Recent content by DavidCT

We’ve been moving clients to 1-VoIP from Flowroute for this very reason… Proactive Business IT Support | Cybersecurity | On-premise and Hosted VoIP https://www.canaantechnology.com

Posted to the support site. I was told since it's not a major release, a system licensed for release 11.0 can be upgraded without an upgrade license. Who's willing to live on the edge and upgrade their system?? Proactive Business IT Support | Cybersecurity | On-premise and Hosted VoIP...

Your best bet is to do an SD card recreate. Stepping up from r6 to r11 will leave you with next to no available space on the SD card. Make sure to have your PLDS license file and don’t forget to backup the config to your computer. Proactive Business IT Support | Cybersecurity | On-premise...

Using analog lines severely limits you. Your only option in this scenario is the 9608 phones utilizing a VPN. No special licenses are required, but you do need enough Avaya IP endpoint licenses. What you really need to do is: - Upgrade to r11 - Add SIP trunks to your system - Utilize Mobile...

I’d install one of these: https://www.vikingelectronics.com/products/e-1600a/ Viking e-1600a Emergency Phone with Announce

Opened a ticket with Avaya and sent traces, etc. They came back with the official fix (for now) is to downgrade to 11.0.4 SP1 and apply a hot fix/patch. I'm waiting to hear back on the purpose of the hot fix, rather than just a plain downgrade.

If you click on the link to download you will see: *** We have temporarily disabled 11.0.4 SP2 downloads while investigating a number of issues ***

After upgrading from 11.0.4.1, I can make our 500v2 crash on demand just by making two or three calls in quick succession to the same phone number over SIP using a J179 phone. Seems to be fine from 96xx series H323 phones. Sysmon shows just before the crash: PRN: .FATAL TLB Data Load Error...

I have this working... Your SIP provider must support it. If they do: Change the PAI fields for “forwarding/twinning” to original caller. Remove diversion header. On your ARS, use Nsxxxxxxxxxx where xxxxxxxxxx is your regular office caller ID.

Here's a guide I wrote to confige a SonicWALL for SIP Trunking with Flowroute. The document can be adapted to just about any SIP provider. https://www.canaantechnology.com/wp-content/uploads/2019/07/SonicWALL-Flowroute-Avaya-IP-Office-Port-Forwarding.pdf

This is a firewall issue. There’s only so much you can do on IP Office to compensate for it. At some point (now), you need to undo all the workarounds and put in another firewall, even if just to prove the theory. We use Sonicwall, but have a few sites with Fortinet. Here is a link to setup...

Have you tried: 1) On the Network Topology Tab, changing the value for binding refresh? 2) A different firewall make/model? 3) Changing your FlowRoute POP to a different server, even if it's further away from your location?

You don't have it correct. You are only allowing audio from one of the Flowroute POPs, not "anywhere".

Flowroute requires that you have the RTP port range configured in the IP Office open from ANY to the private IP of the PBX on your firewall. Have you done that?

Here's a guide I wrote to confige a SonicWALL for SIP Trunking with Flowroute. The document can be adapted to just about any SIP provider. https://www.canaantechnology.com/wp-content/uploads/2019/07/SonicWALL-Flowroute-Avaya-IP-Office-Port-Forwarding.pdf