Migration E1 to Sip --- cannot make or receive calls

[sabams08]

We just migrated from E1 to Sip but, we cannot make or receive calls

Incoming calls ring on our extension, when you answer the call it will not connect and it will continue to ring at the other end. I reported this to the provider, they said no acknowledgment is not received at their end that we should check our config.

Also, they said sip was configured as interworking at their end., so credentials are not required.

Have tried all I can with no success. I will appreciate any help I can get.

See attachments for logs.

[Nithin Eluvathingal]

incoming calls ring on our extension, when you answer the call it will not connect and it will continue to ring at the other end. I reported this to the provider, they said no acknowledgment is not received at their end that we should check our config.:- is there a  firewall some where  in between the gateway and CUCM ?

2.12.0.0

[sabams08]

No firewall. the VoIP in running on a CME

[Mohammed al Baqari]

In addition check if they are expecting PRACK for their ring messages and
enable it.

See this doc to enable PRACK.

https://www.cisco.com/c/en/us/support/docs/voice/session-initiation-protocol-sip/116086-configure-cube-cucm-sip-00.html

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[Roger Kallberg]

Please share your running configuration for us to verify it. Also please share the output from both debug ccsip message and debug voip ccapi inout in the same file, one per direction of the call, ie one file for inbound and another for outbound.

[sabams08]

Have attached the running-config and logs

[sabams08]

outgoing logs

[Roger Kallberg]

Looks like you did not have both debugs active at the same time. Can you please redo the calls and repost?

[Roger Kallberg]

Looks like you’re running config is incomplete. Can you please repost the entire configuration?

[sabams08]

running config

[Roger Kallberg]

I would recommend you to clean up your configuration of dial peers, from what I can make out you should only need to keep 1111 and 1000. For dial peer 1000 your missing the bind statement and it has a session target definition, that’s not needed on an inbound dial peer as that’s an outbound option. Also your SIP-UA configuration should be altered to cater for the need of SIP CME phones, the sip server command is not required. SIP CME phones need the gateway to act as a SIP registrar. Search for this on the community and I’m sure you’ll find what you need.

For inbound VOIP dial peers my general advise is to if possible use information in the VIA header to match the dial peer. Information on this and many more tidbits on dial peer configuration can be found in this superb document. https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html

[sabams08]

I'm yet to resolve this.

iI saw this in the log "resending 180 response due to no PRACK" 

what is the meaning

3: //37125/F5C5069A9EB1/SIP/Info/info/4096/sipSPIResendInviteResponse: resending 180 response due to no PRACK

[Roger Kallberg]

You got an answer from @Mohammed al Baqari with details about PRACK. Suggest that you read that and use the information provided. Once done and if it still does not work please repost the debugs as requested earlier, with both of them running simultaneously.

[sabams08]

The issue has been resolved

The problem was the port I'm using to connect to the provider, my nat was applying dynamic port, they advised I to connect to them using port 5060

Adding this to my config solved the problem

registrar ipv4:xx.xx.xx.xx:5060

sip-server ipv4:xx.xx.xx.xx:5060
connection-reuse

Thank you, guys. I really appreciate your help

[Roger Kallberg]

Glad to hear that you have solved your problem. FYI Adding :5060 after the FQDN on these commands has the actual effect in IOS to stop the router from trying to do an SRV DNS lookup. Not saying that it wasn’t your NAT configuration that was at fault with this, only to provide you with the complete details on what your changes does.