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Hi, I'm having the following setup: 7 MxeIII's in a cluster, all version 6.0 SP3. In the system options, the number of hops is 10. 4 of them are expanded ones, with trunks (sip, tdm e1 and qsig) connected, so they have except the e2t also one or more dual e1/t1 interfaces. Let's call these...

Hi all, Short version: customer has 7 MxeIII expanded systems version 6.0SP3, about 20 E1 connections spread over all these Mxe's, 800 sip trunks (2 times 400 to a vMBG), 1400 pixel based (5320 and 5330) devices (200 full ip that we still need to add, 1200 device only for hotdesking) and...

thank you all. I used the summerize data and it worked fine. HAd to summerize 3 months and it took an hour/month :-)

Hi, I've a ccm version 5.8 running. When we generate a report from an earlier date than the actual day itself, all fields display 0. I already looked into the sql database, knowing that for older versions a database bigger then 4 Gb raised some problems... I looked on the Datadirectory and...

Hi, We use a 8961 with expansion module as a 'Receptionist' phone. If you receive a call, can you transfer this call immediately to an internal user configured on the blf/speed dial buttons? While on the line, we press the transfer button, receive dial tone, press the speed dial, but we don't...

I configured a few as a Simple SIP phone. In fisrt instance they were not working, but then we received firmware 02.02.3.5i and this works fine... What's not working? Regards, Gerry

Hi, I first created a custom cptone to recognize busy signal with the frequency and cadence, but in the end I did not use it. I simply called it ANADISC (from analog disconnect): voice class custom-cptone ANADISC dualtone disconnect frequency 425 cadence 500 When I attributed this...

Hi Guys, Short description of the setup: CUCM 8.6 with a 881 gateway. The FXO port on that gateway should be my local breakout... On the CUCM, a sip phone (Stentofon Intercom) is configured and making an external call, using his local route group which contains the analog line. I used the...

Hi guys, We recently update a few of our customers towards MCD 5.0. And we noticed a lot of 'endpoint' errors. When we contact Mitel, I always get the feeling they suggest we're the ones making the errors... So I just wonder if you guys faced some problems to after the upgrade. Here's what...

I can understand the reroute of the new device is not followed. I don't understand that it is routed back to a position that didn't answer before. Is there a way to get the external callers into the operators queue?

Thanks for the responses. Does this all mean there's no way to get callers who called the ddi numbers at a customer's site which are forwarded towards the operator console using the Call Rerouting alternatives will never get into the operators queue? And most important, how do we get them into...

Hi, Have the following setup: 1)Rad ports from the embedded voicemail into a huntgroup (type voice) 2)Call fwd 1ste alternative of the 5550 console towards that huntgroup Calls directly towards the operator have the desired behaviour: after the call forwards no answer timer of the COS of the...

Hi Irwin, It makes sense, in the way that you kind of agreed on the combination phone_ip/portnr towards the SIP provider. That's why you can hear ringback tone. Then, when my speech-RTP starts, the phone sudenly uses another port number without updating the provider, who still sends incoming...

Hi, We've a 3300 Cx (version 4.1) using sip-trunks for incoming/outgoing traffic. In some call scenario's, we have one way audio: on an outgoing call, the called party hears the calling party, but the calling party hears nothing. After hours of investigation, we found out that the initial RTP...

I once had the same problem. Because you're speaking about two main controllers, it's possible you have the same issues. Are the main controllers Mxe's? Do they have two network links (to 2 different switches)? In this case it is perfectly possible you forgot to enable spanning tree on the...

That's right. I contacted Mitel Technical support last week with the same question. In system speedcalls you can't enter a pause. It ignores the P's... Regards

Hello, Here's the scenario: A calls to B. B answers the call and they are in speech. C calls to A or B and pushes his 'conference' button. C is in the call from A to B. How can we switch this of? Regards.

You define the loss in dB. This is a value between -3 and 15. Giving in 15, means 15 decibels of loss. Giving in -3, means a loss of -3 ( -(- 3) = + 3 ) or a gain of 3 decibel. Good luck

Thanks a lot.

Yes, the call goes thru, meaning, I can hear the dtmf tones together with the ringback tone, but nothing is happening.