Route SIP REFER to SIP Trunk based on DN

Cisco UCM 9 is connected to a third-party PBX over SIP Trunk. Third-party PBX sends a SIP REFER message to Cisco UCM to call a DN on the third-party PBX. Cisco UCM responds with SIP 404 Not Found as it does not recognize the DN of the third-party PBX.
How do I configure Cisco Unified Communication Manager 9 to route this call back out over the SIP Trunk to the third-party PBX based on the DN (Not IP)?
Cisco UCM contains a route pattern 53xxx to route to SIP_Trunk_3rdParty.
Third-party PBX contains a SIP Proxy and Call Server. The call should route to the SIP Proxy IP. The SIP REFER contains "Refer-To" 53xxx@ThirdPartyCallServerIP
I added a SIP Route Pattern on CUCM to route calls for ThirdPartyCallServerIP to SIP_Trunk_3rdParty. This works in routing the call to ThirdPartyCallServerIP, however I need the call to route to 53xxx@ThirdPartySIPproxyIP for it to be successful.
Direct calls from CUCM to ThirdParty PBX 53XXX@ThirdPartySIPproxyIP are successful. SIP REFER coming into CUCM to request CUCM to call ThirdParty fail.
Any ideas on what configuration on CUCM I could try to get CUCM to route the call to thrid-party based on the SIP REFER?

Thanks for the reply Vivek.
Partitions:
- ThirdPartyPBX
- CiscoEndpoints
Calling Search Space: "ThirdParty_Cisco" contain both of the above partitions.
Route Pattern 531XX and 80965 are assigned to Route Partition "ThirdPartyPBX"
Cisco UCM Main site phones are in CSS "ThirdParty_Cisco" and DN is in Route Partition "CiscoEndpoints". DN is in CSS "ThirdParty_Cisco".
Trunk "SIP_Trunk_3rdParty" - Inbound and Outbound Calls are in CSS "ThirdParty_Cisco".
Trunk SIP information has "Rerouting CSS", "Out-of-Dialog Refer CSS", and Subscribe CSS as "ThirdParty_Cisco".
Cisco continues to respond to with SIP 404 not found. CUCM does not seem to match the SIP refer to the CSS or Route partition with with 531XX route pattern.
The SIP Refer is coming from DN 80965 over the SIP Trunk from the Third-party PBX.
Perhaps I'm missing something in my CSS config?
Any other method for CUCM to match SIP Refer to a Route Pattern?

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HTTP Authentication Digest for SIP messages in a trunk SIP CUCME

Hello,
we would like to implement HTTP Authentication Digest for SIP messages in a trunk SIP between a Cisco 2851 and an Asterisk server.
We are using CUCM Express with 15.1(4)M (CME 8.6) as voice gateway to connect to PSTN.
According to Cisco documentation:
"To configure a gateway to use HTTP Authentication Digest, give the following command in each dial peer or SIP-UA configuration mode:
authentication username username password password [realm realm]."
The problem is that when call is from CISCO to ASTERISK, Asterisk sends a challenge to Cisco to do Authentication:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.70.11:5060;branch=z9hG4bK3E205D
Remote-Party-ID: "DN1001" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "DN1001" <sip:[email protected]>;tag=5317D4-2271
To: <sip:[email protected]>
Date: Thu, 20 Feb 2014 10:55:56 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
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Cisco-Guid: 1679566433-2572423651-2156454406-1292596908
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1392893756
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 208
<--- Reliably Transmitting (no NAT) to 10.0.70.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.70.11:5060;branch=z9hG4bK3E205D;received=10.0.70.11
From: "DN1001" <sip:[email protected]>;tag=5317D4-2271
To: <sip:[email protected]>;tag=as665c9410
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="559bd1d2"
Content-Length: 0
However, when call is for ASTERISK to Cisco, there is no challenge sent.
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
Max-Forwards: 70
From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Thu, 20 Feb 2014 09:58:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
<--- SIP read from UDP:10.0.70.11:60829 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
To: <sip:[email protected]>
Date: Thu, 20 Feb 2014 10:58:27 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
To: <sip:[email protected]>;tag=556830-757
Date: Thu, 20 Feb 2014 10:58:27 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "DN1001" <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
My configuration in Cisco device is:
dial-peer voice 1 voip
description **Calls to ASTERISK **
destination-pattern 9T
session protocol sipv2
session target sip-server
codec g711ulaw
sip-ua
keepalive target ipv4:10.1.32.70
authentication username CCME password 7 070E234F4A realm asterisk
sip-server ipv4:10.1.32.70:5060
To avoid that the ASTERISK is blocked by Cisco TOLLFRAUD_APP I have added:
voice service voip
ip address trusted list
ipv4 10.1.32.70 255.255.255.255
allow-connections sip to sip
sip
registrar server
The issue is that I would like that Cisco also send a challenge to asterisk server to authenticate SIP messages.
Any ideas?.
Regards.

Hello,
yes, but credentials command configure credentials that are used when Cisco UA must register in a server.
I do not need register Cisco into Asterisk server. What I want is that Cisco authenticate SIP messages that receive. I know
that can be enough with TOLLFRAUD_AP where remote IP is checked, but I want to do something like others routing
protocols (as OSPF, BGP) where every message must be authenticated.
Thanks.
Regards.

Snom phones in secondary subnet unable to call out - SIP CANCEL in SIP log

I've been trying to diagnose this very strange problem we are having. All our servers and some SNOM phones are in the subnet 192.168.100.0, the main building. They all work fine. Phones located in two other buildings connected with high-speed fiber use subnets
192.168.1.0 and 192.168.200.0. They can receive calls but are unable to call out. This doesn't affect the Lync 2010 and 2013 desktop clients with enterprise voice...they work fine anywhere, even externally.
We are running Lync Server 2013 Standard Edition, with the latest updates applied. Mediation role is co-located. Edge server is setup and I think I have configured everything correctly. I have two network adapters, one external facing and one internal facing.
External facing one has dns settings and gateway, internal facing has neither. I have setup persistent routes that enable the edge server to ping hosts in 1.0 and 200.0 no problem. DNS is setup internally so anyone anywhere can ping the edge server (its dns
entry is routable lync2013edge.network.domain.ca). Phones used are the SNOM 720, I have the latest updates applied (8.8.3.27 UC)
On the actual SNOM phone, I will dial 7804636201. It will call and start ringing the other party. Almost exactly 10 seconds later I will hear a busy signal and then the phone displays "Media Connectivity Failure". I ran a log on SIP from the FE
Standard Edition server, here are some entries that I noticed that may have something to do with it (see bottom four paragraphs for SIP CANCEL)
TL_VERBOSE(TF_PARSE) [0]411C.2DE8::02/24/2015-17:23:42.240.0008db5d (SIPStack,CSIPMessage::ParseBufferChain:SIPMessage.cpp(694))( 0000005F03D806F0 ) Start Line: INVITE sip:7804636201;[email protected];user=phone SIP/2.0
TL_INFO(TF_PROTOCOL) [0]411C.2DE8::02/24/2015-17:23:42.269.0009106f (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[3706963737] $$begin_record
Trace-Correlation-Id: 3706963737
Instance-Id: 2F91
Direction: outgoing
Peer: lync2013.network.caedm.ca:5070
Message-Type: request
Start-Line: INVITE sip:[email protected]:5070;user=phone;maddr=lync2013.network.caedm.ca SIP/2.0
From: "Joel Smith" <sip:[email protected]>;tag=2ksjs48fxg;epid=000413774E0401
To: <sip:7804636201;[email protected];user=phone>
Call-ID: 3faa35f677ef48719b27c796251b0519
CSeq: 1 INVITE
Contact: <sip:[email protected];opaque=user:epid:cO0WSS9wCFqUnP0dpEh6uQAA;gruu>;reg-id=1
Via: SIP/2.0/TLS 192.168.100.17:55489;branch=z9hG4bKE036484A.405BD23C943B158E;branched=TRUE
Via: SIP/2.0/TLS 192.168.1.201:51470;branch=z9hG4bK-fdh7rhbbvsri;rport;ms-received-port=51470;ms-received-cid=600
Record-Route: <sip:Lync2013.network.caedm.ca:5061;transport=tls;opaque=state:T;lr>;tag=B39FB8145D545F357B2737F43833CEB4
Max-Forwards: 69
Content-Length: 3563
Content-Type: multipart/alternative;boundary="next_part_u00iwyrezkkuxf3d"
P-Asserted-Identity: "Joel Smith"<tel:+17808092404;ext=2404>
Message-Body: --next_part_u00iwyrezkkuxf3d
Content-Type: application/sdp
Content-Transfer-Encoding: 7bit
Content-Dis; handling=optional; ms-proxy-2007fallback
TL_INFO(TF_DIAG) [0]411C.2DE8::02/24/2015-17:23:42.270.000915ec (SIPStack,SIPAdminLog::WriteDiagnosticEvent:SIPAdminLog.cpp(802))[3706963737] $$begin_record
Severity: information
Text: Routed a locally generated response
SIP-Start-Line: SIP/2.0 100 Trying
SIP-Call-ID: 3faa35f677ef48719b27c796251b0519
SIP-CSeq: 1 INVITE
Peer: 192.168.1.201:51470
Data: destination="[email protected]"
$$end_record
TL_INFO(TF_PROTOCOL) [0]411C.2DE8::02/24/2015-17:23:42.274.000928d4 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[3706963737] $$begin_record
Trace-Correlation-Id: 3706963737
Instance-Id: 2F93
Direction: outgoing;source="local"
Peer: 192.168.1.201:51470
Message-Type: response
Start-Line: SIP/2.0 101 Progress Report
From: "Joel Smith" <sip:[email protected]>;tag=2ksjs48fxg;epid=000413774E0401
To: <sip:7804636201;[email protected];user=phone>
Call-ID: 3faa35f677ef48719b27c796251b0519
CSeq: 1 INVITE
Via: SIP/2.0/TLS 192.168.1.201:51470;branch=z9hG4bK-fdh7rhbbvsri;rport;ms-received-port=51470;ms-received-cid=600
Content-Length: 0
ms-diagnostics: 12006;reason="Trying next hop";source="LYNC2013.NETWORK.CAEDM.CA";PhoneUsage="Long Distance";PhoneRoute="LocalRoute";Gateway="208.68.17.53";appName="OutboundRouting"
$$end_record
TL_INFO(TF_PROTOCOL) [1]411C.2DE8::02/24/2015-17:23:42.488.000930bc (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[741182734] $$begin_record
Trace-Correlation-Id: 741182734
Instance-Id: 2F96
Direction: incoming
Peer: lync2013.network.caedm.ca:5070
Message-Type: response
Start-Line: SIP/2.0 183 Session Progress
FROM: "Joel Smith"<sip:[email protected]>;tag=2ksjs48fxg;epid=000413774E0401
TO: <sip:7804636201;[email protected];user=phone>;tag=d265bdc1c8;epid=0A24894D6D
CALL-ID: 3faa35f677ef48719b27c796251b0519
CSEQ: 1 INVITE
CONTACT: <sip:[email protected];gruu;opaque=srvr:MediationServer:0wzNMLUTNFKXO5KjW1mbdQAA>;isGateway
VIA: SIP/2.0/TLS 192.168.100.17:55489;branch=z9hG4bKE036484A.405BD23C943B158E;branched=TRUE,SIP/2.0/TLS 192.168.1.201:51470;branch=z9hG4bK-fdh7rhbbvsri;rport;ms-received-port=51470;ms-received-cid=600
RECORD-ROUTE: <sip:Lync2013.network.caedm.ca:5061;transport=tls;opaque=state:T;lr>;tag=B39FB8145D545F357B2737F43833CEB4
CONTENT-LENGTH: 1388
CONTENT-TYPE: application/sdp
TL_VERBOSE(TF_NETWORK) [0]411C.2DE8::02/24/2015-17:23:51.369.00098f6b (SIPStack,CRecvContext::CreateIncomingRequest:RecvContext.cpp(920))[3030787245]( 0000005F01E739D0 ) creating SIP_MID_CANCEL request
TL_VERBOSE(TF_PARSE) [0]411C.2DE8::02/24/2015-17:23:51.369.00098f90 (SIPStack,CSIPMessage::ParseBufferChain:SIPMessage.cpp(694))( 0000005F03D7E2E0 ) Start Line: CANCEL sip:7804636201;[email protected];user=phone SIP/2.0
TL_VERBOSE(TF_PARSE) [0]411C.2DE8::02/24/2015-17:23:51.369.00099054 (SIPStack,CSIPMessage::ParseNextHeader:SIPMessage.cpp(1532))( 0000005F03D7E2E0 ) Found Header: Reason: SIP;cause=488;text="Media Connectivity Failure"
TL_INFO(TF_PROTOCOL) [0]411C.2DE8::02/24/2015-17:23:51.369.000990c6 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[3706963737] $$begin_record
Trace-Correlation-Id: 3706963737
Instance-Id: 2FA0
Direction: incoming
Peer: 192.168.1.201:51470
Message-Type: request
Start-Line: CANCEL sip:7804636201;[email protected];user=phone SIP/2.0
From: "Joel Smith" <sip:[email protected]>;tag=2ksjs48fxg;epid=000413774E0401
To: <sip:7804636201;[email protected];user=phone>
Call-ID: 3faa35f677ef48719b27c796251b0519
CSeq: 1 CANCEL
Via: SIP/2.0/TLS 192.168.1.201:51470;branch=z9hG4bK-fdh7rhbbvsri;rport
Max-Forwards: 70
Content-Length: 0
$$end_record
I thought it might be a timeout issue, so I tried following these steps located here:
http://ipfone.hu/lync-mediation-server-cancel-problem/ After rebooting the server no changes were noticed.
I also checked out this website
http://blog.insidelync.com/2013/04/sip-trunking-101-with-lync-server-2013/ regarding disabling the check box "enable outbound routing failover timeout". Doing that had no effect.
Any other ideas would be appreciated.

Hi,
yes I see the config file is very simple and standard.
So the issue with snom on branch sites is random, it's correct?
From what I read in your answer, sometimes you can establish a correct communication between a snom and the called number +17804636201.
Have you tried to collect a network capture on a snom at branch location?
Do you have some other version of snom phone (300, 710, 821) to test?
Do you have some LPE ip-phone (Polycom CX600 o HP4110-4120) to test?
Regards
Luca
Luca Vitali | MCITP Lync/Exchange | snom Certified Engineer | Sonus SBC1000 Engineer

SIP Server vs SIP Servlet

Hello,
Can someone let me exact what the difference between SIP Server and SIP Servlet.?
Regards,
Deepak

It's hard to be precise since "SIP Server" does not have a precise definition, but I'll try to help.
The term SIP Server can apply to any network node meant to affect a SIP session. A server can act as SIP User Agent (UA), proxy, or back-to-back UA. Nothing about the way these applications are programmed is implied by the term "SIP Server."
The SIP Servlet API (SSAPI) is a popular technology for implementing SIP applications; therefore you could say that a SIP Servlet container is a type of SIP Server. The SSAPI is specified in [JSR 289|http://jcp.org/en/jsr/detail?id=289] , and is an extension of the familiar servlet model that is also used for programming HTTP applications.
OCCAS is a SIP Servlet container, and therefore also a SIP Server.
Hope this helps,
Tom.

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I planned to do with route groups, but I can not add on a route group a H323 gateway and a SIP trunk at the same time.
How can route calls in different ways?
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Thanks!
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Ok thanks but one question more.... if I have a MGCP Gateway? Can I do this from my MGCP Gateway? or I need an H323 Gateway.
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Can I do this with Partitions and CSS?
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Sip refer getting 404 & 503

Hi All,
We have a cisco 2911 running 15.0(1r)M6. We have configured sip trunks within CUCM 7.1(5) to the sip router single ip address of 192.168.10.51. When we use the AT&T trunk to make a sip call, everything works as expected (inbound and outbound calls).
We are trying to add a directory handler server (paralance) into the mix. It's ip address 10.13.115.140. The idea behind this system is to call it,tell it who you are looking for, and the call is transfered. So far the system accepts the call, confirmed who you want to be transfered too, and attempts the transfer.
During the transfer, the attached debug is getting us 404, and 503 responses. The caller hears dead air.
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dial-peer voice 296 voip
description CUCM Internal Dialing to CMS1
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session protocol sipv2
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voice-class codec 299
dial-peer voice 297 voip
description CUCM Internal Dialing to CMS2
huntstop
preference 2
destination-pattern 1....$
session protocol sipv2
session target ipv4:10.13.120.6
session transport udp
voice-class codec 299
dial-peer voice 2912411 voip
huntstop
preference 1
destination-pattern 12411
session protocol sipv2
session target ipv4:10.13.115.140
session transport udp
voice-class codec 299
dtmf-relay rtp-nte
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codec preference 1 g711ulaw

This could by one of three things:
1. Your softswitch is setup incorrectly.
2. "789096120445" is not a dialable number.
3. "1026" is not an acceptable caller-id. (this is less likely, I would expect a different error -- but still, is this caller-id allowed?)

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The other idea was to insert a custom header in CVP I could then pull out at the SBC and replace with a REFER. Does anyone have any links to documentation on this?
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TC

how about check "enable send calls to originator" for the refer label routing in CVP? this would bypass proxy.

Video call from Cisco 8945 SCCP to Polycom VVX 1500 Third party Sip Phones between intercluster trunks is not working

Dear All,
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Video Phones (Polycom VVX 1500 ) Third Party SIP Phones located in second cluster, From 1 st cluster cisco 8945 Video
phone to 2nd cluster Polycom Video phone all calls are works for voice call only, but no video ,
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Thank you,
Sriman

Try setting up a SIP trunk between the two clusters and set a route patten just to the VVX 1500 and check how that goes.
From memory inter-cluster trunks are a H.323 like protocol which might have video inter-op issues with the Polycom device.

SIP Refer problem

Hi Guys,
I have problem at one of our customer side. When an operator received a call, and try to do a transfer (consult transfer) to a colleague, the colleague's phone ring one time and the call failed. The operator have to take back the call and give the full external number of the colleague to the caller.
After a reboot of the phone (unplugged/plugged) the transfer work fine !
I tried to find the cause in the CUCM traces but I need help.
Saw that Refer messages:
14:49:26.294 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 22.212.126.35 on port 50108 index 227
[21667403,NET]
REFER sip:[email protected]:50108 SIP/2.0
Via: SIP/2.0/TCP 22.214.10.18:5060;branch=z9hG4bKf63095b1b37b1
From: <sip:[email protected]>;tag=435147058
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 REFER
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=tcp>
User-Agent: Cisco-CUCM8.6
Require: norefersub
Expires: 0
Refer-To: cid:[email protected]
Content-Id: <[email protected]>
Content-Type: application/x-cisco-remotecc-request+xml
Referred-By: <sip:[email protected]>
Content-Length: 341
<x-cisco-remotecc-request>
<answercallreq>
    <dialogid>
      <callid>[email protected]</callid>
      <localtag>6511241~ada938a2-d3e6-4928-be65-07fc5a6e0f23-49485024</localtag>
      <remotetag>3cce73acbe00bd3fda57ded7-91adfcf9</remotetag>
    </dialogid>
</answercallreq>
</x-cisco-remotecc-request>
|2,200,21,1.3256060^22.214.10.2^SEP3CCE73ACBE00
After the OK:
14:49:26.317 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 22.212.126.35 on port 50108 index 227 with 442 bytes:
[21667404,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 22.214.10.18:5060;branch=z9hG4bKf63095b1b37b1
From: <sip:[email protected]>;tag=435147058
To: <sip:[email protected]>;tag=3cce73acbe00bd40ed051b81-4b2d068b
Call-ID: [email protected]
Date: Thu, 07 Feb 2013 13:49:17 GMT
CSeq: 101 REFER
Server: Cisco-CP7962G/9.3.1
Contact: <sip:[email protected]:50108;transport=TCP>
Content-Length: 0
|3,100,63,1.7871803^22.212.126.35^*
And thats the last messages with call ID starting by: 2e587c80
So could you say me how I can find the Transfer "destination" and why the call failled ?
Thanks,
Hervé Jacquemin

Hi hjacquemin,
Have you solved this problem? I found the same thing with my customer. We are using
6921 with SIP Firmware. It is very weird that my IP Phone was hanged and must reset by unplug and plug.
Winai.

Lync Federation - Accept SIP Reverse Negotiation (SIP Invite without SDP)

Hello,
Recently I tested a SIP Federation trunk between Lync Server 2013 and non-Lync Client.
In this scenario the Lync Client 2013 support SIP Reverse Negotiation, by other words if SIP Invite without SDP it's sent to Lync Client 2013 it will be accepted by any configuration option?
With the default settings seams that it's not supported with error reason "Error parsing body"
Trace-Correlation-Id: 3549384327
Instance-Id: 4C9
Direction: outgoing
Peer: lynctest.domain.com:2138
Message-Type: response
Start-Line: SIP/2.0 488 Not Acceptable Here
From: "User4" <sip:[email protected]>;tag=3794445243
To: <sip:[email protected]>;epid=abad235729;tag=a130a7e357
Call-ID: [email protected]
CSeq: 12784624 INVITE
Via: SIP/2.0/TLS 172.16.3.51:5065;branch=z9hG4bK-5765F571;rport;alias;received=172.16.3.51;ms-received-port=2138;ms-received-cid=1200
Content-Length: 0
ms-client-diagnostics: 52009;reason="Error parsing body"
Regards,
Claudio

Hello All,
After some analysis I got the following conclusions.
Lync PC Client doesn't accept initial Invite without SDP ( Delayed Offer ).
However our goal was to test the SIP Reverse Media Negotiation mechanism, so we sent initially a dummy SDP for the initial invite and after the connect send a SIP INVITE without SDP and for my surprise the Lync Client accepted and sent his own SDP on the
200 OK and we sent the new SDP offer in the ACK.
However the result was no Audio, and Lync Client kept sending the Audio to the initial INVITE SDP and ignored the new SDP offered in the ACK message.
So my conclusion it's that LYNC Client doesn't support SIP Reverse Media Negotiation (Delayed Offer) at all since it ignores the new SDP offered in the ACK message for the mid call media renegotiation attempt with SIP INVITE without SDP.
Traces:
INVITE sip:172.16.1.87:64425;transport=tls;ms-opaque=a3edae884b;ms-received-cid=1F9C00;grid SIP/2.0
Record-Route: <sip:LYNC2013-FE.domain.sifi:5061;transport=tls;opaque=state:F:Ci.R1f9c00;lr;ms-route-sig=aaCRxLomQ6J6ATKjSZx4vJQ22miSAfUAExqMcDvEWyHdss4x_99VHTLQAA>;tag=C161B833E3EAA57C26010E775AC607C8
Via: SIP/2.0/TLS 172.16.0.37:5061;branch=z9hG4bK97FD40ED.FD1FE32C7CB76CCD;branched=FALSE;ms-internal-info="bb0yvN-Txta-aXcfTMPmVSdyK0kBz7b-pamgWfOIbn8vks4x_9o9kUwQAA"
Via: SIP/2.0/TLS 172.16.13.192:5065;branch=z9hG4bK-57656CED;rport;alias;received=172.16.13.192;ms-received-port=2051;ms-received-cid=1D0800
Proxy-Authentication-Info: Kerberos qop="auth", opaque="D83CD7C8", srand="F5054EF3", snum="104", rspauth="040401ffffffffff0000000000000000e9693240576b479326af5617", targetname="sip/LYNC2013-FE.domain.sifi",
realm="SIP Communications Service", version=4
Max-Forwards: 56
From: "" <sip:[email protected]>;tag=3691888833
To: <sip:[email protected]>;epid=8a34f77d58;tag=4066ac742a
Call-ID: [email protected]
CSeq: 12046301 INVITE
Contact: <sip:[email protected]:5065;transport=TLS>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE,PUBLISH
Content-Length: 0
Require: 100rel
Supported: 100rel,replaces,privacy,timer,from-change,histinfo,answermode
User-Agent: (Virtual Appliance)
P-Asserted-Identity: "" <sip:[email protected]>
Session-Expires: 720;refresher=uac
P-Sig-Options: Sending-Complete
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 172.16.0.37:5061;branch=z9hG4bK97FD40ED.FD1FE32C7CB76CCD;branched=FALSE;ms-internal-info="bb0yvN-Txta-aXcfTMPmVSdyK0kBz7b-pamgWfOIbn8vks4x_9o9kUwQAA"
Via: SIP/2.0/TLS 172.16.13.192:5065;branch=z9hG4bK-57656CED;rport;alias;received=172.16.13.192;ms-received-port=2051;ms-received-cid=1D0800
From: "" <sip:[email protected]>;tag=3691888833
To: <sip:[email protected]>;epid=8a34f77d58;tag=4066ac742a
Call-ID: [email protected]
CSeq: 12046301 INVITE
User-Agent: UCCAPI/15.0.4701.1000 OC/15.0.4701.1000 (Microsoft Lync)
Proxy-Authorization: Kerberos qop="auth", realm="SIP Communications Service", opaque="D83CD7C8", targetname="sip/LYNC2013-FE.domain.sifi", crand="785246a1", cnum="92", response="040400ffffffffff000000000000000000b60640ac2c60c49bc1b427"
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.16.0.37:5061;branch=z9hG4bK97FD40ED.FD1FE32C7CB76CCD;branched=FALSE;ms-internal-info="bb0yvN-Txta-aXcfTMPmVSdyK0kBz7b-pamgWfOIbn8vks4x_9o9kUwQAA"
Via: SIP/2.0/TLS 172.16.13.192:5065;branch=z9hG4bK-57656CED;rport;alias;received=172.16.13.192;ms-received-port=2051;ms-received-cid=1D0800
From: "" <sip:[email protected]>;tag=3691888833
To: <sip:[email protected]>;epid=8a34f77d58;tag=4066ac742a
Call-ID: [email protected]
CSeq: 12046301 INVITE
Record-Route: <sip:LYNC2013-FE.domain.sifi:5061;transport=tls;opaque=state:F:Ci.R1f9c00;lr;ms-route-sig=aaCRxLomQ6J6ATKjSZx4vJQ22miSAfUAExqMcDvEWyHdss4x_99VHTLQAA>;tag=C161B833E3EAA57C26010E775AC607C8
Contact: <sip:[email protected];opaque=user:epid:wc5Y6-kDo16CxuVbyxqk9gAA;gruu>
User-Agent: UCCAPI/15.0.4701.1000 OC/15.0.4701.1000 (Microsoft Lync)
Supported: histinfo
Supported: ms-safe-transfer
Supported: ms-dialog-route-set-update
Proxy-Authorization: Kerberos qop="auth", realm="SIP Communications Service", opaque="D83CD7C8", targetname="sip/LYNC2013-FE.domain.sifi", crand="e903d142", cnum="93", response="040400ffffffffff0000000000000000dbe0e9524a1031ef81a19d2f"
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 0 1 IN IP4 172.16.1.87
s=session
c=IN IP4 172.16.1.87
b=CT:99980
t=0 0
m=audio 12530 RTP/SAVP 8 0 13 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mIMHiJBpn4ZRZfg2VXYSTdQfS4wyJ0x57QQ0q4kU|2^31
a=maxptime:200
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
ACK sip:172.16.1.87:64425;transport=tls;ms-opaque=a3edae884b;ms-received-cid=1F9C00;grid SIP/2.0
Via: SIP/2.0/TLS 172.16.0.37:5061;branch=z9hG4bK301D467E.2E943CC97CBC4CCD;branched=FALSE
Via: SIP/2.0/TLS 172.16.13.192:5065;branch=z9hG4bK-57656CEE;rport;received=172.16.13.192;ms-received-port=2051;ms-received-cid=1D0800
Proxy-Authentication-Info: Kerberos qop="auth", opaque="D83CD7C8", srand="B8AB5336", snum="105", rspauth="040401ffffffffff0000000000000000de85d6c7415302c9b7535777", targetname="sip/LYNC2013-FE.domain.sifi",
realm="SIP Communications Service", version=4
Max-Forwards: 69
From: "" <sip:[email protected]>;tag=3691888833
To: <sip:[email protected]>;epid=8a34f77d58;tag=4066ac742a
Call-ID: [email protected]
CSeq: 12046301 ACK
Contact: <sip:[email protected]:5065;transport=TLS>
Content-Length: 326
Content-Type: application/sdp
v=0
o=- 262 2 IN IP4 172.16.13.192
s=session
t=0 0
m=audio 16392 RTP/SAVP 8 101 13
c=IN IP4 172.16.13.191
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:X0rDwl9KxCJfSsRaX0rEkl9KxNJfSsUCX0rFOtIK|2^31

Trying to use an SPA3102 to connect a SIP and non-SIP Telephone System

I have a new Mitel 3300 system which is working very well, but for 12 months or so ( until I can replace it with another MITEL ) I need to connect it to an old INDEX system.
I have got an SPA3102, and for the moment it is connected to an extension on the INDEX and a phone, which I will be replacing with an analogue trunk once testing is complete.
The SIP portion is giving me a huge headache. The MITEL side is nice and easy, poking the right range of extensions to a SIP gateway won't take long
The SPA3102 is a real pain, the Manual I have for the SPA just seems to list ranges of values and does not seem to show how to configure it to talk to my MITEL, I can't work out how to get it to make SIP calls across the ethernet interface at all.
Can anyone help, or perhaps provide a decent guide on how to do this.

Headbolt wrote:
This is what I want to achieve.
From the MITEL, when you dial a 4 digit extension 47xx the MITEL will make a SIP call to the SPA3102, which will then in turn use the port that is plugged into an INDEX extension to make an internal call on the INDEX to extension 47xx.
You will be using the pstn tab and the voip-to-pstn gateway on the SPA3102. There are two different ways you can configure the SPA3102 for the voip-to-pstn gateway.
One way is with no authentication and the Mitel will send a call to the sip uri represented by the userid@ipaddress: port_number and the SPA3102 will return a dial tone to the caller and the caller will then enter the INDEX extension they wish to dial.
The other way is with http_digest authentication where the Mitel system will send a Sip Invite to the SPA at the extnumber@ipaddress: port number with the INDEX extension to be dialed and will respond with the authid and the password at the appropriate time.
Without a personal understanding the capabilities of the MITEL system to send and receive sip calls I can't give more specific instructions.
From the INDEX when we dial a 4 digit 65xx, the INDEX will send the call out an Analogue trunk that is connected to the SPA3102 which will then make a SIP call to the MITEL which will ring it's extension 65xx
The forwarding of the call from the INDEX system, after it is answered by the pstn-to-voip gateway of the SPA3102 is done thru the appropriate dial plan of the 3102. If the INDEX system is sending the dial digits after the 3102 pstn-to-voip gateway answers the call then the call will be sent to the userid and ip_address in the proxy/userid on the spa3102's pstn line tab.
Without a personal understanding of the capabilities of the MITEL system to send and receive sip calls I can't give more specific instructions.

WRT310N No Way To Setup and Configure Router. CD Setup No, Web Based Setup No

Purchased New WTR310N yesterday. LESA does not run from CD. Web based Setup Page Will Not Load. Did two 2 hr Remote Setup sessions so far with Linksys Techs in Phillipines and India. After 1st Tech was done, nothing would connect to router. Tech tried to setup our 4 PCs by setting up one as a server and using ICS to setup the rest. Tech disabled TTP, DHCP, ete everything that works as a home network. I think Tech also setup a password and user name on Router without telling me. I had to spend 2 hrs using Vista and XP Wizards to setup a router network, both wired and wireless. Since Router is not configured yet can only access unsecured wireless networks. But 2 wireless printers do not work yet. Tech today was excellent in India. She reinstalled my Linksys USB Wireless Adapter that I use to print with. Still not printing, and green light not coming on, on USB Adapter. Tech managed to upgrade Router Firmware, but could never get web based router setup to load. Got the password and user name box to show up once, but admin and blank username does not work. My XBox 360 installed in 15 seconds and is fast so Router working fine. So need help with getting Linksys USB Wireless Adapters to access wireless printers and somehow get router admin pages to load so I can turn on WEP and type in our WEP Code. Last router was Linksys 54G, all worked fine for years, then quit working so got new router. Any ideas before I try calling Tech Support again, thanks.

As you want to make the network secured but the admin password is not working....You have to reset the router and re-configutre it again...
Press and hold the reset button for 60 seconds...Release the reset button...Unplug the power cable from your router, wait for 30 seconds and re-connect the power cable...Now re-configure your router...
If your Internet Service Providor is Cable follow this link
If your Internet Service Providor is DSL follow this link

Route Map Policy on SVI - Trunk from ESX

Hi,
I have a question regarding the following configuration.
A route map matches traffic from a particular subnet, say on VLAN 10 (using an ACL).
A route map policy is applied on this SVI (int vlan 10)
A server on this subnet is running on ESX which is connected to the switch on a trunk port.
The ESX host tags all frames from this server as VLAN 10.
In this scenario, should the route map pick up the traffic from this server? I don't see why not, but in my testing it doesn't seem to be working :)
Thanks for any help.

Hi Alex,
It's a 3750x (stack) with 12.2(55)SE5.
I've already changed the SDM template to routing and rebooted the switch.
I don't think the route map is working at all actually :) See config below, let me know if you can spot anything obvious but the networks on the ACL are definitely correct.
Thanks again.
Extended IP access list UPLINK2
10 permit ip 192.168.1.0 0.0.0.255 any
20 permit ip 192.168.4.0 0.0.1.255 any (305 matches)
route-map ROUTE1 permit 10
match ip address UPLINK2
set ip next-hop 10.1.1.253
interface Vlan10
ip address 192.168.5.254 255.255.254.0
ip policy route-map ROUTE1
end

Configure SPA2102 as SIP Phone to SIP Proxy Server

I have a SIP2102 which needs to be configured as SIP phone to a SIP Proxy Server. All calls will stay within the local network. I need to point SPA2102 to my SIP Proxy Server and assign an extension such that it is recognized by the SIP Proxy Server as part of its pool of valid extensions. The documentation is not clear when just trying to set these parameters.

if what you are trying to accomplish is simply register the SPA2102 with your Proxy server then the only thing you need to do is configure a USER ID (this is the extension number you want the SPA to have) and a Proxy (IP address of your SIP server) -- outbound proxy is only needed if your server requires the device to have this..
Both of these parameters can be found under Line 1 tab and these settings should come from your SIP server-- Internet port of the device must be connected to your network
SIP Port is another to consider in case your SIP server is using another port other than 6060..
| isolate! isolate! isolate! |

Router 2811 and C2960 Switch Trunking Problem

Hi all
I got an problem with a trunking problem between Router 2811 and C2960 switch
In router 2811 - I created f0/0.1 10.65.20.1 (VLAN 1) and f0/0.48 10.65.23.1 (VLAN 48)
In C2960 - Vlan 1 10.65.20.30 , VLAN 48 10.65.23.30
Finally I can only ping VLAN 1 IP but fail to ping VLAN 48 IP, can help me how to troubleshoot it?
Hugo
Router 2811 Configuration:
interface FastEthernet0/0.1
encapsulation dot1Q 1 native
ip address 10.65.20.1 255.255.255.0
interface FastEthernet0/0.48
encapsulation dot1Q 48
ip address 10.65.23.1 255.255.255.0
C2960 Configuration:
interface FastEthernet0/24
switchport mode trunk

2811#sh vlans
Virtual LAN ID: 1 (IEEE 802.1Q Encapsulation)
vLAN Trunk Interface: FastEthernet0/0.1
This is configured as native Vlan for the following interface(s) :
FastEthernet0/0
Protocols Configured: Address: Received: Transmitted:
IP 10.65.20.1 388873 262275
Other 0 1723
390760 packets, 71854310 bytes input
263998 packets, 53723195 bytes output
Virtual LAN ID: 48 (IEEE 802.1Q Encapsulation)
vLAN Trunk Interface: FastEthernet0/0.48
Protocols Configured: Address: Received: Transmitted:
IP 10.65.23.1 0 0
Other 0 20
0 packets, 0 bytes input
20 packets, 1883 bytes output
2960_24#sh int trunk
Port Mode Encapsulation Status Native vlan
Fa0/24 on 802.1q trunking 1
Gi0/1 on 802.1q trunking 1
Port Vlans allowed on trunk
Fa0/24 1-4094
Gi0/1 1-4094
Port Vlans allowed and active in management domain
Fa0/24 1,48
Gi0/1 1,48
Port Vlans in spanning tree forwarding state and not pruned
Fa0/24 1,48
Gi0/1 1,48

Route SIP REFER to SIP Trunk based on DN

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