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Assistance with Gateway/CUBE voice recording to multiple destinations

Tori_Ok.
Level 1
Level 1

Could anyone kindly give me a clue how to record audio calls to two different recording servers simultaneously. I've been looking at the CUBE documentation for media recording and it appears SIPREC and the UC Gateway API XMF should allow me record calls off my IOS CUBE/gateway. My gateway is an ISR4351, with license for CUBE.

 

Any assistance appreciated.  

 

 

 

 

1 Accepted Solution

Accepted Solutions

I have done a small number of dial-peer forking where the customer wanted to record calls to their out of hours service. In this case there is a specific dial-peer that only matches the one number and that forks the media.

Every call across the CUBE has a inbound and an outbound dial-peer. If you fork from the inbound dial-peer the called and calling numbers are the correct way around, if you fork from the outbound dial-peer the numbers will be swapped.

This is my setup for recording calls to a specific out of hours number

 

dial-peer voice 3000 voip
 description dial-peer to fork to Calabrio Call Recorder
 preference 1
 destination-pattern 99999
 signaling forward none
 session protocol sipv2
 session target ipv4:192.168.10.2
 session transport udp
!

media profile recorder 10000
 media-recording 3000
!
media class 3
 recorder profile 10000
!

dial-peer voice 500 voip
 description from UCM inbound call leg for call recording
 session protocol sipv2
 session target sip-server
 incoming called-number <Specific Dialled Number>
 voice-class codec 1  
 media-class 3
 dtmf-relay rtp-nte
 no vad
!

If you want to record both inbound and outbound calls you will need at least 4 dial-peers plus the extra ones to send to the subscibers.

In your example below you will need to split those into two dial-peers and for each dial-peer to go to a differant recorder.

Graham

View solution in original post

3 Replies 3

Graham Old
Level 7
Level 7

The easy way to record is the built in bridge in the phone. Then the SIP Invite comes from UCM and the Audio Stream from the phone.

If you have a SIP trunk between the Gateway and UCM then you can use gateway preferred.  That way the SIP invite comes from UCM and the audio from the gateway for external calls. Internal calls that do not cross the gateway continue to send the audio from the phone. The XMF command is part of the gateway preferred setup.

You can also do dial peer recording. In that case both the SIP Invite and the audio stream come from the gateway and this is independent of UCM and the BiB. You can also record from both the inbound and outbound dial peers so you can support 3 separate call recorders. Two from the CUBE doing dial-peer recording and one from UCM.

You can only do dial-peer recording/forking if both call legs are SIP. If you have E1/T1 then it can be done but gets a lot more complex.

 

This is the CUBE Setup for gateway preferred.

HTTP Setup

 

ip http server

ip http max-connections 100

ip http timeout-policy idle 600 life 86400 requests 86400

http client connection idle timeout 600

 

wsapi setup

 

uc wsapi

 message-exchange max-failures 2

 source-address 10.99.18.12

 probing interval negative 10

 probing interval keepalive 255

 probing max-failures 5

 !

 provider xmf

  no shutdown

  remote-url 1 http://10.99.18.53:8090/ucm_xmf

 !

 !

 

Notes

source-address 10.99.18.12

This is the IP address used by the CUBE for the wsapi messages

 

remote-url 1 http://10.99.18.53:8090/ucm_xmf

This is the UCM that is allowed to send wsapi commands to the CUBE

You must do an index for each UCM publisher/subscriber as in

remote-url 1 http://10.12.0.100:8090/ucm_xmf

remote-url 2 http://10.12.0.101:8090/ucm_xmf

 

 Graham

My setup is all SIP, i.e. SIP-only phones (Cisco and third-party), SIP trunk to GW/CUBE, SIP trunk to PSTN. I've been able to record calls to/from Cisco phones using the BIB and phone-preferred as media source. The third-party phones do not have a BIB and my goal is to record calls out to the PSTN through these third-party phones. So I guess I can set those to GW-preferred for media source. 

 

However, crucial piece for my setup is to send same audio stream to both audio recorders. If I understand correctly, I can achieve this using different dial-peers, right? My current config uses same DP for inbound from SP and outbound to CUCM Pub & Sub as follows:

dial-peer voice 12 voip
 description ## Inbound from ITSP & Outbound to CUCMPub##
 translation-profile incoming XLATE-to-4XXX
 destination-pattern [3-4]...$
 session protocol sipv2
 session target ipv4:192.168.30.12
 incoming called-number 5556667777
 voice-class codec 2  
 dtmf-relay rtp-nte
 no vad
!
!
dial-peer voice 13 voip
 description ## Inbound from ITSP & Outbound to CUCMSub##
 translation-profile incoming XLATE-to-4XXX
 preference 1
 destination-pattern [3-4]...$
 session protocol sipv2
 session target ipv4:192.168.30.13
 incoming called-number 5556667777
 voice-class codec 2  
 dtmf-relay rtp-nte
 no vad

Do I break these up into separate DPs ( one for inbound from SP and two outbound to CUCM Pub and Sub) then set the recording server as a session-target for the inbound DP? Still not clear how I can record from two DPs simultaneously. Was hoping to test this today but unable to access the equipment.

 

Thanks.

 

I have done a small number of dial-peer forking where the customer wanted to record calls to their out of hours service. In this case there is a specific dial-peer that only matches the one number and that forks the media.

Every call across the CUBE has a inbound and an outbound dial-peer. If you fork from the inbound dial-peer the called and calling numbers are the correct way around, if you fork from the outbound dial-peer the numbers will be swapped.

This is my setup for recording calls to a specific out of hours number

 

dial-peer voice 3000 voip
 description dial-peer to fork to Calabrio Call Recorder
 preference 1
 destination-pattern 99999
 signaling forward none
 session protocol sipv2
 session target ipv4:192.168.10.2
 session transport udp
!

media profile recorder 10000
 media-recording 3000
!
media class 3
 recorder profile 10000
!

dial-peer voice 500 voip
 description from UCM inbound call leg for call recording
 session protocol sipv2
 session target sip-server
 incoming called-number <Specific Dialled Number>
 voice-class codec 1  
 media-class 3
 dtmf-relay rtp-nte
 no vad
!

If you want to record both inbound and outbound calls you will need at least 4 dial-peers plus the extra ones to send to the subscibers.

In your example below you will need to split those into two dial-peers and for each dial-peer to go to a differant recorder.

Graham

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