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Lync 2010 Server PSTN/Gateway ASTERISK / TRIXBOX for udp<--sip-->tcp

General discussion
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I have successfully configure on a few occasions now, a Lync 2010 with external access / UM integration
LYNC: Single Front End Server, Mediation Server
IP ADDRESS/s: 10.0.0.2
FQDN on instld. Cert: lync.domain.local
RP: Microsoft Forefront TMG - Reverse Proxy for external ABS PUBLIC/PRIVATE(DOMAIN.local) Interface
IP ADDRESS/s: eg, 252.252.252.252, 10.0.0.40
FQDN on instld. Cert: sipweb.domain.com
SIP: Edge Server for external IM + A/V
IP ADDRESS/s: 252.252.252.253, 10.0.0.103
FQDN on instld. Cert: sip.domain.com, sip.domain.local
TRIXBOX: (FROM: SIP@provider.com:UDP) -> (TO: +61@domain.com:TCP)
I have also set up Exchange 2010 SP1 w/ UM integration huntgroup/dialplan and UMIPGateway = LYNC
Lync 2010 i386 Client - Windows 7 External Client (Public access SRV *sip_tls/ % .domain.com )
External Dialing successful (through PSTN:/Gateway-TRIXBOX) Call forwarding also successful (problem with symultaneous ring = essentially working same as forward, lync client receive "XXXXXXX has answered, while the call rings and transfers successfully to external PSTN client )
Recieve External Call (from PSTN:/Gateway-TRIXBOX)
Address Book Synchronising
Voice Mail will download to Push-to-Play under Dial Tab in Lync 2010
Lync to Lync A/V/Conferencing/IM/File/Desktop/Sharing is all working
AUDIO Test Service works
Voice MAIL / UM Auto Attendant ---- Do Not Work
Information;
I have sniffed the ethernet interface on my laptop at home while connected automatically with public cert verification (External)
What i see when dialing call is STUN Traffic to 10.0.0.1 (Exchange2010Sp1, UM,CAS,HUB,MBS)
What do i need to do to enable Voice mail for external clients, making the Polycom CX700 that is connected / Updated to version 4.* (lync phone edition) and everything works, Calendar, Call history and Voice Mail List but 1 touch to play voice-mail not work from external. Why?
All of this has been on virtual hardware in a test environment running Xenserver 5.6 -
All Windows Servers (Edge,RP,SFE,Exch) VM 4096MB RAM - 24GB VHD 4VCPU Xeon 3.0GHz
Trixbox VM
- Edited by Jad Seifeddine Thursday, January 6, 2011 1:44 PM autoformat keep chaning font size
Thursday, January 6, 2011 1:42 PM
All replies
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What version of Trixbox are you running?Thursday, January 6, 2011 9:19 PM
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I am running trixbox 2.8.0.4 on xenserver virt guest with 2048mb and 15GBVHD with 2VCPU
The config is as easy as below, following installation of Trixbox with 2 interfaces (public and private)
once you have that set up in the trixbox conf and DNS is correct
trixboxserv.domain.com/maint
ontop menu
choose PBX_-->PBX Settings
then on left hand menu
choose Trunks
then from this page choose add + SIP Trunk
ALL DEFAULTS ----
until;
Trunk Name:provider
PEER Details:
allow=alaw&ulaw
canredirect=no
context=custom-get-did-from-sip
disallow=all
fromdomain=nsw.sydneyaus.net.au
host=nsw.sydneyaus.net.au
insecure=very
secret=my-sip-secret
type=peer
username=02XXXXXXXX
Incoming Settings
USER Context:02XXXXXXXX
USER Details:
canreinvite=no
context=custom-get-did-from-sip
qualify=no
secret=my-sip-secret
type=user
username=02XXXXXXXX
Registration
Register String:
02XXXXXXXX:my-sip-secret@nsw.sydneyaus.net.au/02XXXXXXXX
Now for the LYNC go click save, apply and go back to the trunks page to add another trunk
all default until
Outgoing Settings
Trunk Name:Connect-with-LYNC
PEER Details:
host=lync.domain.local
transport=tcp
port=5068
insecure=very
type=peer
fromdomain=lync.domain.com.au
context=from-Lync
promiscredir=yes
qualify=yes
canreinvite=yes
Incoming Settings
USER Context:from-LYNC
USER Details:
host=lync.domain.local (OR IP ADDRESS, Hostname requires host record /etc/hosts/)
transport=tcp
port=5068
insecure=very
type=peer
context=from-Lync
promiscredir=yes
qualify=yes
canreinvite=yes
Now you need to add dailin exten handlin rules in ASTERISK (behind trixbox) using the terminal preferably
ssh sipout.domain.local (TRIXBOX IP address added to DNS)cd /etc/asterisk/vi extensions_custom.conf
; I added the following for use my purpose - matching my dailing plan in LYNC
Incoming calls (from the world (PSTN) to LYNC ) note; This is the only rule for calls from [custom-get-did-from-sip]
[custom-get-did-from-sip]exten => _.,1,Noop(External Call coming in from PROVIDER!)exten => _.,n,Set(pseudodid=${SIP_HEADER(To)})exten => _.,n,Set(pseudodid=${CUT(pseudodid,@,1)})exten => _.,n,Set(pseudodid=${CUT(pseudodid,:,2)})exten => _.,n,Set(LYNC_Client=+61${pseudodid:1});exten => _.,n,Set(LYNC_Client=${pseudodid})exten => _.,n,Answerexten => _.,n,Dial(SIP/${LYNC_Client}@Connect-with-LYNC,,tr)
;If I normalize a number like 94811111 make it 0294811111 - note I have many rules like this to achieve complete dialing capabilities
[from-Lync]exten => _0.,1,Set(numDialled=${EXTEN})exten => _0.,n,Set(REALCALLERIDNUM=${CALLERID(number)})exten => _0.,n(start),NoOp(REALCALLERIDNUM is ${REALCALLERIDNUM})exten => _0.,n,Set(USEROUTCID=${CALLERID(number)})exten => _0.,n,Set(CALLERID(number)=0${USEROUTCID:3})exten => _0.,n,Set(TRUNKOUTCID=${OUTCID_${ARG1}})exten => _0.,n,Answerexten => _0.,n,Dial(SIP/${numDialled}@provider,,tr)
NOW ON YOUR Front End Server -
add sipout.hbi.local in your topology for pstn gateway using port 5060 TCPEverythings working perfectly, the phone has all call history and updated to lync phone edition as well as calendar and address book... funny thing is its sitting here re-siging in every 5-10 minutes like clock work... strange
Using the details above
Friday, January 7, 2011 3:01 PM -
I've been able to get to the same level of functionality, with the exception of dial in conferencing with 2.8.0.4, which is why I was curious. I can dial in from the outside to a Lync client, a Lync client can dial out. A meeting attendee can join a meeting and use the call me function and it works successfully. When someone tries to dial one of the published PSTN conference dial in numbers, it accepts the call, has them enter their information, says they are now being joined to the conference. I see the person popup as joined on my Lync client for maybe 2 seconds, then the attendant says they cannot be joined to this meeting. I just keep getting the same error in the tracing, no matter what I do:
TL_INFO(TF_PROTOCOL) [0]0A8C.2324::01/07/2011-21:08:51.397.0001d81f (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record
Trace-Correlation-Id: 3960482742
Instance-Id: 00032E07
Direction: incoming
Peer: invmalyncvm01.involtadc.local:5070
Message-Type: response
Start-Line: SIP/2.0 491 Invite with Replaces failed because Gateway side reinvite failed.
From: <sip:jward@involta.com;gruu;opaque=app:conf:audio-video:id:BGLLFH39>;tag=a76ecf72e5;epid=051387A305
To: <sip:invmalyncvm01.involtadc.local@involta.com;gruu;opaque=srvr:MediationServer:BFHgCdcQO1GRCR3WP3DGKAAA;grid=d721707d4d5941f4b5de18c653c23214>;epid=B19108E3F4;tag=66c420364a
CSeq: 65 INVITE
Call-ID: 49085289-6ee0-4a15-9c28-789160b6f6fe
VIA: SIP/2.0/TLS 10.128.10.57:52813;branch=z9hG4bKA1C4C9D9.25A66D77A9F5F212;branched=FALSE,SIP/2.0/TLS 10.128.10.57:52812;branch=z9hG4bKe71642d9;ms-received-port=52812;ms-received-cid=412800
CONTENT-LENGTH: 0
P-ASSERTED-IDENTITY: "U.S. CELLULAR"<sip:319XXXXXXX;phone-context=DefaultProfile@involta.com;user=phone>
SERVER: RTCC/4.0.0.0 MediationServer
ms-diagnostics: 10010;source="invmalyncvm01.involtadc.local";reason="Gateway side Media negotiation failed";component="MediationServer";SipResponseText="Invite with Replaces failed because Gateway side reinvite failed."
ms-diagnostics-public: 10010;reason="Gateway side Media negotiation failed";component="MediationServer";SipResponseText="Invite with Replaces failed because Gateway side reinvite failed."
ms-endpoint-location-data: NetworkScope;ms-media-location-type=intranet
Message-Body: –
$$end_record
In the mean time, I'll talk to our UM person and see if they can shed any light on the issue you're facing.
Friday, January 7, 2011 9:39 PM -
I'm not sure, but please check your trunk configuration -
at the moment for me this appears to be disabled as i have not ventured to that yet but take not of the below;
"
Incoming Settings
USER Context:02XXXXXXXX
USER Details:
canreinvite=no
"
Sunday, January 9, 2011 2:37 PM -
I am trying to connect Lyc to AsteriskNOW (FreePBX 2.8.1.4, Asterisk Version 1.6.2.17)
I am able to dial Lync extension from my Asterisk but can't dial back to Asterisk from Lync. Here is my Trunk configuration (Peer Details and Incoming Settings)
host=10.10.0.60
transport=tcp
port=5060
insecure=very
type=peer
fromdomain=mydomain.com
context=from-lync
promiscredir=yes
qualify=yes
canreinvite=yeshost=10.10.0.60
transport=tcp
port=5060
insecure=very
type=peer
context=from-Lync
promiscredir=yes
qualify=yes
canreinvite=yes
I am able to trace following error using Lync logging tool
Received status code 503 for request sip:817@10.10.0.7:5070;user=phone;maddr=lyncfe.identitymine.com
(0000000003248F2F)Gateway 10.10.0.7 being taken down. Time of shutdown:7/21/2011 11:57:08 AM; Timer set for:8
Gateway [10.10.0.7] failure - reason [503 Service Unavailable]
Next hop not found. Responding with 503.
An exception 'Microsoft.Rtc.OutboundRouting.OutboundRoutingException' was thrown at ' at Microsoft.Rtc.OutboundRouting.OutboundRoutingDispatcher.PrepareRequest(OutboundRoutingTransaction transaction)
at Microsoft.Rtc.OutboundRouting.OutboundRoutingDispatcher.ReRouteRequest(OutboundRoutingTransaction transaction, Response response, Boolean isCACFailure)'. Message 'Routes available, but no next hop available'Cannot reroute request: Routes available, but no next hop available
Any help will be much appreciated :)
Savi
Thursday, July 21, 2011 7:09 PM -
Asterisk by default uses port 5060 for SIP, (UDP or TCP).
This line from your Lync log shows 5070
"Received status code 503 for request sip:817@10.10.0.7:5070;user=phone;maddr=lyncfe.identitymine.com"
Double check the port number where your Mediaition server is sending to, and double check the port Asterisk is receiving on.
Also, by default, the Mediaition server receives on 5068, but if 5060 is working for you then it's all good.
Regards
Paul Adams
MS Lync Server 2010 Resource Kit writer | website: www.pauladamsit.comFriday, July 22, 2011 11:28 PM -
Paul,
I think this trace is from the Front End and port 5070 is correct, since "Port 5070 is used by the Mediation Server for incoming requests from the Front End Server to the Mediation Server.” i.e. in co-located scenario FE talks to itself on 5070.
A clean "Mediation Server" (no S4, no SIPStack must be examined.
Drago
http://www.lynclog.comSaturday, July 23, 2011 12:33 AM -
I was able to fix the dialing issues, here the steps that I documented http://savithomas.blogspot.com/2011/07/connecting-lync-server-2010-with.html
- Edited by Savi Thomas Monday, July 25, 2011 10:11 PM typo
Monday, July 25, 2011 10:11 PM