Hey all,
Alright. So we decided to not go with Avaya for our next PBX and we are now
full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP
gateway and call center and Lync is our internal UC and IP-PBX server. I've
already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all
is beautiful (except for the Opt11 not receiving names from * but that's
another topic). So, my problem now is with the call center.
This setup may be a bit convoluted at first but it'll make sense I hope. I've
created the queues in Asterisk via FreePBX. I then created a ring group for
each Lync extension so we get the "Confirm Calls" option and dodge the voice
mail problem. The agents the login via their Lync phone with the Ring Group
extension as their Agent ID. It kind of looks like this:
Queue 2001
Agent 4001
Agent 4002
Agent 4003
Ring Group 4001 -> Lync Extention 5001
Ring Group 4002 -> Lync Extention 5002
Ring Group 4003 -> Lync Extention 5003
This all works beautifuly! The problem I have is on transfers. If Lync
extension 5001 trasnfers to Lync extension 5010, Asterisk is unaware of the
transfer and shows that 5001 is still active with the call. We're using
OrderlyStats to monitor the queue so I watch the "Talking" counter just keep
counting instead of being aware the transfer took place. Now to me, that says
to me that the transfer took place within Lync so Asterisk is unaware of the
transfer. So my next step was to enable Refer support in Lync so Lync sends the
refer message back to Asterisk to transfer the call so Asterisk is fully aware
of what's going on. It seems like the refer message is trying to work and Lync
is sending it and Asterisk is receiving it but the "Refer-To" is changing
between the two so I'm at a loss.
(Logs are below signature)
Lync says it's sending the following message with a "Refer-to:
<sip:[email protected]>"
Asterisk is seeing the following and the refer-to changed, it's now "REFER-TO:
<sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2787?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto-tag%3D8be38bb187>".
At first it seems like Lync is sending a true SIP URI so I need to get Asterisk
to know how to handle that SIP URI and then secondly, it seems like Asterisk
doesn't even receive the same REFER-TO message that Lync sent. Is this because
Asterisk doesn't know how to handle the SIP URI?
So I guess I'm left with wondering if fixing the REFER message stuff is going
to fix my problem even? The end goal is for Asterisk to be aware that a call
was transferred to another extension in Lync.
Thanks in advance everyone!
Louis
================================= Begin Lync SIP message
============================================
TL_INFO(TF_PROTOCOL) [0]0B10.1E88::03/04/2011-13:21:17.501.0004fcd9
(SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record
Trace-Correlation-Id: 215606761
Instance-Id: 00011F02
Direction: outgoing
Peer: lyncserver.internal.domain:5070
Message-Type: request
Start-Line: REFER
sip:lyncserver.internal.domain:5070;grid=ed392a6bc0344a30b0841cd69be137ed
SIP/2.0
From: ""
<sip:1173;[email protected];user=phone>;epid=e9688aa93e;tag=8be38bb187
To:
<sip:500;[email protected];user=phone>;epid=B3E26C1E76;tag=9227b8a39d
CSeq: 2 REFER
Call-ID: aa6f8871-4151-4149-ad5a-29ab941bf4d0
Via: SIP/2.0/TLS
20.20.20.20:54166;branch=z9hG4bKEB39D72C.F05E7E34CF9EF4FD;branched=FALSE
Max-Forwards: 69
Via: SIP/2.0/TLS 172.16.2.29:53851;ms-received-port=53851;ms-received-cid=400
User-Agent: CPE/4.0.7577.107 OCPhone/4.0.7577.107 (Microsoft Lync 2010 Phone
Edition)
Supported: ms-dialog-route-set-update
Refer-to: <sip:[email protected]>
Referred-By:
<sip:[email protected]>;ms-referee-uri="sip:500;[email protected];user=phone";ms-identity="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:Fri,
04 Mar 2011 13:21:17
GMT";ms-identity-info="sip:Lyncserver.internal.domain:5061;transport=tls";ms-identity-alg=rsa-sha1
Content-Length: 0
P-Asserted-Identity: <sip:[email protected]>
Privacy: id
Message-Body: -
$$end_record
================================= End Lync SIP message
============================================
================================= Begin Asterisk Debug
============================================
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 0 [ 53]: REFER
sip:[email protected]:5067;transport=TLS SIP/2.0
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 1 [ 78]: FROM:
<sip:[email protected]:5067>;epid=431D53633D;tag=42b6d8c72b
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 2 [ 46]: TO:
<sip:[email protected]:5067>;tag=as0d823373
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 3 [ 13]: CSEQ: 2 REFER
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 4 [ 59]: CALL-ID:
[email protected]:5067
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 6 [ 59]: VIA: SIP/2.0/TLS
20.20.20.20:5067;branch=z9hG4bK4614ad68
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 7 [ 86]: CONTACT:
<sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2787>
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 8 [ 17]: CONTENT-LENGTH: 0
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 9 [179]: REFER-TO:
<sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2787?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto-tag%3D8be38bb187>
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 10 [ 40]: USER-AGENT:
RTCC/4.0.0.0 MediationServer
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 11 [ 0]:
[Mar 4 08:21:05] VERBOSE[18506] chan_sip.c:
<--- SIP read from TLS:20.20.20.20:5067 --->
REFER sip:[email protected]:5067;transport=TLS SIP/2.0
FROM: <sip:[email protected]:5067>;epid=431D53633D;tag=42b6d8c72b
TO: <sip:[email protected]:5067>;tag=as0d823373
CSEQ: 2 REFER
CALL-ID: [email protected]:5067
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 20.20.20.20:5067;branch=z9hG4bK4614ad68
CONTACT:
<sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2787>
CONTENT-LENGTH: 0
REFER-TO:
<sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2787?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto-tag%3D8be38bb187>
USER-AGENT: RTCC/4.0.0.0 MediationServer
<------------->
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 0 [ 53]: REFER
sip:[email protected]:5067;transport=TLS SIP/2.0
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 1 [ 78]: FROM:
<sip:[email protected]:5067>;epid=431D53633D;tag=42b6d8c72b
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 2 [ 46]: TO:
<sip:[email protected]:5067>;tag=as0d823373
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 3 [ 13]: CSEQ: 2 REFER
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 4 [ 59]: CALL-ID:
[email protected]:5067
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 6 [ 59]: VIA: SIP/2.0/TLS
20.20.20.20:5067;branch=z9hG4bK4614ad68
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 7 [ 86]: CONTACT:
<sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2787>
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 8 [ 17]: CONTENT-LENGTH: 0
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 9 [179]: REFER-TO:
<sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2787?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto-tag%3D8be38bb187>
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Header 10 [ 40]: USER-AGENT:
RTCC/4.0.0.0 MediationServer
[Mar 4 08:21:05] VERBOSE[18506] chan_sip.c: --- (11 headers 0 lines) ---
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: = Looking for Call ID:
[email protected]:5067 (Checking From) --From tag
42b6d8c72b --To-tag as0d823373
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: **** Received REFER (9) - Command in
SIP REFER
[Mar 4 08:21:05] VERBOSE[18506] chan_sip.c: Call
[email protected]:5067 got a SIP call transfer from
caller: (REFER)!
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Attended transfer: Will use
Replace-Call-ID : aa6f8871-4151-4149-ad5a-29ab941bf4d0 F-tag: 9227b8a39d T-tag:
8be38bb187
[Mar 4 08:21:05] VERBOSE[18506] chan_sip.c: SIP transfer to extension
Lyncserver.internal.domain:5067@from-internal-xfer by (null)
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: SIP attended transfer: Transferer
channel SIP/Lync-00000003, transferee channel DAHDI/i1/500-2
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Got SIP transfer, applying to
bridged peer 'DAHDI/i1/500-2'
[Mar 4 08:21:05] DEBUG[18502] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: DAHDI/i1/500-2
Variable: SIPREFERRINGCONTEXT
Value: from-internal
Uniqueid: 1299244801.4
[Mar 4 08:21:05] DEBUG[18502] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: DAHDI/i1/500-2
Variable: SIPREFERREDBYHDR
Value:
Uniqueid: 1299244801.4
[Mar 4 08:21:05] DEBUG[18497] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: DAHDI/i1/500-2
Variable: SIPREFERRINGCONTEXT
Value: from-internal
Uniqueid: 1299244801.4
[Mar 4 08:21:05] DEBUG[18497] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: DAHDI/i1/500-2
Variable: SIPREFERREDBYHDR
Value:
Uniqueid: 1299244801.4
[Mar 4 08:21:05] VERBOSE[18506] chan_sip.c:
<--- Transmitting (no NAT) to 20.20.20.20:5067 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/TLS 20.20.20.20:5067;branch=z9hG4bK4614ad68;received=20.20.20.20
From: <sip:[email protected]:5067>;epid=431D53633D;tag=42b6d8c72b
To: <sip:[email protected]:5067>;tag=as0d823373
Call-ID: [email protected]:5067
CSeq: 2 REFER
Server: FPBX-2.8.1(1.8.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5067;transport=TLS>
Content-Length: 0
<------------>
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Trying to put 'SIP/2.0 202' onto TLS
socket destined for 20.20.20.20:5067
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Looking for callid
aa6f8871-4151-4149-ad5a-29ab941bf4d0 (fromtag 9227b8a39d totag 8be38bb187)
[Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Strict routing enforced for session
[email protected]:5067
[Mar 4 08:21:05] VERBOSE[18506] chan_sip.c: set_destination: Parsing
<sip:Lyncserver.internal.domain:5067;transport=Tls> for address/port to send to
[Mar 4 08:21:05] DEBUG[18506] netsock2.c: Splitting
'Lyncserver.internal.domain:5067' gives...
[Mar 4 08:21:05] DEBUG[18506] netsock2.c: ...host 'Lyncserver.internal.domain'
and port '5067'.
[Mar 4 08:21:05] VERBOSE[18506] chan_sip.c: set_destination: set destination
to 20.20.20.20:5067
[Mar 4 08:21:05] VERBOSE[18506] chan_sip.c: Reliably Transmitting (no NAT) to
20.20.20.20:5067:
NOTIFY sip:Lyncserver.internal.domain:5067;transport=Tls SIP/2.0
Via: SIP/2.0/TLS 10.10.10.10:5067;branch=z9hG4bK05f81334
Max-Forwards: 70
From: <sip:[email protected]:5067>;tag=as0d823373
To: <sip:[email protected]:5067>;epid=431D53633D;tag=42b6d8c72b
Contact: <sip:[email protected]:5067;transport=TLS>
Call-ID: [email protected]:5067
CSeq: 103 NOTIFY
User-Agent: FPBX-2.8.1(1.8.3)
Event: refer;id=2
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 49
SIP/2.0 481 Call leg/transaction does not exist
---
================================= End Asterisk Debug
============================================
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
