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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

I have several IPO r11 behind a Sonicwall with SIP trunks and no issues. We’re Avaya and Sonicwall certified... On the Sonicwall, have your IT provider: 1. Make sure SIP transformations are disabled. 2. Create an outbound NAT policy for the IP address of the IPO, and on the advanced tab of...

I was able to get caller ID pass through working on 11.0 FP4. Though there is some very strange behavior: 1) If you press "To VM" while a call is coming in, and that call then gets transferred out to a mobile (using DTMF breakout on the user's Voicemail settings), the pass-through caller ID...

I "downgraded" from 11.0 FP4 to 11.0 SP2 and had the same issue. So I don't think it's fixed. Do you have a ticket # with Avaya you could share? That way I can open a ticket and piggy back on to your hard work and not reinvent the wheel...

No, the name portion of caller ID is controlled by your carrier. You can ask them to create entries in the CNAM database. Think of CNAM like a DNS server.

I have the same issue. The outgoing caller ID over SIP on forwarded and twinned calls was working perfect on r10.1 -- the caller ID of the "original" caller would appear on the mobile phone. I upgraded to r11 (tried both SP2 and FP4) and now its broken. Outbound calls are now using the ICR...