SIP Trunk no service status reason local=0

[falomassor]

Hello,

I am trying to set up an SIP trunk to my cube (ISR4321) but it is always coming back as no service and the status reason is local=0. Cisco has not mentioned the local=0 in any of their documentation. I see reason 1,2,and 3 but not 0. Any help?

Thanks

[falomassor]

Hello Roger,

Here is the running config. Thanks

[Roger Kallberg · Accepted solution]

I would suggest that you do these changes to your configuration.

no ip host cucm01.fistservices.com 10.1.10.202
no ip host washington2.voip.ms 208.100.60.64
!
ip name-server <DNS server 1> <DNS server 2> ! add as many as needed
ip domain lookup source-interface GigabitEthernet0
!
no ip http server
no ip http secure-server
!
ntp source GigabitEthernet0
!
voice service voip
 ip address trusted list
  ipv4 <ITSP IP 1>
  ipv4 <ITSP IP 2> !if applicable, add as many as needed
  ipv4 <CM IP 1>
  ipv4 <CM IP 2> !if applicable, add as many as needed
 address-hiding
 mode border-element ! turns on Cube functionality, you need to reboot after to activate
 sip
  bind control source-interface GigabitEthernet0/0/0.10
  bind media source-interface GigabitEthernet0/0/0.10
!
voice class uri CM sip
 host ipv4:<CM IP 1>
 host ipv4:<CM IP 2> !if applicable, add as many as needed
!
voice class uri PSTN sip
 host ipv4:<ITSP IP 1>
 host ipv4:<ITSP IP 2> !if applicable, add as many as needed
!
voice class e164-pattern-map 1
 description E164 Pattern Map for called number to CM
  e164 <number range in CM 1>
  e164 <number range in CM 2> !if applicable, add as many as needed
 !
!
voice class e164-pattern-map 2000
 description E164 Pattern Map for called number to PSTN
  e164 [2-9]..[2-9]......
  e164 1[2-9]..[2-9]......
  e164 911
  e164 <more numbers to send to PSTN> !if applicable, add as many as needed
 !
!
voice class server-group 2000
 ipv4 <ITSP IP 1> preference 1
 ipv4 <ITSP IP 2> preference 2 !if applicable, add as many as needed
 description PSTN servers
 huntstop 1 resp-code 404 to 404
!
voice class server-group 1
 ipv4 <CM IP 1> preference 1
 ipv4 <CM IP 2> preference 2 !if applicable, add as many as needed
 description CM servers
 huntstop 1 resp-code 404 to 404
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
!
voice class sip-options-keepalive 1
 description Used for Server Group SIP OPTIONS PING
!
voice class sip-options-keepalive 2000
 description Used for PSTN SIP trunk Server Group SIP OPTIONS PING
!
dial-peer voice 1 voip
 description This dial peer is for incoming calls from CM
 no huntstop
 session protocol sipv2
 no session target ipv4:10.1.10.202
 incoming uri via CM
 no codec g711ulaw
 voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet0/0/0.10
 voice-class sip bind media source-interface GigabitEthernet0/0/0.10
 dtmf-relay rtp-nte sip-kpml
 no vad
!
dial-peer voice 2 voip
 description This dial peer is for outgoing calls to CM
 no destination-pattern [2-9]..[2-9]......
 destination e164-pattern-map 1
 session protocol sipv2
 no session target ipv4:208.100.60.64
 session server-group 1
 no voice-class sip early-offer forced
 voice-class sip options-keepalive profile 1
 no codec g711ulaw
 voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet0/0/0.10
 voice-class sip bind media source-interface GigabitEthernet0/0/0.10
 dtmf-relay rtp-nte sip-kpml
 no vad
!
dial-peer voice 3 voip
 description This dial peer is for outgoing calls to PSTN
 no destination-pattern 1[2-9]..[2-9]......
 destination e164-pattern-map 2000
 session protocol sipv2
 no session target ipv4:208.100.60.64
 session server-group 2000
 no voice-class sip early-offer forced
 no codec g711ulaw
 voice-class codec 1
 voice-class sip options-keepalive profile 2000
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 4 voip
 description This dial peer is for incoming calls from PSTN
 session protocol sipv2
 no session target ipv4:208.100.60.64
 no incoming called-number .
 incoming uri via PSTN
 no codec g711ulaw
 voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte
 no vad

Apart from this you need to set the IP on the SIP trunk in CM to 10.1.10.1 as that per your shared topology is the interface on the gateway that faces your CM. As previously stated you need to make sure that you can ping the CM from the gateway and the gateway from CM. If you cannot do that it doesn't matter what changes you do as then you have an issue that is related to network connectivity.

[falomassor]

Thanks turning on the mode border element command did the trick. Thank you very much

[Roger Kallberg]

Glad to hear that you have managed to solve your problem. However I doubt that it was that specific command that made it work for you. The reason being why I don’t think it’s related to that is because you can have a SIP trunk between CM and TDM type of gateway and then you’ll not using the gateway as a SBC, so turning on the Cube functionality is not necessary. Even so you can get the SIP trunk in CM to use SIP option ping to show the status of the trunk. If I were to venture into a guess I would be more acclaimed to say that it was related to missing bind statements on the dial peers and the fact that you had the wrong IP set on the trunk in CM.

[Carlo Poggiarelli]

To add a bit to what Roger correctly stated, the suspect that was an incorrect binded interface, comes from the screenshot of the output of the debug command you sent. There was a option ping message from cucm but no response from VG and that leads to a routing issue also because TCP is the selected transport protocol and the bind command should be enough to bring up the SIP socket on the VG.

My two cents 

Cheers

Carlo

[falomassor]

Thanks, Carlo for your help. I really appreciate it. 

Cheers,

[Carlo Poggiarelli]

Really glad to help!

Thanks a lot

Cheers

Carlo

[falomassor]

Hello Roger,

Yes the changes that I made for the config to bring the sip trunk up was changing the sip trunk IP to the gi0/0/0.10 IP and binding  the control and media source interface to gi0/0/0.10 and turning on the cube functionality.

Thanks a tone for your help.

[Roger Kallberg]

Thanks for confirming.

[Maren Mahoney]

Another amazing assist by @Roger Kallberg ! +5 if I could! -- Maren

[falomassor]

Hello MAaren,

Thnaks for your reply. I believe I configured the trunk right, but I drop some picture to see. 

[Carlo Poggiarelli]

Sorry @Maren Mahoney .

I replied to your post but my intention was to reply to the original one

TGIF

Carlo