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OCS2007


HedgeHog
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Hi,

 

I suppose you deployed pbxnsip, OCS & OCS Mediation Server, Certificate Stuff etc. and you only need some configuration infos.

 

icon13.gifUPDATE:

I have written a new pbxnsip Wiki page with a guide for pbxnsip-OCS2007 setup. Including helpful screenshots:

 

Basic Setup for pbxnsip / Office Communications Server 2007 Interoperability OCS

 

Best regards

 

Jan

 

icon13.gifUPDATE

 

@pbxnsip:

  • Create a trunk to your SIP-provider, Type=SIP Registration, enter all account infos (username, domain, outbound proxy etc.) given by your provider.
  • Create a trunk to OCS-Mediation Server, Type=SIP Gateway, Domain=FQDN of OCSMediation Server (or IP), Username=Anonymous,Password=BLANK, Outbound Proxy=sip:FQDN of OCSMediation Server:5060;transport=tcp (example: sip:jb-ocs-md.ocsdemo.net:5060;transport=tcp), Assume that call comes from user=primary name of an existing pbxnsip-account (Type=extension) which will be charged for calls from OCS-Med-Server to the real world. This account must not be in use / registered.
  • Create or edit a dialplan, Pref=100, Trunk=TRUNK to SIP-provider, Pattern=*, Replacement=* or 0* or something like this (depends on your enviroment)

@OCS:

  • Open the OCS-MMC, Select Forest Properties --> Voice Properties, Location Profile Klick Add..., ProfileName=TypeOne, Description=TypeOne, Normalization Rules click Add..., RuleName=TypeOne, Description=TypeOne, Phone pattern regular expression= ^(\d*)$ Translation pattern regular expression= $1 (These pattern worked for me, maybe you need others! Test and Play with it or ask a .NET developer how it works. :mellow: This phone pattern was ok too: ^\+?(\d*)$ ) Click OK twice
  • Switch to Phone Usages, you should find here: Name=Default Usage , Description=Sample phone usage, if not click Add..., and Add it
  • Switch to Policy, activate your preffered Global policy setting, Defined Policies: you should find here: Name=Default Policy, click Edit, check Allow simultanous ringing of phones and check if the phone usage is configured Default Usage, If not click Add..., Add a policy and configure the usage.
  • Switch to Routes: click add..., Name=TypeOne (example: OCS to PBX via MEDIATION), Description=TypeOne, Target regular expression= ^(\d*)$ (it works for me, maybe you need others! Test and Play with it or ask a .NET developer how it works.), ;) Gateways click add, Your Mediation Server(s) should be listed here. Select one and OK, Phone usages click Configure... See if the Default is configured. OK twice. One OK more and you leave Voice Properties. Check yor voice settings again by clicking Voice beside the Status informations.
  • Select Server/Pool Properties: Frontend Properties - Voice check if the Location Profile is set. Maybe configure phone lock. OK
  • Switch to Users Properties/additional Options: Make sure that your ocs-activated-user's are enabled for Enterprise Voice (not pbx-integration) and that the Line URI is filled corresponding to the user. (example: tel:+303983xxxx) This depends on your enviroment! Especially how the called number was send to the ocs mediation server.

@OCS Mediation Server:

  • Select the Mediation Server Properties, General, If you use 1 NIC/ 1IP in OCS Medation Server (not supported, but it works) the Communications Server listening IP address and the gateway list. IP address are the same. If you use 2 IPs select the corresponding one. (Select the A/V Edge Server if you deployed one, not required). Select your default location profile and change te media port range if you want. For the first tests, use the default settings.
  • Switch to Next Hop Connections, Select the FQDN of your OCS Server/Pool, Port:5061 (default). PSTN Gateway net hop IP address, enter the pbxnsip IP address. Port:5060

@OC client:

  • start communicator, login with an enterprise enabled user and type a phone number, start the call.

Hope this helps you with the first steps. Maybe a I can add some screenshots later.

 

best regards,

 

Jan

 

 

Jan Boguslawski

Consultant IT Infrastructure

MCSE

 

Telefon: +49 30 399 784-0

 

ITaCS GmbH

Friedrichstra?e 121

10117 Berlin

www.itacs.de

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Hi again,

 

I made everything as exact as you describe.

 

And the call comes to the pbxsnip.

 

But it logs.

[5] 2007/11/02 11:25:42: SIP port accept from 10.0.254.15:2733

[5] 2007/11/02 11:25:43: Received incoming call without trunk information and user has not been found

 

Maybe you have another hint?

 

Many thx!!!

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Hi,

 

what pbxnsip version are you using? Please recheck: Assume that call comes from user= and (post your setting). I recommend using an unused extension like 9999 but create a corresponding pbxnsip-account type=extension before.

 

PS: keine Ursache :mellow:

 

best regards

 

Jan

 

 

Jan Boguslawski

Consultant IT Infrastructure

MCSE

 

Telefon: +49 30 399 784-0

 

ITaCS GmbH

Friedrichstra?e 121

10117 Berlin

www.itacs.de

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Hi,

 

pbxsnip Version: 2.1.0.2115 (Win32)

 

Assume that call comes from user= 491805835684540

 

491805835684540 is Primary Name of an unused Account

 

I also enhanced log to 0 and get this...

 

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.0.254.15:3648;branch=z9hG4bKaecd8d52

From: <sip:j.suenram@pc-soft.info>;epid=848AEC7FF1;tag=6228d44daf

To: <sip:04445950215@10.0.254.15;user=phone>;tag=9f31276645

Call-ID: 7d7a1e66-a33e-4650-9766-4e965f1a7c00

CSeq: 24 INVITE

Content-Length: 0

 

 

[7] 2007/11/02 16:29:20: Set packet length to 20

[6] 2007/11/02 16:29:20: Sending RTP for 7d7a1e66-a33e-4650-9766-4e965f1a7c00#9f31276645 to 10.0.254.15:61744

[5] 2007/11/02 16:29:20: Received incoming call without trunk information and user has not been found

[7] 2007/11/02 16:29:20: Set packet length to 20

[9] 2007/11/02 16:29:20: Resolve destination 540: tcp 10.0.254.15 3648

[7] 2007/11/02 16:29:20: SIP Tx tcp:10.0.254.15:3648:

SIP/2.0 404 Not Found

Via: SIP/2.0/TCP 10.0.254.15:3648;branch=z9hG4bKaecd8d52

From: <sip:j.suenram@pc-soft.info>;epid=848AEC7FF1;tag=6228d44daf

To: <sip:04445950215@10.0.254.15;user=phone>;tag=9f31276645

Call-ID: 7d7a1e66-a33e-4650-9766-4e965f1a7c00

CSeq: 24 INVITE

Contact: <sip:04445950215@127.0.0.1:5065;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.0.2115

Content-Length: 0

 

 

[9] 2007/11/02 16:29:20: Resolve destination 541: tcp 10.0.254.15 3648

[7] 2007/11/02 16:29:20: SIP Tx tcp:10.0.254.15:3648:

SIP/2.0 404 Not Found

Via: SIP/2.0/TCP 10.0.254.15:3648;branch=z9hG4bKaecd8d52

From: <sip:j.suenram@pc-soft.info>;epid=848AEC7FF1;tag=6228d44daf

To: <sip:04445950215@10.0.254.15;user=phone>;tag=9f31276645

Call-ID: 7d7a1e66-a33e-4650-9766-4e965f1a7c00

CSeq: 24 INVITE

Contact: <sip:04445950215@127.0.0.1:5065;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.0.2115

Content-Length: 0

 

 

[7] 2007/11/02 16:29:20: SIP Rx tcp:10.0.254.15:3648:

ACK sip:04445950215@10.0.254.15;user=phone SIP/2.0

FROM: <sip:j.suenram@pc-soft.info>;tag=6228d44daf;epid=848AEC7FF1

TO: <sip:04445950215@10.0.254.15;user=phone>;tag=9f31276645

CSEQ: 24 ACK

CALL-ID: 7d7a1e66-a33e-4650-9766-4e965f1a7c00

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 10.0.254.15:3648;branch=z9hG4bKaecd8d52

CONTENT-LENGTH: 0

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Hi, please tell me more about your setup. Are you running OCS Mediation Server and pbxnsip on separated machines or on one?

In your trace:

 

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.0.254.15

This must be the Medations Servers IP address

 

To: <sip:04445950215@10.0.254.15;user=phone>;tag=9f31276645

This must be the pbxnsip IP adress

 

In your case they are the same !!! You can try to add an IP and run OCS Med and pbxnsip on separeted IP's , but from my experience it wont work. Reason is the windows network behavior, preffering the lowest IP on one machine. Maybe this is the reason for this: Contact: <sip:04445950215@127.0.0.1:5065;transport=tcp>

in your trace.

 

btw: I had the same effect: Received incoming call without trunk information and user has not been found

when running pbxnsip and the Dialogic/Eicon Diva Server (a SIP-PSTN Gateway application) on one machine.

 

Please try running it on separated machines and if you want it the "Microsoft best practice way" then put two network cards in the Mediation Server. One directs to pbxnsip and one to OCS SE or a Enterprise pool.

 

Have a nive weekend!

 

Jan

 

 

Jan Boguslawski

Consultant IT Infrastructure

MCSE

 

Telefon: +49 30 399 784-0

 

ITaCS GmbH

Friedrichstra?e 121

10117 Berlin

www.itacs.de

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Ok finaly I got it.

 

Many thx again.

 

Transfered pbxnsip to another machine, recreated config with localhost domain and at least it does what it should. :-)

 

Many many thx.

 

One last question. Is it somehow possible to create multiple SIP-Register-Accounts and use the tel:URI in OCS-Users to bring a communicator call out via pbxnsip on a specific account?

 

Tried to create Accounts with the Primary name same as the TEL:URI. But I only get an access denied error. I think it is because SIP-From changes to internal Domain-Name when disabling "Assume that call comes from" to nothing.

 

Log about this

 

INVITE sip:01724025362@10.0.254.4;user=phone SIP/2.0

FROM: <sip:+491805835684540@ops.internet.pc-soft.info;user=phone>;epid=D446D4C154;tag=3b5cd8562

TO: <sip:01724025362@10.0.254.4;user=phone>

CSEQ: 38 INVITE

CALL-ID: 6b082cfa-bc4c-4d44-846a-749741eef94a

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 10.0.254.14:4859;branch=z9hG4bK46b5b52

CONTACT: <sip:ops.internet.pc-soft.info:5060;transport=Tcp;maddr=10.0.254.14;ms-opaque=e6946a50e9b9afc2>

CONTENT-LENGTH: 299

SUPPORTED: 100rel

USER-AGENT: RTCC/3.0.0.0 MediationServer

CONTENT-TYPE: application/sdp; charset=utf-8

ALLOW: UPDATE

ALLOW: Ack, Cancel, Bye,Invite

 

v=0

o=- 0 0 IN IP4 10.0.254.14

s=session

c=IN IP4 10.0.254.14

b=CT:1000

t=0 0

m=audio 63216 RTP/AVP 97 101 0 8

c=IN IP4 10.0.254.14

a=rtcp:63217

a=label:Audio

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=ptime:20

 

[7] 2007/11/05 15:53:05: UDP: Opening socket on port 64056

[7] 2007/11/05 15:53:05: UDP: Opening socket on port 64057

[5] 2007/11/05 15:53:05: Identify trunk (IP address and domain match) 6

[9] 2007/11/05 15:53:05: Resolve destination 47: tcp 10.0.254.14 4859

[7] 2007/11/05 15:53:05: SIP Tx tcp:10.0.254.14:4859:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.0.254.14:4859;branch=z9hG4bK46b5b52

From: <sip:+491805835684540@ops.internet.pc-soft.info;user=phone>;epid=D446D4C154;tag=3b5cd8562

To: <sip:01724025362@10.0.254.4;user=phone>;tag=add65be01d

Call-ID: 6b082cfa-bc4c-4d44-846a-749741eef94a

CSeq: 38 INVITE

Content-Length: 0

 

 

[9] 2007/11/05 15:53:05: Resolve destination 48: tcp 10.0.254.14 4859

[7] 2007/11/05 15:53:05: SIP Tx tcp:10.0.254.14:4859:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/TCP 10.0.254.14:4859;branch=z9hG4bK46b5b52

From: <sip:+491805835684540@ops.internet.pc-soft.info;user=phone>;epid=D446D4C154;tag=3b5cd8562

To: <sip:01724025362@10.0.254.4;user=phone>;tag=add65be01d

Call-ID: 6b082cfa-bc4c-4d44-846a-749741eef94a

CSeq: 38 INVITE

User-Agent: pbxnsip-PBX/2.1.0.2115

WWW-Authenticate: Digest realm="ops.internet.pc-soft.info",nonce="8000590fc939980dd38f090b01ca7883",domain="sip:01724025362@10.0.254.4;user=phone",algorithm=MD5

Content-Length: 0

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Hello again, pbxnsip does now everything as expected. Thx. And we will buy a license after evaluation is over.

 

Maybe someone has a hint on a more OCS2007 specific problem.

 

Now when we call a number in Communicator it rings two times before the real target-number phone rings. Then upon the first real ring at target phone communicator cancels with an error, that it does not received any audio from "number"....

 

Looks like Communicator does not wait long enough for establishment of the real call?

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Maybe someone has a hint on a more OCS2007 specific problem.

 

Now when we call a number in Communicator it rings two times before the real target-number phone rings. Then upon the first real ring at target phone communicator cancels with an error, that it does not received any audio from "number"....

 

Looks like Communicator does not wait long enough for establishment of the real call?

 

Foget this.... it was more a Firewallissue at a Branchoffice.

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  • 1 month later...

I have written a new pbxnsip Wiki page with a guide for pbxnsip-OCS2007 setup. Including helpful screenshots:

 

http://wiki.pbxnsip.com/index.php/Office_C...ications_Server

 

Best regards

 

Jan

 

 

Jan Boguslawski

Consultant IT Infrastructure

MCSE

 

Telefon: +49 30 399 784-18

ITaCS GmbH

Friedrichstrasse 121

10117 Berlin

http://www.itacs.de

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