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Has anyone successfully connected Shoretel to Cisco via SIP trunks? I've searched for all the articles related to SIP trunks, and can't find anything definate.

I've read some things that say you only need an Ingate if trying to connect to a Telco SIP provider or going through a firewall, but not if you are connecting PBX to PBX on the same network.

If you do need some type of Session Border Controller, will a cisco router setup as a CUBE work?

I think I have the Trunk Group, Individual Trunks, and Switch's configured correctly, but I can't seem to get anything going. While trying to use the trunk tool to make a test call, nothing happens (log length doesn't even change).

The Cicso system use's 5 digit dialing, we currently use 4. I'm trying to configure it so you dial 8 + 5 digits to get from Shoretel to Cisco extensions.

I'm not familiar with Cisco, but that team use's Verizon SIP trunks. They would be dialing a 6+4 digits to go from Cisco to Shoretel extensions.

Initially, we'd just like to get extension to extension dialing, but would like to eventually use their Verizon sip trunks for outbound calling.

the main thing the SIParator does that makes SIP trunks work is deal with the NAT problem by providing a public IP for the far end to direct packets to. The InGate then direct the packets to the correct (and possibly changing) internal IP address. If the far end could directly route to any IP phone on your network then you wouldn't need an InGate in the middle (don't actually put your whole network on the Internet!)

If you are dealing with two phone systems you own that can directly route to every one of the other systems end points then you don't need a device to do NAT translation. I've seen Cisco to ShoreTel SIP tie lines work without issue. You will need a digit translation table to map 4 digit extension numbers to 5 digit extension numbers using Off System Extensions.

Actually, we just upgraded to 5 digit's to match the Cisco system, so dialing 8 to access the trunk wont be needed anymore. Now that both are 5 digits, I can configure Off System Extensions.

But the problem I'm still having is the SIP trunks aren't working. I'm going to try and delete the tie Trunk group and individual trunk's and recreating. When using the TrunkTest tool, it shows the SIP trunks, but wont let me test them.

On the Cisco side, the guys have told me I should be able to use 1 of 2 different IP's. The first IP is a Cisco CallManager subsciber, and the other is a Cisco CallManager Publisher server, both are on Cisco CallManager 7.0(2a).

I'm getting closer, still having an issue with Cisco users calling Shoretel user's though.

Shoretel can calls Cisco extensions without any problems, but Cisco user's get a fast busy when trying to call Shoretel user's.

You're going to have to set something up on the Cisco so that it knows that certain extensions are reachable through the tie line. This is what off system extensions do for ShoreTel.

I think our cisco guy(CCIE) have gotten that far. I have a copy of the logs I can send if you can help. I don't completely understand the logs, but I do see where trying to dial my extension reference's our shoretel SIP switch's ip address.

I'll post an example of the log. Does anyone know where I can find similair SIP logs for Shoretel?

05/04/2010 11:58:19.952 CCM|//SIP/Stack/Info/0xdf094c8/****Adding to UAC table.|<CLID::StandAloneCluster><NID::emscucm02><C T::3,100,37,1.965131031><IP::10.5.101.242><DEV::SE P001E13E5BC23><LVL:etailed><MASK::40000>
05/04/2010 11:58:19.952 CCM|//SIP/Stack/Info/0xdf094c8/Added to table. ccb=0xdf094c8 [email protected]|<CLID::StandAloneCluster><NID: :emscucm02><CT::3,100,37,1.965131031><IP::10.5.101 .242><DEV::SEP001E13E5BC23><LVL:etailed><MASK::40000>
05/04/2010 11:58:19.952 CCM|//SIP/SIPHandler/ccbId=2696041/scbId=0/findDevicePID: Routed by ccbId/scbId|<CLID::StandAloneCluster><NID::emscucm02><CT ::3,100,37,1.965131031><IP::10.5.101.242><DEV::SEP 001E13E5BC23><LVL:etailed><MASK::20000>
05/04/2010 11:58:19.952 CCM|//SIP/Stack/Transport/0x0/Dialog Transaction Address 10.30.0.17,Port 5060, Transport 2, SentBy Port 5060|<CLID::StandAloneCluster><NID::emscucm02><CT: :3,100,37,1.965131031><IP::10.5.101.242><DEV::SEP0 01E13E5BC23><LVL:etailed><MASK::40000>

Turns out Cisco UCM will accept incoming SIP calls on either TCP or UDP 5060, and by default will send outbound calls on TCP 5060. But ShoreTel will only accept incoming SIP calls on UDP 5060.

Is there anyway to set Shoretel to answer both UDP & TCP for excepting calls?

I'm also still looking for a way to check Shoretel SIP logs (TrunkTest tool doesn't given enough info).

Did anyone find the resolution to this yet. I'm currently hip deep myself in this exact same problem but with a Mitel5000.

The closest thing I got to an answer was to get a SBC device like an Ingate or Cisco Cube.

I've been busy with other projects, and I haven't been able to revisit this yet...

You can configure the Cisco SIP Trunk Security Profile to send using UDP instead of TCP. Then apply that Security Profile to the SIP trunk built using the Shoretel IP addresses.