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OCS 2007 R2 problems adding pstn users via sip trunk to pc audio conference

Posted on 2010-01-10
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Dear all,

We're using OCS 2007 R2 on Server 2008, we have standard edition servers installed for front end, an edge server and a mediation server. All users are using the R2 client, we do not have pbx integration.

Our mediation server has only one nic.

We have enterprise voice enabled with a sip trunk from interouteone, it seems to work very well after some initial issues with a Checkpoint firewalls interpretation of the call setup procedure. We can dial out and voice quality is great. Our problems come when establishing conferences between pc audio clients and pstn clients.

If i call a PSTN user first, i can subsequently succesfully, add internal, communicator/pc audio clients to a conference with the pstn user.

If i call a communicator/pc audio client first i cannot subsequently add a pstn user to the conference, when i do this i receive "An error occurred during the call. More Details (ID:404) in communicator

If i snoop the S4 logs on the mediation server I see

SIP/2.0 404 Not Found both in and outbound.

and:

Ms-diagnostics: 10404;source="mymediationserver.atmycompany.net";reason="Gateway returned a SIP failure code";component="MediationServer";SipResponseCode="404";SipResponseText="Not Found";GatewayFqdn="89.202.*.*"

My interpretation of this is that this is an error from my SIP trunk providers gateway? However i find that difficult to believe as they provide this service to hundreds of companies and i expect the issue is more down to my misconfiguration.....

Is this due to a difference in the way conference calls are dialled as compared to normal calls? Perhaps a number translation issue?

If i examine my communicator logs i see this when i try to add the external pstn user to the conference already in progress: SIP/2.0 481 Call Leg/Transaction Does Not Exist.

I suspect this is closely related as sounds very similar: http://social.microsoft.com/Forums/en-US/communicationsservertelephony/thread/3583bae8-ef28-41d8-9108-66e0df6bc814?prof=required but i don't have the first clue regarding adding number translations for conference calls, my current translation is the one i was instructed to set up by interoute.

Any guidance appreciated.

Best,

Jim.
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Question by:apcoexch
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10 Comments
 
LVL 33

Assisted Solution

by:Busbar
Busbar earned 1600 total points
ID: 26280828
Well, as usuall do you have the latest hotfixes applied specially those for november 2009 ones, they have a lot of fixes specially for mediation.

I feel that OCS cannot route the call as he tries to call the pstn number internally, can you tell how do you add a pstn user to the conf.
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Author Comment

by:apcoexch
ID: 26281053
Hi there,

That would figure i think.

I applied the mediation server patch, and all outbound calls stopped working (i presume because my production server isn't up to date with the November patches, our mediation server isnt in production yet). I'll apply these this evening to our front end server and test.

Once the patch was installed i saw the following errors when snooping "SIP/2.0 503 MediationCall SetupCall error: Could not load type 'Microsoft.Rtc.Internal.Signaling.SignalingSessionExtensions' from assembly 'Microsoft.Rtc.Collaboration, Version=3.5.0.0, Culture=neutral". When i reverted back to before the patch the errors went away and direct outbound dialling was possible again.

When i dial someone from scratch i simply enter their phone number with 00 or + in front of it, when adding a pstn user to a conference, i drag the dialled number into the conference window.

Will report back later, thanks for the guidance.

Jim.
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LVL 6

Assisted Solution

by:MohammadSaeed
MohammadSaeed earned 400 total points
ID: 26282804
did you try to invite users by typing the number, instead of drag and drop it to the call ??
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Author Comment

by:apcoexch
ID: 26289657
Hi there,

Ok both servers now updated with the November updates, mediation and front-end.

Mohammad, I tried typing the number and establishing the call before dragging the pstn user into the pc audio conference, same issue

Outbound dialling continues to work, and i can still add internal pc audio users to a pstn call.

If i snoop S4 and MediationServer on the Mediation Server I see a number of failures when dragging pstn callers into a pc audio call:

In:TL_INFO(TF_PROTOCOL) [0]0F04.0EFC::01/12/2010-01:56:44.277.00000adf (S4,SipMessage.DataLoggingHelper:sipmessage.cs(581))
<<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_17501DB>], 192.168.**.**:49350<-89.202.**.*:5060
SIP/2.0 404 Not Found
FROM: <sip:jbullock@mydomain.com;gruu;opaque=app:conf:audio-video:id:6A960F54E6063B47A8C16B372C2B4479>;tag=966b6881b;epid=02E06364A3
TO: <sip:+852926005**@89.202.**.*;user=phone>;tag=4bcca59-13c4-4b4bd6db-a4b2fd97-38ba4076
CSEQ: 6 INVITE
CALL-ID: 59157e76-13d7-481e-9f93-529224015fb1
VIA: SIP/2.0/TCP 192.168.***.**:49350;branch=z9hG4bK38755a93
CONTENT-LENGTH: 0

Out:Ms-diagnostics: 10404;source="mymediationserver.net";reason="Gateway returned a SIP failure code";component="MediationServer";SipResponseCode="404";SipResponseText="Not Found";GatewayFqdn="89.202.**.*"

Thanks for your help so far.

Cheers,

Jim



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LVL 33

Assisted Solution

by:Busbar
Busbar earned 1600 total points
ID: 26300165
can you tell what happens when you dial :+852926005 alone?
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Author Comment

by:apcoexch
ID: 26300170
works perfectly. (i removed last two digits as public posting)
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LVL 33

Assisted Solution

by:Busbar
Busbar earned 1600 total points
ID: 26300285
odd,
have you tried to create a confrence with the OCS user before adding the PSTN no. any erros in the OCS or the mediation
0
 

Author Comment

by:apcoexch
ID: 26300307
yes it's that thats the problem, i cannot subsequently add pstn users to an ocs user conference, but i can add ocs users if i START the call with the pstn user and drag them in afterwards.

there are no errors in the event logs, all i see is what i find with snooper as outlined above.

Jim.
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LVL 33

Assisted Solution

by:Busbar
Busbar earned 1600 total points
ID: 26300443
0
 

Accepted Solution

by:
apcoexch earned 0 total points
ID: 26423857
HI there, it appears that it's not a problem on our side at all, but the gateway providers pre-paid platform will not allow conferencing with pstn calls, it's a fault rather than a disabled feature. We are moving to post paid and that will apparently resolve this.

Cheers for the help and suggestions,

Jim
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