Converting Tied Trunk T1 to SIP 1

[MrCaptObvious]

Hello All,

My location is in the process of changing circuits around and part of those changes is eliminating T1 circuits from our current infrastructure. With that being said, the company is pushing services to convert to the "obvious" SIP trunks. Our service is directly tied to the other location at the moment via T1 on a tied trunk. I've been learning this Avaya system as I go but would like to know what steps/hardware would be needed to convert? Mainly, when the new SIP line comes in, where would this need to be plugged into? I figure I can use SMGR to add the SIP config I need. On a side note, we are looking to convert digital phones to all IP phones and use only one ethernet cable as a pass-through for this service. Where would I need to connect internet connection through to route the internet via voip phones. I assume I will need to connect it through one of the extreme switches and Vlans.

Our current setup:
2 G450
CM 7.1
SMGR 7.1
SM 7.1
2 ExtremeSwitches 4900

 

[biglebowski]

Converting E1/T1 to SIP is fairly simple. Not sure about the US but in the UK SIP is generally delivered over an etherlink circuit via fibre. Get an SBC to terminate it on as it makes life simpler (and more secure)

CSP >> Fibre connection >> fibre to ethernet converter (ADVA in UK) >> Router (supplied by CSP) >> Outside interface of an SBC >> session manager

We also use data passthrough on our 9608 handsets, you don't need to worry about the "internet connection" that will be done by your network team. We manage it via DHCP and on the access switches have two VLANs one voice VLAN and a data VLAN. When the phone boots it goes on to to the data VLAN and via option 242 gets told it has two vlans it then sends a DHCP request on the voice VLAN where (via 242) gets back the procr and http address etc.

Option 242 = L2Q=1,L2QVLAN=xxx,PHY2VLAN=xxx in our case L2QVLAN is the voice vlan and PHY2VLAN is the PC VLAN which has internet access via a default gateway

If you are going with SIP handsets read this:

https://documentation.avaya.com/bun...riesIPPhone_r4.0.12/page/VLAN_separation.html

 

[jimbojimbo]

CM/SM/SMGR 7.1 is definitely capable of supporting SIP. As biglebowski noted, you should install a Session Border Controller between your Session Manager and the SIP Carrier. Most SIP carriers will utilize a dedicated circuit (or part of one) to deliver the SIP trunks. There are SIP carriers who will deliver services over the internet however it does increase your exposure to hackers.

The "tie trunk" shouldn't be a problem (assuming both ends are Avaya 7.1+). CM->SM<-CM. G450 and CM Processor Ethernet will handle the signaling and audio/RTP traffic.

Keep in mind Modem/FAX may present an issue. G450 with MP160 DSP resources supports T.38 with G.711 fallback and Error Correction Mode. Alternatively you can use a FAX server or retain digital/analog trunks for FAX. I haven't seen a carrier saying they can support high speed analog data over SIP.

You really need to engage someone who does this on a regular basis.