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.


Topics - Tech Electronics

Pages: [1] 2 3 ... 7
1
SIP On Mitel / Mitel's new Analog Gateway is a Dinstar
« on: December 22, 2023, 10:45:31 AM »
Everyone,

I was just going taking the new MiVB 10.1 training and they were referring to their new AG4124 Analog Gateway.

https://www.mitel.com/document-center/devices-and-accessories/networking-equipment/ag4100-analog-gateway/ag4124/all-releases/en/ag4124-analog-gateway-administration-guide

Having seen this interface before I immediately remembered the Dinstart DAG2000-24S being similar.

After I looked at it more closely, they are exactly the same.

https://www.dinstar.com/analog-voip-gateway/24-fxs/

Thanks,

TE


2
We are trying to use a Dell R350 as an ISS for an MBG.

Since Dell requires the use UEFI we used RUFUS to create a bootable USB of the ISO.

We can get the MSL installation software to load but have issues with it finding media to install on, as in it can't find the internal HD.

I have tested the Hard Drive and it is good, but just to make sure we replaced it with a known good one and still have the same problem.

Has anyone else run into this issue and been able to solve it?

I am thinking about just using the free version of VMWare ESXi Hypervisor and turning the ISS into a Virtual Host Machine.

Thanks,

TE

3
To those who may know,

I have been working with the 3300 for a bit recently and I have never been able to figure out how to change the first key on the phones without going into each phone individually.

The second key can be done quickly by just exporting and importing the Multiline Set Keys form, but it won't allow you to change the first key.

Since the 69xx phones came out with their My Phone as the first key label we have been directed to change it to be the extension instead; kind of how the 53xx phones did by default.

Does anyone know of a way to change the first key's label to be the extension without going into each extension individually?

Thanks,

TE

P.S. Sometimes I feel like a mental midget on crack who can't figure out which peg is round, and which is square; don't get me started on the star shaped one.

4
This topic has been moved to Aastra - MiVoice Office 400 and MiVoice MX-One as the MiVO-250 doesn't support AD integration.

https://mitelforums.com/forum/index.php?topic=14782.0

5
Non-Mitel Chatter / OpenSIPs | Kamailio (OpenSER) Projects
« on: July 11, 2022, 08:31:02 PM »
Community,

Has any worked on or installed either an OpenSIPs or Kamailio Proxy server before?

If so do you have any insights as to how to setup and configure one for Class 5 trunking services?

Out of curiosity I will setup a poll to see how many people even know about these two SIP Proxies.

Thanks,

TE

6
Mitel MiCollab Installers,

I am having an issue where Accept Message is not showing up in Managed Statuses even though the users have the Chat Feature enabled.

I see there is a new User Profile added to the MiCollab Client tab under each user, but that only allows me to add Organization Managed statuses.

The other issue I see is that Chat is disabled on all users even with the Chat Feature enabled.

Thanks,

TE


7
Mitel Techs,

Alright, I am definitely at a loss here with this one. Whenever a phone is forwarded to an Outside Number and a call is transferred from the NuPoint is shows up the Caller-ID as Restricted or Private Call.

I have traced it down and figured out why it is doing it, but I am not sure how to make this change. The Contact Header on calls that are Transferred from the NuPoint always show up as Anonymous; see below.

Quote
Contact: "N.Port 019"<sip:anonymous;tgrp=Socket;trunk-context=tXXXXXXXXXXXXa.voip.sockettelecom.com@XXX.XXX.XXX.XXX:5060;transport=UDP>

If I call the users DID or get Transferred from another user the number sent out correctly as the Contact header is no longer set to Anonymous. It seems in all other cases the call goes out with the correct Caller-ID, but with NuPoint Transfers it doesn't.

I have setup the extension ports of the NuPoint to be associated with CPN Substitution, but that still doesn't fix the problem.

Does anyone know how to get the MiVB not to send NuPoint Transferred calls as Anonymous?

Thanks,

TE

Quote
Solution:

In the Class of Service on the MiVoice Business PBX for the NuPoint Ports set the following flag to Yes.

Display Held Call ID on Transfer: Yes

8
Mitel MiVoice Business/MCD/3300 / SIP Forwarding DTMF Issue
« on: May 26, 2021, 03:08:53 PM »
Guys,

We have an issue where a call comes in to the System and then gets sent back out to an 8xx number. When the callers get the IVR at the 8xx number it is not recognizing DTMF, but it was at the NuPoint.

Working with the SIP Provider and getting packet captures, we don't have an MBG, they are saying that the second call, Forwarded, going out to the 8xx number is not putting any SDP in the message and therefor they are sending it out as an In-Band DTMF call instead of Out-Of-Band.

Does anyone know how to solve this problem as I already have Send SDP in the Initial INVITE set to yes as that was a requirement originally?

Thanks,

TE

9
Mitel Techs,

Has anyone successfully setup a SIP Peer Profile with a Metaswitch/Metasphere, and if so can I get a copy of the Export?

The SIP Provider states they are using a Metaswitch CFS, but there isn't a SIP CoE document for that product that I can find.

Quote from: SIP Trunk Provider
Our platform is a Metaswitch CFS platform, but should be standards compliant.  No authentication, standard SIP Trunk.

Thanks,

TE

10
Mitel Software Applications / OAISys Error - No Recording Port
« on: March 26, 2021, 10:36:53 AM »
OAISys Guys,

We are having an issue where we changed the Public IP Address of the MBGs that are also setup as the SRC for the OAISys. Since that change there has been issues with calls not being recorded as it shows, "No Recording Port" next to the call.

We verified there were only 3 out of 50 calls being used and the next call will show "No Recording Port". The MBG has 110 SRC Ports so the issue isn't with ports on that side either.

We rebooted the MBGs and the OAISys again and that did not fix the issue.

Does anyone have any experience with this type of issue?

Thanks,

TE

11
Mitel "Bandwidth" Gurus,

I am trying to setup Bandwidth as the SIP Trunk Carrier and they are requiring a 911 Test that requires the MCD to Failover to the secondary 911 route when they send a 4xx, 5xx or 6xx response to an Emergency Call.

Here is what they are requiring us to do for the Failover Testing.

Quote from: Bandwidth 911 Testing
Simulated failover testing
This test will validate your system’s ability to successfully redirect failed calls to your secondary route.

During this test we'll simulate a scenario in which your 911 call is sent to your primary route and will then fail with a SIP 410 response. When you receive a 410 response, you should send the SIP INVITE to the opposite data center.

To complete testing, simply make a call from the following ANIs:

If ATL is the primary, 15555558888
If DFW is the primary, 15555559999
Once you’re setup to do these things, you’re ready to migrate traffic. Please notify us if you require any assistance with the migration of services. 

Note: If the 933 call isn't successful, please halt testing and open a ticket with your Bandwidth Support Team.

We have two SIP Peer Profiles for 911 that go to two different geographical locations [IPs] that Bandwidth supports. Their on-boarding team is requesting the previous testing to be completed, but I don't see a way to make that happen in the MCD. We do have an MBG setup and I am looking into whether or not we can Identify 4xx, 5xx, 6xx responses and redirect using the SIP Adaption feature, but so far I haven't successfully got that to work either.

I performed a search on the forums and seen that people have used Bandwidth Trunking before so hopefully someone can explain how they approached this scenario.

Thanks,

TE


12
Mitel MiVoice Business/MCD/3300 / One-Way Audio on MBG Failover Testing
« on: December 18, 2020, 08:28:43 AM »
Guys,

I have a problem that just isn't working out in my head so I thought why not put it on here to find out how stupid I really am.

Anyway, I have a virtual MCD at a customer site that talks directly to a SIP Providers SBC for SIP Trunks; everything works as expected.

On this particular site they have a MiVCR and a virtual MBG with TAP Licenses to record the calls of the ACD Group. These ACD phones are pointed to the Public IP Address of the Primary vMBG.

We added a secondary virtual MCD and virtual MBG for Failover capabilities on each of the respective networks. The Primary and Secondary MCDs are on the same network and the Primary and Secondary MBGs are on the same network in the DMZ.

Primary vMCD: 10.1.1.10/20
Secondary vMCD: 10.1.1.110/20

Primary vMBG: 192.200.100.10/24
Secondary vMBG: 192.200.100.110/24

Everything seems to be work just fine except for this one issue.

If the Primary vMBG goes down then all of the ACD and Teleworker phones fail over the Secondary vMBG; works great.

If we make an Inbound call via a DID to any of the non-Teleworker phones we have two-way audio; as expected.

If we make an Inbound call via a DID to any of the ACD or Telework phones then we have one-way audio, Inbound Only, until one of two things happens.

1. They put the call on hold and pick it back up.
2. They transfer the call to another phone; Teleworker or Normal.

If they perform either of the two above steps then the call will have two-way audio.

For the life of me I can't figure out why they are having a problem and where it would reside.

Since there is a time constraint on when we can perform the testing I wasn't able to figure out what to do in the time I had left.

I tried a packet capture at the Secondary vMBG thinking the audio would travel through there, but it doesn't show anything going to the IP Address of the Teleworker/ACD phone we were testing with. I did verify that the phone we were testing with was connected to it, but I guess the Audio doesn't pass through it for this scenario.

I didn't have time to start a TCPDUMP on the Primary vMCD as I though the problem would have been with the Secondary vMBG not sending the audio to the phone.

Has anyone run into this situation before and what can I look at?

We are going to try the test again on Monday morning, but I want to be as prepared as possible. I know that I am going to start a TCPDUMP on the Primary MCD, but can anyone think of anything else to do?

Thanks,

TE

13
Mitel MiVoice Business/MCD/3300 / Audit Trail Logs
« on: June 22, 2020, 11:40:25 AM »
Guys,

Does anyone know if you can have the Audit Trail logs automatically either sent to or pulled from an external syslog type application whenever a change is made in the MCD?

Thanks,

TE

14
Guys,

I have a customer that wants their users to have both the Mobile and PC softphones for the MiCollab Clients. I have UCCv4 Standard licensing that allows both Mobile and PC softphones, but I don't know how to set this up and what the limitation of it would be.

Any help with this would be appreciated as I can't find any documentation on how to setup either softphone.

Thanks,

TE

15
Non-Mitel Chatter / Telecommunications Training
« on: May 28, 2020, 12:35:08 PM »
Telecom Trained Technicians/Engineers,

Since they don't teach telephony in colleges around here anymore it is hard to find people with the right set of skills and knowledge anymore.

Does anyone know of or has used any non-Mitel Training Platforms [On Site, On Line, or Learning Management Systems] that provide all the knowledge and skills required to work in the field these days; especially with SIP?

I am looking for training that can be brought in-house or done online that will bring new people up to speed on Telephony and SIP.

Thanks,

TE

Pages: [1] 2 3 ... 7