Cisco CSR1000v blocks all outbound data

[ahaight10]

Hello,

My CSR1000v blocks all outbound data, For example I can hear a caller but they can't hear me. I also can't ping things like google

My config is attached below

[Ritesh Desai]

Hi ahaight,

I just checked the configuration and what it looks is,

1). There is 1 incoming voip dialpeer.
2). For One way audio, i see there is 64.16.240.36 which i assume is your SIP provider IP address. Please note if the SIP provider IP is routable using ip route command you have defined.
3). You cannot ping google because you have not defined domain which can resolve google ip address OR your router is not in domain. To do so add #ip domain name cisco.com
3). Please follow CUBE standard practices to avoid issues. You need to create new outgoing dial-peer I don’t see outgoing dial-peer pointing for outbound calls.

4) Recommend you to add this to your dial-peer pointing to SIP PROVIDER.

#voice-class sip options-keepalive - This command will help you to exchange SIP OPTIONS. Like in data network we execute ping to identify the response from far side, same like in SIP its called OPTIONS.

# show dial-peer voice summary - This command will help you to monitor SIP TRUNK status. Refer to KEEP-ALIVE column. If you see BUSY OUT that means sip trunk is DOWN, if you see ACTIVE means sip trunk is UP.

regards,

Ritesh Desai

please rate helpfull post.

[Vaijanath Sonvane]

Hi,

Please try below configuration. You need to make highlighted changes according to your environment:

!
voice call send-alert
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 192.76.120.10
  ipv4 64.16.240.36
  ipv4 192.168.15.21
 mode border-element
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 supplementary-service media-renegotiate
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  bind control source-interface GigabitEthernet4
  bind media source-interface GigabitEthernet4
  header-passing
  error-passthru
  options-ping 60
  no update-callerid
  early-offer forced
  midcall-signaling passthru
  privacy-policy passthru
!
voice class codec 1
 codec preference 1 g711ulaw
!
voice class server-group 1000
 ipv4 10.10.200.21 preference 1
 ipv4 10.10.200.22 preference 2
 ipv4 10.10.200.23 preference 3
 description ** CUCM SERVER GROUP **
!
voice class server-group 1001
 ipv4 12.25.15.11 preference 1
 ipv4 12.19.12.22 preference 2
 ipv4 12.25.17.13 preference 3
 description ** SERVICE PROVIDER SIP SERVER GROUP **
!
voice translation-rule 100
 rule 1 /18454041836/ /3005/
!
voice translation-profile XLATE_INCOMING_DID
translate called 100
!

dial-peer voice 1000 voip
 description ** INBOUND CALLS FROM CUCM CLUSTER **
 session protocol sipv2
 session server-group 1000
 incoming called-number 1[2-9]..[2-9]......
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!
dial-peer voice 1001 voip
 description ** OUTBOUND CALLS TO CUCM CLUSTER **
 destination-pattern 3...$
 session protocol sipv2
 session server-group 1000
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!
dial-peer voice 9000 voip
 description ** INBOUND CALLS FROM SERVICE PROVIDER **
 translation-profile incoming XLATE_INCOMING_DID
 session protocol sipv2
 session server-group 1001
 incoming called-number 845.......$
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!
dial-peer voice 9001 voip
 description ** OUTGOING 911 CALLS TO SERVICE PROVIDER **
 destination-pattern 911
 session protocol sipv2
 session server-group 1001
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!
dial-peer voice 9002 voip
 description ** OUTGOING CALLS TO SERVICE PROVIDER **
 destination-pattern 1[2-9]..[2-9]......
 session protocol sipv2
 session server-group 1001
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!

[ahaight10]

Thank you, I will try this out. 

[ahaight10]

Hello,

With everything applied, I still get error code 407 Proxy Authentication Required when making calls.

[Ritesh Desai]

Hi,

407 Proxy Authentication Required is because the SIP telco has enabled authentication parameter on SIP trunk to allow making calls. If the credentials are matched then Its allowed to make call.

Verify the SIP TRUNK status using show dial-peer voice summary. The dial-peer will show busyout means Sip trunk is down.

You need to ask telco for this details;

1. Username
2. Password
3. SIP REALM Domain
4. SIP REGISTRAR (ip / fqdn)

Once you get the above details;

You need to configure above parameters in CUBE aka Cisco SBC.

Under sip-ua
1). Set registration #credentials username password realm
2). Set authentication #authentication username password realm
3). #registrar (use help ? for supported paramaters)

Verify whether CUBE is registered with SIP SP domain,
#sh sip-ua register status

To Troubleshoot registration and authentication;
#deb ccsip messages
#deb ccsip non-call

Thanks and regards,
Ritesh Desai
Please rate helpful post.

[ahaight10]

The trunk is registered... Please find my debug attached. 

[Ritesh Desai]

EDITED POST****

@ahaight10  So have you setup Authentication in sip-ua? If Yes, have you tested and does call goes through? If no, can you attach debugs logs along with show run of your configuration please?

#debug voip dialpeer inout

#show dial-peer voice summary

#deb ccsip messages

#deb voip ccapi inout

Share the called number, calling number and timestamp.

Please rate helpful post.

[ahaight10]

Hello,

For some reason, PUTTY isn't taking my password. I will troubleshoot this first then I will get back to you!