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Cannot Make Inbound Calls through AudioCodes Media Gateway RRS feed

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    I have a Mediation Server and Media Gateway setup and I have no problem making outbound calls.

     

    The Media Gateway is connected directly to the PSTN via analog lines into the gateway, an AudioCodes MP118FXO. When I dial from a PSTN phone to a number that is a line attached to the gateway it answers and gives a second dial-tone waiting for me to enter in 3 digits which are supposed to be used to route the call. It accepts the 3 digits and then hangs up.

     

    I am pretty sure I am just missing something in the setup but I am very unclear on a few points:

     

    1.       How is an Office Communicator 2007 user associated to this inbound call? I have been assuming it is the 3 digits they enter somehow.

    2.       If so, where does it get the map of the 3 digits to the appropriate OC user? Is it required to normalize this to E.164 format?

    3.       If it does have to be normalized the 3 digits not represent or correspond to an actual phone number, just an extension, how do I normalize it to E.164 format if that is required?

     

    Are there any other steps I have likely missed in this configuration that may be preventing it from working?

     

    Is there a particular one of the OCS 2007 documents that discusses this topic in depth?

     

    Thank you in advance.

    Tuesday, November 27, 2007 1:32 AM

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    You should have a look in the OCS Voice Planning and Deployment Guide available here http://technet.microsoft.com/en-us/library/bb676082.aspx. It explains it all.

     

    Basically you need to define the phone number of the user in the msRTCSIP-Line attribute. We recommend to always use E.164 format within OCS, so the msRTCSIP-Line attribute should be in the form tel:+<E.164 number>. On the media gateway you should normalize the numbers to E.164 before sending them to the Mediation Server. If you can't do that the Mediation Server will ask the OCS server to normalize them using the location profile defined on the Mediation Server.

     

    best regards

       Jens

     

    Tuesday, November 27, 2007 7:33 AM
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    Jens, thank you for the info. A few more questions then:

     

    1. The link in your post doesn't seem to work. Got another link or a title or something I can search for?
    2. How do I configure the msRTCSIP? It says here: http://technet.microsoft.com/en-us/library/bb803644.aspx that I can use the ADUC but I don't see this setting so either I don't have something loaded properly or I need to use a different tool. What I am using is the ADUC on the OCS front end.
    3. I am assuming I write the normalization rule on the Media Gateway to normal the digits that they enter after it picks up so if they key in 100 I normalize that to +15555550100 (or something like that) and have the msRTCSIP-Line attribute set to tel:+15555550100 correct?

    Thank you again in advance.

    Tuesday, November 27, 2007 4:23 PM
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    OK, skip question 2. I was able to use adsiedit to modify the msRTCSIP-Line attribute. I setup a rule in the gateway to normalize the 1000 I typed in after it answered to +15555551000 and the value I put in for the msRTCSIP-Line was tel:+15555551000. So far I still don't have any luck with inbound calls but I think the setup is closer than before.

     

    Any more thoughts anyone?

     

    Thanks again in advance.

    Tuesday, November 27, 2007 6:01 PM
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     wjvii580671 wrote:

    The link in your post doesn't seem to work. Got another link or a title or something I can search for?

     

    Here's a link directly to the guide.

    http://www.microsoft.com/downloads/details.aspx?familyid=24e72dac-2b26-4f43-bba2-60488f2aca8d&displaylang=en

     

    And the OCS 2007 TechNet Library:

    http://technet.microsoft.com/en-us/library/bb676082.aspx

    Wednesday, November 28, 2007 5:54 AM
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    The main missing pieces for me were

     

    1. Setup the Tel to IP Routing. This was not a step covered in the AudioCodes OCS 2007 guide. For anyone else that has this issue just make sure to go to the Protocol Management, Routing Tables, Tel to IP Routing an enter the Media Gateway facing IP addresses of your Mediation Server into the routing table. In may case it was * for destination phone prefix, * for source phone prefix, IP address of mediation server and 0 for profile ID.

     

    2. Making sure that the Source Phone Number Manipulation Table for Tel to IP Calls needed to add a prefix of +1 so that OCS could handle the call. Go to Protocol Management, Manipulation Tables, Source Tel to IP.

     

    Also just a note, you want to be using firmware 5.20A.026.007 or later for the unit to be OCS compatible according to what I was told.

    Sunday, December 2, 2007 6:42 PM
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    Sorry about the wrong link, I can see another poster provided the correct link, thanks.

     

    Ad 2) Yes, should be able to use ADUC. Right-click on a user, properties, Communications tab, Configure, Enable Enterprise Voice and then fill in Line URI. Line URI in the UI is the msRTCSIP-Line attribute in AD.

     

    Ad 3) Yes, you need to have E.164 numbers flowing inside OCS. So the the media gateway should try to normalize inbound numbers to E.164. On outbound calls it need to strip + and digits from E.164 to the local convention before it puts them on the wire.

     

    best regards

       Jens

    Monday, December 3, 2007 9:26 AM
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    I am also having problems with Audiocodes gateway normalizing numbers.

     

    My PBX does not support DID. So my end user's URI look like this: tel:+1xxxxxxxxxx;ext=123

     

    I am trying to deploy OCS with PBX intergration. The Gateway I am using is Mediant 2000.

     

    If I call use from OC, the call is also forked to Gateway. From the Gateway log I can see the number called is :

     

     <sip:+1514xxxxxxx@mediantserver.mydomain.biz;user=phone>;epid=C7BFAE3090;tag=ebf12afc2<013><010>To: <sip:+1514xxxxxxx;ext=123@10.10.100.10;user=phone>;tag=1c320725806<013><010>Call-ID: f91c2aad-e959-45e1-969a-c840bdc73e8a<013><010>CSeq: 197 INVITE<013><010>Supported: em,timer,replaces,path,resource-priority<013><010>Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE<013><010>Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.00A.035.003<013><010>Reason: Q.850 ;cause=1<013><010>Content-Length: 0

     

    The problem is that the gateway does not know how to manipulate +1514xxxxxxx;ext=123.

    I need 123 be send to PBX box so that it can be routed to the desk phone.

     

    I have also tried to put extension numbers into "phone-office-other" field. This does not work since OCS does not call this number.

     

    How should I resoved this problem?

     

     

    Saturday, May 10, 2008 1:00 AM