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CUCM Recording with secure SIP Trunk to CUBE

Djeten
Level 1
Level 1

I have an issue with our recording setup.

I have created the recording profile and secure sip trunk to our cube. 

When I make a call, I see the call coming in, but the call is not sent to the recorder.

When I do 'debug voice dialpeer all' I see that the outgoing dial peer is matched, but my incoming dialpeer is not matched.

Result=NO_MATCH(-1) After All Match Rules Attempt

My incoming dial peer is configured with 'incoming called-number .' so it should match any incoming call.

I have also tried to match the dial peer on the ip of the cucm, but I have the same problem then.

Any ideas what else I can do?

 

Kind regards

Didier Derveaux

17 Replies 17

I concur with @b.winter there is no usable information in the shared output from the debugs. It just contains an inbound invite and then basically no more, only a SIP option ping dialog. With this it's not very useful.

Can you please check that your dial peers are in an operational state with "show dial-peer voice summary", see the below as an example from one of our system.

selurt-cube-01#sh dial-peer voice summary 
dial-peer hunt 0
             AD                                    PRE PASS SESS-SER-GRP\  OUT 
TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT    KEEPALIVE    VRF
1000   voip  up   up                                0  nse                                           NA                  
1010   voip  up   up             map:1              0  nse  SESS-SVR-GRP: 1               active     NA                  
100    voip  up   up                                0  syst                                          NA                  
110    voip  up   up             map:2000           0  syst SESS-SVR-GRP: 2000            active     NA                  
 For server-grp details please execute command:show voice class server-group <tag_id>
 To see complete session target for ipv6 use 'sh running-config | section dial-peer <tag>
selurt-cube-01#

Explanation of the dial peers in the list, 1000 is the inbound from CM, 1010 is outbound to CM and 100 is inbound from a ITSP and 110 is outbound to the ITSP.



Response Signature


There are a few things that sticks out in your configuration. I would suggest that you try with adding this.

 

voice service voip
 ip address trusted list
  ipv4 x.x.x.x
  ipv4 y.y.y.y

voice class sip-options-keepalive 1
 description Used for Server Group SIP OPTIONS PING

no voice class uri 101 sip

voice class uri CUCM sip
 host ipv4:x.x.x.x
 
voice class uri REC sip
 host ipv4:y.y.y.y

dial-peer voice 8000 voip
 description Outbound dial peer to recording system
 destination-pattern 8888$
 voice-class sip options-keepalive
 codec g711ulaw

dial-peer voice 1000 voip
 description Inbound dial peer from CM
 no voice-class sip options-keepalive
 no incoming called-number .
 incoming uri via CUCM

dial-peer voice 2000 voip
 description Outbound dial peer to CM
 destination-pattern 2...$
 no voice-class sip options-keepalive
 voice-class sip options-keepalive profile 1 
 codec g711ulaw

dial-peer voice 8100 voip
 description Inbound dial peer from recording system
 session protocol sipv2
 session transport tcp tls
 incoming uri via REC
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte
 srtp
 codec g711ulaw
 no vad

 

Also I would recommend you to upgrade IOS to something more recent than 16.9. That's back from the pre-color TV era.



Response Signature


Djeten
Level 1
Level 1

OK Thanks, I will add this to the configuration. I am not on site now, but I will update you when I have done this.