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Lync Direct SIP to Call Manager 8.x Can't dial in to conference

Question
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Hello,
I've seen lots of posts on the CallManager to Lync integration, must be something stupid I'm missing, but the problem is as follows: I can dial-out from Lync to PSTN through CallManager, we're not using enterprise voice only trying to enable some Dial-In numbers for conference. The dial in successfully routes to Lync, in logs I can see invite to sip:+xxxx@IP_of_lync_std
TL_INFO(TF_PROTOCOL) [1]0DC8.1868::03/31/2012-23:52:56.568.000079d5 (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record
Trace-Correlation-Id: 3549442660
Instance-Id: 00005992
Direction: incoming
Peer: IP_CallManager:42036
Message-Type: request
Start-Line: INVITE sip:+xxxxxxxx@IP_Lync_Std:5060 SIP/2.0
From: <sip:+xxxxxxxxxxxxx@IP_CallManager>;tag=683~ebf3e994-f07f-4b37-8d5c-9a6ca2ad1e1f-30156396
To: <sip:+xxxxxxxx@IP_Lync_Std>
CSeq: 101 INVITE
Call-ID: aad51300-f77198e7-1b-58017ac@IP_CallManager
Via: SIP/2.0/TCP IP_CallManager:5060;branch=z9hG4bK1a31288700
Date: Sat, 31 Mar 2012 23:53:11 GMT
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:IP_CallManager:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 2866090752-0000065536-0000000006-0092280748
Session-Expires: 1800
P-Asserted-Identity: <sip:+xxxxxxxxxxx@IP_CallManager>
Remote-Party-ID: <sip:+xxxxxxxxxxx@IP_CallManager>;party=calling;screen=yes;privacy=off
Contact: <sip:+xxxxxxxxxxx@IP_CallManager:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 317
Message-Body: v=0
o=CiscoSystemsCCM-SIP 683 1 IN IP4 IP_CallManager
s=SIP Call
c=IN IP4 IP_ISR
b=TIAS:64000
t=0 0
m=audio 23020 RTP/AVP 0 8 15 18 4
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:15 G728/8000
a=ptime:30
a=rtpmap:4 G723/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=ptime:20
then
TL_INFO(TF_DIAG) [1]0DC8.1868::03/31/2012-23:52:56.568.00007dd9 (SIPStack,SIPAdminLog::TraceDiagRecord:SIPAdminLog.cpp(147))$$begin_record
LogType: diagnostic
Severity: information
Text: Response successfully routed
SIP-Start-Line: SIP/2.0 401 Unauthorized
SIP-Call-ID: aad51300-f77198e7-1b-58017ac@IP_CallManager
SIP-CSeq: 101 INVITE
Peer: IP_CallManager:42036
Data: destination="Unknown"
$$end_recordand call fails with SIP/4
TL_INFO(TF_PROTOCOL) [1]0DC8.1868::03/31/2012-23:52:56.568.00007e0a (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record
Trace-Correlation-Id: 3549442660
Instance-Id: 00005993
Direction: outgoing;source="local"
Peer: IP_CallManager:42036
Message-Type: response
Start-Line: SIP/2.0 401 Unauthorized
From: <sip:+xxxx@IP_CallManager>;tag=683~ebf3e994-f07f-4b37-8d5c-9a6ca2ad1e1f-30156396
To: <sip:+Dial-In@IP_Lync_Std>;tag=9FC9B8C043A1EDC06E9B923A8A7AEB4C
CSeq: 101 INVITE
Call-ID: aad51300-f77198e7-1b-58017ac@IP_CallManager
Date: Sat, 31 Mar 2012 23:52:56 GMT
WWW-Authenticate: NTLM realm="SIP Communications Service", targetname="FQDN_Lync_Std", version=4
WWW-Authenticate: Kerberos realm="SIP Communications Service", targetname="sip/FQDN_Lync_Std", version=4
WWW-Authenticate: TLS-DSK realm="SIP Communications Service", targetname="FQDN_Lync_Std", version=4, sts-uri="https://FQDN_Lync_Std:443/CertProv/CertProvisioningService.svc"
Via: SIP/2.0/TCP IP_CallManager:5060;branch=z9hG4bK1a31288700;ms-received-port=42036;ms-received-cid=FD00
Server: RTC/4.0
Content-Length: 0
Message-Body: –
$$end_recordMy dial-in sip address is sip:+xxxx@sip_domain obviously, not sip:+xxxx@IP_Lync_Std and on Cisco (I'm not a CallManager expert though) I can only set destination as IP or server name "server.sipdomain" which is not my sip domain
many thanks in advance!
Monday, April 2, 2012 12:37 PM
Answers
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I have the exact same problem using CUCM 7x.
I started an almost identical thread and am seeing the same logs as you are when i do a trace
- Proposed as answer by Shawn Harry Monday, April 2, 2012 8:38 PM
- Unproposed as answer by Shawn Harry Monday, April 2, 2012 8:38 PM
- Edited by Shawn Harry Tuesday, April 3, 2012 6:40 AM
- Proposed as answer by Shawn Harry Tuesday, April 3, 2012 6:40 AM
- Marked as answer by fmamonov Tuesday, April 3, 2012 9:13 PM
Monday, April 2, 2012 4:58 PM
All replies
-
I have the exact same problem using CUCM 7x.
I started an almost identical thread and am seeing the same logs as you are when i do a trace
- Proposed as answer by Shawn Harry Monday, April 2, 2012 8:38 PM
- Unproposed as answer by Shawn Harry Monday, April 2, 2012 8:38 PM
- Edited by Shawn Harry Tuesday, April 3, 2012 6:40 AM
- Proposed as answer by Shawn Harry Tuesday, April 3, 2012 6:40 AM
- Marked as answer by fmamonov Tuesday, April 3, 2012 9:13 PM
Monday, April 2, 2012 4:58 PM -
Many thanks, reading the guide http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns784/1071259.pdf it definitely says to route to Mediation, but MS one said to 5060. It worked for me even without changing SipServerTcpPort to $null
- Edited by fmamonov Wednesday, April 4, 2012 11:28 AM
Wednesday, April 4, 2012 11:27 AM -
Mine was set to 5060 as that's the port that default TCP port SIP listens on. I guess MS only tested this on stand alone FE's that didn't have a Mediation server collocated on. Either way glad it worked out for you too.Thursday, April 5, 2012 9:36 AM