation (*87)
•   Call Forwarding Always Activation (*72)
•   Call Forwarding Always Deactivation (*73)
•   Call Forwarding Always To Voice Mail Activation (*21)
•   Call Forwarding Busy Activation (*90)
•   Call Forwarding Busy Deactivation (*91)
•   Call Forwarding Busy To Voice Mail Activation (*40)




8
 Service Provider Looking for to allowl * codes for example *77 is activating some feature in service Provider side but I see Phone not allowed me to dial any digits after * and displayed “call cannot be completed ”as soon as I dial 8(FAC) followed by*, The invite is being sent out without any digits or *  and call failed due to 404 not found from trunk. The trace attached. Looks like Mitel PBX strip * and not allowed any digits after * dial.


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