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Topics - handwritten

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1
We recently removed a 200ICP from our environment. We have a MXe-III running 9.1.1.32, which previously had ARS routes pointed at the 200. We removed the 200 from the Network Elements page in the 3300, but we still get ICP Comms Card alarms complaining about the absent 200. What else do we need to do? The extensions on the 200 are no longer in service.

2
We use SIP trunking between our Mitel environment and our Teams environment, with an SBC tying it all together. For calls from Mitel to Teams, the 'From' and 'P-asserted Identity' fields in the SIP INVITE include the calling user's name. We want the DN/extension instead. I've been through the SIP Peer Profile extensively, as well as the COS settings. Nothing! If I can get the DN to show up somewhere in the INVITE, I can massage things in the SBC, but it's nowhere to be found. Any help would be greatly appreciated!

3
Mitel MiVoice Business/MCD/3300 / TA7108: Packet capture protocols
« on: January 07, 2022, 02:05:25 PM »
We have a couple of TA710X units, and the learning curve is very steep. We're trying to use packet capture to help us get to the bottom of an issue, but there's no documentation on how to send the .pcap to a remote machine. The built-in help states:

Code: [Select]
URL where to send the packet capture. The URL should follow this format:
protocol://[user[:password]@]hostname[:port]/[path/]filename

The brackets [] indicate an optional parameter.

The filename may only be composed of alphanumerical and '-._%$' characters as well as macros. The macros used in this field are replaced by the unit's MAC address or date/time of when the capture was started.

The supported macros are:

%mac% - the MAC address of the unit.
%date% - the date if the capture start in format YYYYMMDD.
%time% - the time if the capture start in format hhmmss.
The supproted transfer protocols are:

HTTP
HTTPS
FILE
Examples of valid URLs:

http://httpserver.com:69/folder/capture.pcap
http://guest@httpserver.com/capture_%mac%_%date%_%time%.pcap
https://username:password@httpserver.com/capture.pcap
file://capture.pcap
The protocol default port is used if none is specified.

This seems odd to me - we're used to using TFTP/FTP for this kind of thing. Has anyone done this successfully? What utility did you use?

4
This is an odd one. We use DHCP relay on a Cisco 9500 core switch. We recently upgraded from 16.something to 17.3.4, and now all the DHCP Discover messages from 5212 phones are getting dropped by the core. The core gives us an error "DHCPD: Invalid option 124 information. Dropping DHCP message". I confirmed with a packet capture that the option 124 'V-I Vendor Class' in Discover messages from those phones is 4 bytes long, where the standard is 5 bytes or longer. Wireshark flags this as an error. This seems to be the bug ('caveat') that Cisco acknowledges here: https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvt34738. We have not tried the proposed workaround because it's not relevant for our environment. We're currently stumped.

5
Mitel MiVoice Business/MCD/3300 / ARS last-ditch route
« on: December 06, 2021, 03:20:59 PM »
Is there a way to program an ARS Digits Dialed route to catch any dialed digits that a) do not exist as users local to the system and b) do not match any other ARS Digit Dialed entries? I thought maybe Intercept Handling was the way to go, but that just sends such calls to a specified extension. I need to send those calls to a trunk (ie, a route number).

6
How can we display the calling party number for incoming calls on a SIP trunk? At the moment, the phone display shows the string in the Trunk Label form. We know that the calling party information is included in the P-asserted Identity header (and privacy:id has been removed). I've been digging through the Trunk Attributes, and the CoS settings, but nothing seems to work. Any advice would be greatly appreciated!

7
We have a 3300 with a SIP trunk to an Audiocodes Mediant SBC. We use this for a Teams Direct Routing implementation. We can call from the 3300 to Teams via the SBC, but not the other way around. Calls are dropped, and the 3300 returns an ICMP error. We double checked the port and protocol (UDP 5062) as set up in the Network Elements form on the 3300. Any clues?

8
We have a 3300 (with backup VMBG) plus two MBGs for SIP trunking. We have about 600 users. We are interested in using Microsoft's Direct Routing for Teams users. This requires an SBC.  Unfortunately, Mitel's MBG is not listed.  Here's the list of compatible SBCs: https://docs.microsoft.com/en-us/microsoftteams/direct-routing-border-controllers . Mitel has their own list of supported SBCs in the 'Interop Reference Guide' but none of those platforms are also on Microsoft's list.

Perhaps I don't fully understand what the SBC is doing in the Direct Routing environment - it seems like that role could be filled by the MBG.

Has anyone implemented Direct Routing with a Mitel environment?  What SBC did you use?

Thanks in advance!

9
We have two MCD installs: one MXe and one VMCD, both running 7.0 PR1.  We use SIP trunking with two MBG units running V8.0.27.0.  Both MBG units are configured for both MCDs.  During a recent power disruption that lasted 10 minutes*, phones failed over from the MXe to the VMCD as expected, but incoming calls received dead air.  Outgoing calls were fine.  How should I go about troubleshooting this?


*By the way, I expected that having two power supplies would maintain service during the replacement of one power supply (this is mentioned in the documentation).  However, when power is lost to one power supply, the unit reboots when power is restored.  Has anyone else seen this?  Is this a feature of the platform, or do I have a faulty unit?

10
This is a weird one.  I have a 5304 that has two multicall lines as well as the prime DN.  When I call the multicall number, other phones can pick it up, but not this one.  When I try to pick up the call, I hear a dial tone!  There's no DND set on the phone, and nothing different about this configuration.  What am I missing?

11
We have two physical MBGs that are only responsible for SIP trunking duties.  They are both currently exposed directly to the internet.  We would like to virtualize one of them, and to do so we need to move it behind our enterprise firewall (a pair of Fortigate 1000Ds).  I set up the incoming/outgoing firewall rules using a VIP and NAT (respectively)*, and changed the network profile to LAN mode.  I see that the RTP streaming IPs are now both set to the LAN IP.  The SIP trunking status looks OK (green checkmark!), but I can't make any calls through that MBG (I just get a busy signal).  I'm snooping traffic from that MBG, and it doesn't appear to be communicating with the service provider.  If I replace the MBG with a workstation configured with the same IP, it has internet access. 

When I make the switch to LAN mode, I disconnect the WAN NIC on the MBG to make sure traffic goes out the LAN interface.  Curiously, this breaks the internet connectivity test.  Should I perhaps configure the LAN IP on the NIC known as WAN, and use that instead?   

Any clues? 

*As per the MBG Engineering Guidelines:
EXT to INT: TCP/UDP 5060, UDP 20,000-31,000 (using a Virtual IP pointing to the MBG LAN IP)
INT to EXT: TCP/UDP 5060, HTTPS, SSH, UDP 1024 - 65535


12
I'm deploying some Talkaphone emergency phones on my 200ICP.  The phone dials an extension hosted on a 3300, and the call works as expected.  However, when the remote party hangs up, the line stays open on the phone.  There is a fast-busy, then eventually the phone's timer expires and it hangs up.  Talkaphone support tells me their phones need a CPC.  Can the 200ICP provide this?  They also mention I could use a Viking CPC-1, which I may have to use if I can't get this to work.

From the device manual:"If connected to a PBX, your extension must provide:  at least 24 Volts at 20 mA off-hook (no current is drawn on-hook) either a disconnect pulse (voltage drop at end of call) or 30-seconds of silence after hang-up (no re-order or howler feature)."

Any ideas?

Thanks in advance!

13
We have a 3300 clustered with a VMCD instance, both running 7.0 PR1.  The 3300 has an audit failure alarm.  There are no problems reported when we run a DBMS CHECK FULL, and a DATABASE AUDIT results in "Database audit succeeded".  We can't make a backup: "Backup Fail".  The documentation suggests this can only be cleared with a database restore.  We reset the system this morning but the alarm persists.  Does anyone know a way around this?

14
A few weeks ago, conferencing stopped working for all users on 5312/5212 and 5304 sets.  Interestingly, it still works for users on 5340 sets. 

When Alice presses the trans/conf button while on a call with Bob, she is able to dial Carol and talk to her.  But when she presses the button again, there is no incoming/outgoing audio, even though the screen shows 3 PARTY CONF.

The 3300 has been rebooted since this started, and that has no effect.

I made sure that the System Options fields were set correctly:

Code: [Select]
Maximum CO Trunks In A Conference 3
Maximum Parties In A Conference 5
Maximum Trunks In A Conference 4

Any thoughts?

15
Can anyone comment on experiences using the 5550 or 5540 consoles?  Is there one that you'd recommend over the other?

Also, what about using a 5340 as a console?  One notable drawback is that operators would have to press an extra key (transfer) instead of simply typing the extension to transfer calls. 

We currently use a Superconsole 1000.

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