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¶© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 1 Configuration Example Analog FXS port SIP Registration with CUCM

/mcp© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 1 Configuration Example Analog FXS port SIP Registration with CUCM

© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 2 Version: 23rd February 2023 Introduction This document helps configure FXS ports as SIP endpoints registered to Cisco Unified Communications Manager (CUCM) in order to support supplementary services on SIP Endpoints. Prerequisites Cisco recommends to have knowledge of these subjects SIP protocol Foreign Exchange Station (FXS) ports Cisco Unified Communications Manager (CUCM) Cisco Analog Voice Gateway (VG Series) Requirements The FXS ports for Supplementary Services support CUCM version 12.5.1 SU1 or later with IOS XE 16.12.1 and above. CUCM and IOS release for all platforms ISR4461/VG450 IOS: 16.12.1 CUCM: 12.5.1 SU1 VG420 IOS: 17.6.1 CUCM: 12.5.1 SU4, 14.0 SU1 ISR4K, C8300, C8200 and VG400 IOS: 17.8.1 CUCM: 14.0 SU1 Supported Platforms Cisco ISR4451-X/K9 Cisco ISR4461/K9 Cisco ISR4431/K9 Cisco ISR4351/K9 Cisco ISR4331/K9 Cisco ISR4321/K9 C8300-2N2S-4T2X C8300-2N2S-6T C8300-1N1S-4T2X C8300-1N1S-6T C8200-1N-4T C8200L-1N-4T VG400-2FXS/2FXO VG400-4FXS/4FXO VG400-6FXS/6FXO

© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 3 VG400-8FXS VG420-144FXS VG420-84FXS/6FXO VG420-132FXS/6FXO VG450/K9* VG450-72FXS/K9 VG450-144FXS/K9 *Requires a NIM Supported Features: The following supplementary services are supported. Call Hold Call Waiting Call Transfer (unattended/attended) Call Forward no Answer Audio Message Waiting Indicator Call Forwarding Unrestricted Call Forward Busy Call Park Directed Call Pickup Directed Call Pickup Group Three-way Conference Configure To implement supplementary services for Foreign Exchange Station (FXS) ports the call control server (CUCM) should be able to subscribe to the hookflash or onhook events. This requires FXS ports to be registered to CUCM as SIP endpoints. The use of SIP over SCCP facilitates features such as SIP Header modification, endpoint based call routing and enables new features such as directed call retrieval. Network Diagram


© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 4 Configuration steps This section describes the configuration required for this feature to work: Configuring the Device Control Session Application DSAPP (Device control Session Application) is the application that drives these Hook Flash features. It can be configured globally or on a dial-peer basis. SUMMARY STEPS
© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 5 for any outbound call. If all outbound calls that use the hookflash functionality are on the same server, it is recommended to use the param dial- peer command. When multiple matches are possible on hookflash, enable peer parameters callXXXX TRUE for DSAPP to interpret hookflash to SIP supplementary service messages Step 5 param callWaiting string Example: Router(config)#application router(config- app)#service dsapp Router(app-global)#param dialpeer 100 Router(app-global)#param callWaiting TRUE Enables call waiting feature. Step 6 param callConference string Enables call conference feature. Step 7 param callTransfer string Example: Router(config)#application router(config- app)#service dsapp Router(app-global)#param dialpeer 100 Router(app-global)#param callWaiting TRUE Router(app-global)#param callConference TRUE Router(app-global)#param callTransfer TRUE Enables call transfer feature. Configuring the Outbound VoIP Dial-peer Outbound dial-peer is configured like regular voip dial-peer for SIP. In addition to the parameters required, the following configurations are required: • service dsapp– specifies this dial-peer will be controlled by dsapp application • session transport tcp – specifies only TCP signaling is supported now • voice-class sip extension gw-ana – used to interop with CUCM • voice-class sip bind control source-interface GigabitEthernet0/0/1 – need this interface’s mac as the base mac dial-peer voice 714281111 voip service dsapp destination-pattern .+ session protocol sipv2 session transport tcp session target ipv4:172.16.10.10 incoming called-number 7141116... voice-class sip bind control source-interface GigabitEthernet0/0/0
© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 6 codec g711ulaw no shut Note- G711 is the only codec supported for conference calls. It is recommended to add this command. Configuring Pots Dial-peer You can configure the pots dial-peer like a regular pots dial-peer for FXS. In addition to the parameters required, you have to configure the following command under pots dial-peer to interpret hook flash correctly and interop with CUCM: • service dsapp — specifies this dial-peer to be controlled by DSAPP application. • voice-class sip extension gw-ana – this parameter is used to interop with CUCM. dial-peer voice 19993000 pots service dsapp destination-pattern 2124506300 voice-class sip extension gw-ana port 3/0/0 Configuring Voice-card and SIP When you configure the voice-card, all the traffic should go through the CUCM, the hairpin calls are not supported. You have to configure no local-bypass command for the voice-card that have FXS SIP endpoints. For FXS SIP endpoints to register, configure the registrar IP address under the sip-ua mode and use the TCP as the transport type. UDP protocol is not supported. voice service voip sip bind control source-interface GigabitEthernet0/0/0 session transport tcp no shut ! voice-card 3/0 no local-bypass no watchdog ! ! sip-ua registrar ipv4:172.16.10.10 expires 3600 tcp protocol mode dual-stack !
© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 7 Note - SIP service should be shut down before configuring the protocol mode. After configuring the protocol mode as dual-stack, SIP service should be reenabled. Enabling Device Control Session Application Line features To register to CUCM as a SIP endpoint, and to distinguish line feature from trunk, configure the dsapp line command. SUMMARY STEPS
© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 8 Router(config)#dsapp line feature access-code Router(config-dsappline-fac)#prefix *# Router(config-dsappline-fac)#cancel-call-waiting **4 Router(config-dsappline-fac)#exit Router# show dsapp line feature codes dsapp line feature access-code prefix *# call forward all *#1 call forward cancel *#2 pickup local *#5 pickup group *#7 pickup direct *#6 cancel-call-waiting **4 last-redial *#3 If the dsapp line feature access-code is not configured, the voice gateway does not translate the FAC to the format that the CUCM understands. The whole FAC digits is sent to the CUCM. After the FAC is disabled and re-enabled, all the FAC and prefix are rolled back to the default values. Router(config)#no dsapp line feature access-code Feature access-code disabled Router(config)#do show dsapp line feature codes dsappline feature access-code disabled Router(config)#dsapp line feature access-code Router(config-dsappline-fac)#do show dsapp line feature codes dsapp line feature access-code prefix ** call forward all **1 call forward cancel **2 pickup local **5 pickup group **7 pickup direct **6 cancel-call-waiting **9 last-redial **3 Router(config-dsappline-fac)#do show run | b dsapp line dsapp line ! dsapp line feature access-code Auto Configuration To enable the auto-configuration, use the ccm-manager sipana auto-config local command. To get the XML configuration file, use the ccm-manager config server command to download the configuration file
© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 9 from the CUCM TFTP server. Configurations are needed on both CUCM and voice gateway. CUCM needs to be configured first, then those configurations can be pushed to the voice gateway. ! ccm-manager sipana auto-config local GigabitEthernetx/y/z ! ccm-manager config server 172.xx.0.0 Note- Auto-Config only adds the dialpeers for each endpoint configured on CUCM. All other required SIP CLI commands need to be added. CUCM Configuration VG450 is used in this example. 1. Navigate to Device> Gateway>Add New>Gateway Type. 2. Select SIP as the protocol and click Next 3. Enter the mac address of the interface used in sip bind control.


© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 10 4. Configure the voice module and individual voice ports.



© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 11 5. Add a directory Number > Save>Apply config. 7. The Analog port would now show as registered Limitations



Registered with Call Manager: Yes Local interface: GigabitEthernet0/0/0 (2c5a.0fc8.8b70) Current version-id: 1541004382-f60b9ac2-ce5b-439e-92e5-02b62e26d15c Current config applied at: 16:47:40 UTC Oct 31 2018 Gateway downloads succeeded: 2 Gateway download attempts: 2 Last gateway download attempt: 16:47:40 UTC Oct 31 2018 Last successful gateway download: 16:47:40 UTC Oct 31 2018 Current TFTP server: 172.19.156.84 Gateway resets: 1 Managed endpoints: 3 Endpoint downloads succeeded: 6 Endpoint download attempts: 6 Last endpoint download attempt: 16:47:40 UTC Oct 31 2018 Last successful endpoint download: 16:47:40 UTC Oct 31 2018 Endpoint resets: 0 Endpoint restarts: 0
© 2023 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information 13 Configuration Error History: Troubleshoot For registration issues capture CUCM SDL/SDI traces and run the following debugs on the gateway debug voip application session debug ccsip messages debug ccsip states debug ccsip error For auto-configuration issues run the debug debug ccm-manager config-download all Related Information Configuring the Cisco Fourth-Generation T1/E1 Voice and WAN Network Interface Module
