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Toshiba IPEdge WatchGuard SIP Trunk Issues

Toshiba IPEdge WatchGuard SIP Trunk Issues

Toshiba IPEdge WatchGuard SIP Trunk Issues

(OP)
OK Network guru's, i really need some guidance on this one.

Setting up Toshiba IP Edge 1.7.1 F/W build for SIP trunks. Trunks come from provider on a block of 3 IP addresses, run through a WatchGuard XTM series firewall. No other equipment between the firewall and the network switch. Any content filters or service blocks are handled by the firewall only. Current setup has a PRI circuit coming into an Adtran 908 and converting to SIP trunks to send to the IP Edge. Everything works correctly.

Called Toshiba to verify config, worked with Watchguard Tech Support on this, and 2 other network admins couldn't figure out why this is happening.

Inbound call to DID> rings phone> call connects and has 2 way audio.

Outbound call from phone> rings cellphone> call connects and has 2 way audio

Outbound call from phone> rings cellphone> call connects and has 2 way audio> transfer outbound phone call to different extension and everything works great.

Inbound call to DID> rings phone> call connects and has 2 way audio> transfer inbound call to different extension and call connects but we completely lose audio.

The phone system is changing the port number when the call is transferred. We can watch the traffic monitor on the firewall and see it blocking the port change. But everyone who monkeyed with the firewall said that all the IP's and ports needed were opened. Yet it was still blocking us. Again, we only lose audio if we transfer an inbound call from one extension to another. And it doesn't matter if the transfer is manually from phone to phone or a blind transfer from the auto attendant. We still lose Audio on the inbound call.

Does anyone have any suggestions on how to resolve this? And telling me that we need a different firewall isn't going to work. (I tried to say that when the task first came up and i was told it's not an option to change firewalls.)

RE: Toshiba IPEdge WatchGuard SIP Trunk Issues

So look at the refer or reinvite settings concentrate on the change in the SDP , i dont know the hardware but for my money thats where you need to start.......

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

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