WTF is OOD Refer ?

According go SIP System Administration Guide:

Out-of-dialog REFER (OOD-R) enables remote applications to establish calls by sending a REFER message to Cisco Unified SRST without an initial INVITE. After the REFER is sent, the remainder of the call setup is independent of the application and the media stream does not flow through the application. The application using OOD-R triggers a call setup request that specifies the Referee address in the Request-URI and the Refer-Target in the Refer-To header. The SIP messaging used to communicate with Cisco Unified SRST is independent of the end-user device protocol, which can be H.323, POTS, SCCP, or SIP. Click-to-dial is an example of an application that can be created using OOD-R.

A click-to-dial application enables users to combine multiple steps into one click for a call setup. For example, a user can click a web-based directory application from his or her PC to look up a telephone number, off-hook the desktop phone, and dial the called number. The application initiates the call setup without the user having to out-dial from his or her own phone. The directory application sends a REFER message to Cisco Unified SRST, which sets up the call between both parties based on this REFER.

Figure 3 shows an example of OOD-R being used by a click-to-dial application. In this scenario, the following events occur (refer to the event numbers in the illustration):

1. Remote user clicks to dial.

2. Application sends out-of-dialog REFER to Cisco Unified SRST.

3. Cisco Unified SRST 1 connects to SIP phone 1 (Referee).

4. Cisco Unified SRST 1 sends INVITE to SRST 2.

5. Cisco Unified SRST 2 sends INVITE to SIP phone 2 (Refer-Target) and the call is accepted.

6. Voice path is created between the two SIP phones.

Note The connectivity to Cisco Unified Communications Manager has been lost and, therefore, IP phone 1 and IP phone 2 have registered to SRST routers.

Figure 3 Click-to-Dial Application using Out-of-Dialog REFER

via Cisco Unified SIP SRST System Administrator Guide – 4.1 Features  [Cisco Unified Survivable Remote Site Telephony] – Cisco Systems.

Dependency records

Do not forget to enable dependency record !

Phone booting process

1. CTLSEP MAC File—For example, CTLSEP003094C25D2E.tlv.
2. SEP MAC File—For example, SEP003094C25D2E.cnf.xml.
3. SIP MAC File—For example, SIP003094C25D2E.cnf.
4. MGCP MAC File—For example, MGC003094C25D2E.cnf.
5. XML Default File—For example, XMLDefault.cnf.xml.
6. SIP Default File—For example, SIPDefault.cnf.
7. MGCP Default File—For example, MGCDefault.cnf.

sip 7940 without registar on CME

How 7940 behave withour registar enabled ? Look

#SIP/2.0 503 Service Unavailable – registrar unavail or not enabled
Date: Fri, 27 Mar 2009 03:34:10 GMT
From: ;tag=001a2ff59e1d000d78edc158-640d9129
Content-Length: 0
To: ;tag=155644-149B
Call-ID: 001a2ff5-9e1d000d-5461353a-4e97def1@172.30.2.101
Via: SIP/2.0/UDP 172.30.2.101:5060;branch=z9hG4bK51f260c8
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 106 REGISTER

How to fix ?

voice service voip
allow-connections sip to sip
sip
registrar server expires max 1200 min 300

Lab #1

Register phones according to table 1 of http://bitswap.wordpress.com/2009/03/25/t/

1D 7940 as SIP on CME1
43 7960+7914 as SCCP on CME1
CD 7961 as SIP on CME2
8A 7940 as SCCP on CME2
E7 7960 as SIP on CM
9B 7960 as SIP on CM
31 7941 as SCCP on CM

Test call completion, calling/ed name presentation , issues
CM1->CM2(forwarded to neighbor)
CM1->CM2(forwarded to CM1)
CM1->CM2(forwarded to CM)

CM->CM2(forwarded to neighbor)
CM->CM2(forwarded to CM1)
CM->CM2(forwarded to CM2)

Connect CM & CME via H323 trunk (full mesh)
Test call completion, calling/ed name presentation , issues

1. Default configuration.
2. H.450 extension turned on

Connect CM & CME via gatekeeper-controlled H.323 trunk
1. Default configuration.
2. H.450 extension turned on

Connect CM & CME SIP trunk(full mesh)
1. Default configuration.
2. REFER off

Connect CM & CME via ip-ip gateway (SIP)
1. Default configuration.