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I can make a call on the mobile app but have no audio. I am on the latest release and have a FQDN setup using Cisco Meraki firewall with 1:1 NAT on a dedicated WAN IP. I have all ports forwarded.

After upgrading the phones are still searching for firmware. The upgrade completed successfully, there are 9508 phones and 1416 phones. manager is set to SD card and there are no apparent errors

After completing a call the system does not disconnect and plays a tone/hum. I have adjusted the settings in Manager 9.1 (upped the disconnect to 255) with no success.

The customer is complaining about DTMF not forwarding to various AA/IVR systems across the PSTN. We tried adjusting the Auto Gain Control setting and also verified they are using the handset when making calls. Comcast came in and connected a TBIRD to the PRI to show the customer it's the Avaya...

The Bogen PCMSYS adapter answers the call (analog port) and prompts for the user to enter a code for the zone. Is it possible to program the button to wait for the Bogen to answer (about 5 seconds) and press the code automatically?

Can this be reset?

Does anyone have information regarding any planned releases with regards to the web manager? It would be nice to have a fully functioning web manager vs having to load the utility for changes to the AA & ICR.

R9 VM Pro on Linux Apps Box Time through System Status, Voicemail Pro Client and Web Portal for Apps Server all show correct time. When viewing a message through visual voicemail the time is off by 5 hours. Any thoughts?

Note: IP Office Release 9.1 will no longer support the following legacy IP 400 modules: • Phone 8/16/30 • DS 16/30 • DIGITAL S0x8 and DIGITAL S0x16 • Extension/VCM Carrier Is the PRI still supported on a legacy card carrier?

We have our SCN working and can dial extensions and hunt groups directly between sites. However, we cannot add each other's hunt group into overflow. How is this acheived? I do see "SCN 1" when trying to manage but it says there are two different subnets, we are using a 192.168.3.0/24 and...

The system answers the call and shows the call connected to the AA and then routing to the appropriate group. However, the call is placed on "hold" the issue is the hold music is not on the Avaya so this has to be coming from the carrier, Comcast. Has anyone seen this issue? Of course this is...

Is it safe to copy a config file for two identical systems with the same hardware and software release? Also, is it safe to build a base config and use that config for multiple deployments moving forward?

We cannot get calls to forward unconditionally from any users phone. I have tried setting up an additional SC using 8N and dialing out directly on the line group 0 skipping the ARS. With and without the 1 (15085551212-5085551212) It just drops the call back to the main auto attendant.

IPO R9 Essentials 9608 IP Phones Isolated POE LAN Called party does not recognize DTMF if the phone is on speakerphone. If the call is initiated by lifting the handset DTMF works fine. If the call is initiated on speakerphone or initiated on the handset and then switched to speakerphone it does...

What's the best way to find out the location of all extensions/ports in a Merlin Magix system? What is the typical numbering plan? We have two cabinets - from left to right: processor 412 tdl 800 iclid 800 iclid 016 t/r 016 t/r ---------- messaging 016 t/r 100 ds1 blank blank blank

Is this possible on R6 without a VPN?

We recently deleted the user accounts in security settings that we do not use to prevent toll fraud. We also changed the Administrator and Security passwords. It seems like any new phones we connect will not pull DHCP from the IPO. This could be a coincidence but could we have deleted a user...

Is it generally a best practice to assign a public facing IP to the WAN port? We're struggling with SIP through a firewall and will most likely place a public IP on the WAN port. This also seems like it will be convenient for remote phones (R9 9608 Off-Site). What are some best practices in...

We have about 25 POTS lines that are undocumented, we have three main CO 300 pair feeds coming into the building untagged. What's the best way to verify this? Will the phone company give out the pair number?

Where do you set the limit for upgrading to Voicemail Pro, an example would be a system with 30-40 users who doesn't need call recording, conference bridge or campaigns. Would they still qualify for Embedded Voicemail? Is their a maximum amount of seats you would use for embedded?