CM and Session manager setup. (With possible lync integration)

[niftynifler]

Hi folks!
I have a question, one that might seem basic, but one I cant figure out nonetheless. My goal, get session manager and CM talking so that I can make a call from a softphone to CM (internal) via sip, and hopefully later to the pstn. (Or if I am completely missing how SM works, let me know )
I have installed SMGR, and SM, both 6.3.2, and have a CM 6.1 up and running. I have added CM in SMGR's managed elements, added CM as a SIP Entity, created the entity lync between CM and SM (port 5061, TLS)
This is where I am stuck. I've never played with session manager, or set up a complete system, as my previous workplace only had good old CM and my role was mainly linux admin. I dont know what to do next, set up a SIP trunk to SM I assume? Is there an install doc for that? If so, where in the world can i find it? (i've exhausted my normal research options, tek-tips - IAUG - support.avaya.com [which only gives 700page bs docs])
Basically, I'm a complete n00b with Session manager and do not understand it. How does it work, and how do I configure it to work with my CM? (I've had a look at this tek-tips post but the reference docs are unavailable and seem to relate to nortel cs1k. I just want a basic working sip system, but fear I wont even know if I have one.
Then, lastly, Lync integration. Have anyone done this before? I basically want to be able to dial an internal extension from the lync client, and reach a deskphone. And also be able to make pstn calls to outside numbers from the lync client. Any advise or docs on that? (Although, once I have the CM and SM talking, I feel like I would understand the whole system better, at which point I can probably wing it and install it myself)

 

[montyzummer]

First of all what trunks are you using currently ??

Session Manager is a sip proxy there are plenty of documents on how to set up a sip link between CM and SM (try tcp to start off with), you need a trunk and sig group in CM with SM added as a node name.

http://www.youtube.com/watch?v=T6sL50wlql4

session manager sip entity (create a trunk between sm and CM)

http://www.youtube.com/watch?v=Sius0AaLoBw

http://www.youtube.com/watch?v=4N07rAZ5zWo

setting up sip trunk in CM both links ,

As for trunking you then need my first question you cant simply point a PSTN sip trunk at Session Manager and get it to work , you need to provide more information on what you are doing as you may need ars tables route patterns to send your digits to SM then thats when you need to worry about getting the call from SM to the outside world.This is where SBC`s come into play and all that good stuff.

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[niftynifler]

Hi There, Thanks for the reply.

The trunks we are currently using; isdn (2 members) and one co trunk. (Its tiny, I know.) We predominantly use Lync as our voice solution, and have a solid running system. But every so often, someone wants to make a call to a telephone number in the pstn, and that is where we want to send the call through. In short, Lync client call hits our Lync server, goes to session manager, and use CM isdn/co trunk to breakout. So we dont care for inbound calls at all really, as our office is pretty much just a server room.

I checked out those videos, and have managed to setup a trunk to SM now. I'm quite embarrassed to admit that I never thought it would just be that. I assumed there would be more to configure, especially on the SM side.

So help me understand how this works. Now that my trunk is in service between CM and SM, and everything configured on both ends, one SIP user setup in SM (say, tester1@company.com), and one extension(say 5100) configured in CM. Would my sip user now be able to register his sip client to the SM, and perhaps initiate a call to the extension number? Or do I have to create the extension as a sip user as well?

 

[montyzummer]

Yes your sip user can be set up in SM but needs administering in CM to provide button features and the like as for extn 5100 what type of user is that ?

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[niftynifler]

The extension 5100 user is simply a CM pbx extension voip desk phone. I want to initiate a call to that extension from my sip client that I just set up.

For example, tester1@test.com registers to SM, and initiates a call on his sip softphone to extension 5100. My hope is that the call rings the 5100 extension and the two can talk.

 

[montyzummer]

have you tried it as that is certainly the idea , just make sure the domain on the signalling group and network region in cm, the phones are on match in your example case @test.com and all will be sweet

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[montyzummer]

I am assuming that you have your entity links set up and the like also in CM you will need something that tells the cm station to route down the sm trunk , this is easy to do in uniform dial plan and aar table

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[niftynifler]

Yup, got that all done. Thanks so much for your help here! I'm sure i have a good enough understanding to find my own way.

I'll double check the domain setups, as the sm links are up, but saying 403 forbidden (unknown far end) It seems to be coming from the CM, so i'll poke around!

 

[montyzummer]

where are you seeing the 403 forbidden ?? it sounds to me like the domains don't match , a TraceSM would help if i could see that here or a list trace on CM

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[niftynifler]

I see it on the traceSM from CM towards SM

20:06:09.487 |--OPTIONS->| | | (13) sip:172.0.0.11 -CM ip address
20:06:09.488 | |--OPTIONS->| | (13) sip:172.0.0.11
20:06:09.494 | |<--Forbidd-| | (13) 403 Forbidden(Unknown Far-End)
20:06:09.494 |<--Forbidd-| | | (13) 403 Forbidden(Unknown Far-End)

on a side note, Do i put the security module ip as the nodename or the normal SM ip? (in other words, what is the security module used for?

 

[montyzummer]

The security module is the node name that in essence is session manager , your need to add an adaptation to session manager
Something like

DigitConversionadaptor

iosrcd=cm domain on sig group / network region of cm ipphone

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[niftynifler]

Aaah! Awesome! two unknowns answered in one hit. I'll let you know how I fair.

 

[montyzummer]

i am guessing your domain on sesion manager and cm far end domain on the sig group match or do they not ??

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[niftynifler]

They match, yeah.

 

[montyzummer]

can you call from cm to sm , can you post a full list trace of the call ??


APSS (SME)
ACSS (SME)
ACIS (UC)

 

[montyzummer]

Also in your trace sm i see no invite and from info can you post that to actually see what information session manager is initially getting and trying to deal with.

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[montyzummer]

Another thing i forgot to mention is that you sig group needs to be st to ims

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[niftynifler]

Hi Montyzummer.

Apologies about the late reply, I have been swamped with other duties. I want to take a moment to say thanks for your help. I have managed to get my system working 100%. My main mistake was over-estimating the complexity of setting it up, and using the wrong ip address of the security module. (can you believe that.)

Cheers

 

[montyzummer]

Glad you got it working

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[wpetilli]

Can you share exactly how you were dialing to and from the endpoints? Also, what's the adaptations purpose? I am pretty ok with SM admin, but very cloudy on the whole adaptations.