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Topics - Tech Electronics

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1
Guys,

I have an issue with EHDUs on a system. It was working and at "some point in time" it stopped working. I looked at what I could remember of how it should be programmed and I don't see anything wrong.

It looks as though it is trying to call out as I see the users second multicall key flashe when a call comes in, but it never makes it to the cell phone.

Thanks,

TE

2
MiVoice Office 250/Mitel 5000 / Socket SIP Trunks - Hold Issue
« on: December 05, 2017, 10:15:13 AM »
Anyone,

Has anybody ever worked with Socket Telecom using SIP Trunks over a third-party private network? The issue we are running into is if a call gets put on Hold for any reason then Socket will send a private address to respond back to. When we respond to that IP Address the packets are dropped. Socket wants us to only send SIP Messages to the FQDN and disregard any other messages they send that have any changes to the Route or Contact information. As you might have guessed, by default, the system will send the message back to where it is told to and then the call is dropped when there is no reply; go figure.

The only thing I can think to do is to put something in between that will perform  a Stateful Packt Inspection and then a SIP Transformation on the packets that have the Private IP Address. I am not even sure such a thing exists that can be programmed to make specific changes like that.

Thanks,

TE

3
Mitel Gurus,

Alright, here is a situation that I have been trying to figure out with Mitel with no success so far.

We have a customer with a single MBG in their network for SIP trunks. Currently they are setup in Server-Gateway Mode with the first LAN NIC setup on the Voice Network and the WAN NIC and Gateway setup on the SIP Providers LAN. In this configuration it sets the SET-SIDE Streaming Address to the SIP Providers WAN Interface and the ICP-SIDE Streaming Address to the LAN Interface; all works as expected.

Now, the customer wants to add Teleworker phones into the mix and I have been unable, up to this point, to get it working. Mitel had me change the server configuration to the following.

Reconfigured the server to set both NICs to be LAN NICs as follows.

LAN NIC 1: Voice Network
LAN NIC 2: SIP Provider Network
Gateway: Tried Both Voice Network & SIP Provider Network; no difference

Then I went into the Network Profile and created a Custom Mode Profile
SET-SIDE Streaming Address: Public IP Address pointed to LAN NIC 1
ICP-SIDE Streaming Address: LAN NIC 1

After that I went to SIP Trunking and set the following.
Remote Trunk Endpoint Address: Carriers LAN IP Address
RTP Address Override: Third Interface [LAN NIC 2]

After setting that all up I Stopped and Started the MBG Service as required due to the RTP Address Override and tested.

The MBG was able to get to the AMC with no problems so licensing is good. I then called into the system from cell phone and answered the call on one of the desk phones. At that point I didn't have audio in either direction, and after a few seconds the cell phone got a busy signal while the desk phone still had no audio. I currently had the Gateway for the MBG setup for the SIP Provider's Gateway Address. After changing the Gateway to the Voice Network there was no change to how this was working.

Next I tried to make a call from the Teleworker phone to a phone on the Main Campus. The desk phone was able to hear the teleworker phone with no problem, but the teleworker phone was unable to hear anything. So, I reconfigured the server to change the Gateway Address to the Voice Network and tested again; no difference.

So, at this point, I put everything back the way it was so they can at least get calls in and out of the system. If anyone has any ideas on how to make this happen I would appreciate it. I am still working with Mitel on this as well to see if we can get it working, but their next step was to make both NICs setup as WAN Interfaces, but I couldn't figure out how to do that.

Thanks,

TE

P.S. - I have attached a pdf of a very high level drawing on how this is setup.

4
Mitel Gurus,

So, I have a system that uses SIP Trunks for outgoing calls. When the customer tries to setup a conference it gives an Invalid error on the phone and provides an error tone in the handset.

Steps.

1. Dialed first number. OK
2. Press Transfer/Conference Key
3. Dial 8+ second number. Invalid

At that point I tried to make it work with an internal phone.

1. Dialed first number. OK
2. Press Transfer/Conference Key
3. Dialed DN 1234. OK

Since that worked I tried it a different way.

1. Dialed first number. OK
2. Press Hold
3. Dial 8+ second number.
4. Add Held
5. 3-Party Conference. OK

Now, at this point I can't figure out what the issue is with the conferencing. Hopefully someone here will have an idea that I can try out tomorrow as I have to leave the site at this point.

Thanks,

TE

5
MiVoice Office 250/Mitel 5000 / MiVO-250 SIP DNS SRV [Information]
« on: November 06, 2017, 07:03:18 PM »
SIP Enthusiasts!!

Let's look at how an SRV Record is created and what to look for when you are given one.

The SRV record contains more information than the typical DNS record as it contains specific information that can be used to locate a specific resource at an address. This is an example of what an SRV appears like in a typical BIND-style zone file:

_sip._udp.example.com.  86400 IN SRV 0 5 5060 sipserver.example.com.
|__________________||_____||_||____|_|_|____|__________________|
          1                  2      3  4      5 6   7             8
1   Address - Location of the SRV record, including the resource type (_sip) and protocol (_tcp or _udp) - This information is often given by the provider of the service the SRV is being set up for.
2   TTL - Expiration value of the record.
3   Internet-type - Standard BIND notation indicating that the record is on.
4   InternetRecord type - Standard BIND notation indicating that it’s an SRV record.
5   Priority - Much like an MX record, one is able to set priority of more than 1 record at the address.
6   Weight - There is provision for a rudimentary load balancing schema. All values across all records at the address must add up to 100. The higher the weight, the more often the specific record will be served and vice versa.
7   Port - TCP or UDP port where the specified service can be found.
8   Target Endpoint - the canonical hostname of the machine providing the service.

Now go to: MX Toolbox SRV Lookup

For instance if you go to that link and put the following in the field: _sip._udp.nexvortex.com you will get the following results.

px5.nexvortex.com - 66.23.190.100
px1.nexvortex.com – 66.23.129.253
px7.nexvortex.com – 209.193.79.80

As you can see px3.nexvortex.com is not showing up even though we have that in our default settings in most systems I have done.

If you look at the registrar address. _sip_udp.reg.nexvortex.com you will get the following results.

reg1.nexvortex.com – 66.23.129.110
reg2.nexvortex.com – 66.23.190.120

You can then click the link associated on each one of those and get the actual IP Address associated with them. Now, Nexvortex also has an actual A-Record associated with the base as well so that will give you a completely different IP Address that isn't associated with their SIP service.

If you wanted to do this with NSLOOKUP here is how. Go to Command Line. CMD.EXE and then just look at the entries at the command prompt >

> nslookup
> set type=SRV
> _sip._udp.nexvortex.com
> _sip._udp.reg.nexvortex.com

Microsoft Windows [Version 6.1.7601]
Copyright (c) 2009 Microsoft Corporation.  All rights reserved.

C:\Users\sbay>nslookup
Default Server:  homeportal
Address:  xxxx:xxx:xxxx:xxxx::x

> set type=SRV
> _sip._udp.nexvortex.com
Server:  homeportal
Address:  xxxx:xxx:xxxx:xxxx::x

Non-authoritative answer:
_sip._udp.nexvortex.com SRV service location:
          priority       = 30   This server is the last one to take calls.
          weight         = 0   Not Load Balanced
          port           = 5060
          svr hostname   = px7.nexvortex.com
_sip._udp.nexvortex.com SRV service location:
          priority       = 20   Followed by this server
          weight         = 0   Not Load Balanced
          port           = 5060
          svr hostname   = px5.nexvortex.com
_sip._udp.nexvortex.com SRV service location:
          priority       = 10   Most of the calls will be taken by this server.
          weight         = 0   Not Load Balanced
          port           = 5060
          svr hostname   = px1.nexvortex.com
> _sip._udp.reg.nexvortex.com
Server:  homeportal
Address:  xxxx:xxx:xxxx:xxxx::x

Non-authoritative answer:
_sip._udp.reg.nexvortex.com     SRV service location:
          priority       = 10
          weight         = 0
          port           = 5070
          svr hostname   = reg1.nexvortex.com
_sip._udp.reg.nexvortex.com     SRV service location:
          priority       = 20
          weight         = 0
          port           = 5070
          svr hostname   = reg2.nexvortex.com

So, there you have it a simple and easy way to figure out what A-Records are associated with a DNS Service [SRV] record, go forth and be better armed!!

If you are working on a MiVO-250 you can put the A-Records as individual entries inside the Route Sets section of your SIP Peer Configuration as it doesn't support DNS SRV, but you can trick it to do what you need it to do.

Thanks,

TE

6
Mitel Software Applications / Welcome E-Mail Links
« on: October 25, 2017, 08:20:11 AM »
Does anyone have all the links that are sent out in a welcome e-mail that they could share?

Thanks,

TE

7
Mitel Jobs and Mitel Careers / Mitel Technician in St. Louis, Mo.
« on: October 24, 2017, 10:47:24 AM »
Guys,

Tech Electronics is looking for [2] technicians to work in the St. Louis area. Take these requirements with a grain of salt as they are obviously a wish-list.

https://techelectronics.aaimtrack.com/jobs/186512.html

Thanks,

TE

8
Mitel MiVoice Business/MCD/3300 / Configuring MCD - Mitel TA7108 ATA
« on: October 10, 2017, 08:53:30 AM »
Guys,

Does anyone have a document that explains how to configure the 3300 to work with the Mitel TA7108 ATA. I downloaded the SIP CoE document, but it says I need IP User licenses not Single Line licenses to make it work. I remember there was a change to this not long ago, but I don't have a document that explains how to configure the 3300 for this setup.

Thanks,

TE

9
Guys,

I have to perform a system restore this morning on our lab environment before fixing the database and restoring it remotely at a customer location. It is telling me that I have to perform a manual reboot, but there isn't a reboot command in the list of Maintenance Commands. I did find a Reset System command, but it states that it won't reboot the APC. Is the reset command sufficient for this process since the manual doesn't state how to actually perform the manual reboot?

Thanks,

TE

10
Mitel MiVoice Business/MCD/3300 / Major Alarm but no Active Alarms show
« on: September 29, 2017, 12:29:15 PM »
Guys,

I have a new installation on a virtual MCD:

Release: 8.2 SP-2
Version: 14.0.2.26

The system shows a Major Alarm for the Group Alarm Status and when I look at the Admin Alarm Group Summary it shows the Major Alarm on the node I am on. I click on the link for that node and it doesn't show any alarms at all. I then went to every other node and none of them show any alarms. I then went to Maintenance Commands and typed in Show Status Alarms All and it shows no alarms as well.

I then went into System Properties\System Administration\SNMP Configuration and set it to set the Enable to No and then back to Yes and the alarm came back.

Has anyone run into this problem and figured out a way to get rid of the alarm?

Thanks,

TE

11
Guys,

Alright, here is another request that the sales person said that we can do. The salesperson no longer works here and this can't be verified on how they expected this to work.

The customer has Call Center Agents that they want to be able to change their Calling Party Number Identification for each call that they make to their customers; based on the product that the customer is being call for.

For instance, if the customer has a Valvoline account and they want to call them then they want a CPN that would be different than a customer who has an Integra account. This way when the customer calls back they will do so to the 800# that the Agent Called from.

They want this to be as easy as possible for their Agents two switch between CPNs; if this is possible at all.

The phone system is Virtual and they have 30 SIP Trunks to use for all call types.

Thanks,

TE

13
Mitel MiVoice Business/MCD/3300 / DTMF Issues - SIP Trunks
« on: August 11, 2017, 02:08:24 PM »
Guys,

Does anyone have a good way to track down and resolve issues with DTMF over SIP Trunks on a MiVoice Business Express?

The system doesn't have this issue with all Auto Attendants and it isn't always the same button press. The MiVB-X is using its internal MBG to talk to an SBC on the same VLAN and for the most part I have not been able to replicate the issue myself, but it is happening enough with the customer that they want us to look into it as they see it as a problem.

I had them try to call our test number it works fine every time from their phone as well as any phone on the system. I know that the DTMF Payload is 101, but I can't tell whether or not it is set for RFC-2833 or whether it is in-band or out-of-band. I am at a total loss as to how to proceed on a 3300 with this issue.

Thanks,

TE

14
Guys,

I ran into an issue where we had several 6930 phones that came up fine originally, or so we thought, and after a few days they started rebooting continuously. The issue was the phones came at Main Version 1.0.0.125 and had to be manually upgraded to 1.0.0.148 before they would download and install the firmware from the controller; Main Version 1.1.0.151.

You can go to the KB and get the software needed to upgrade the phones.

Thanks,

TE

15
Guys,

When I have the feature bits for both Direct Page and Direct Voice Calls set to Yes the Direct Page through a Page group doesn't seem to come through the phones, but the speaker button lights up. If I use the Direct Voice to any of the same phones it works as expected.

If I set the Direct Voice Call bits to No then the Direct Page works as expected.

The customer is wanting both features on the phones as it was part of their RFP for the job. Does anybody know if these features are supposed to work together or not?

Thanks,

TE

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