Solved: SIP Trunk no service status reason local=0

falomassor · ‎10-26-2023

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

Roger Kallberg · ‎10-28-2023

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.

View solution in original post

Maren Mahoney · ‎10-27-2023

It would imply a local configuration issue. Can you provide a screenshot of the Trunk configuration?

Maren

Carlo Poggiarelli · ‎10-27-2023

Hi

Local = 0 usually means that there is no Socket opened on the remote site on configured Port / Protocol or there is a firewall blocking IP communications between them.

Please check on your VG if you configured the right listening interface and activated the SIP protocol under Voice Service Voip section.

Please let us know

Regards

Carlo

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falomassor · ‎10-27-2023

Hello there,

I thought that was for local=2 https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/213718-calls-through-session-initiation-protoco.html#:~:text=Scenario%202.%20For%20Local%3D2%2C%20possible%20reason%20could%20be%20that%20Uni....

But I have no firewall between the cucm and the voice gateway

Carlo Poggiarelli · ‎10-27-2023

Hi,

Not only 2. On some version has been observed also as 0.

Anyway, if on the defined SIP profile you enabled Ping Option, please activate a debug ccsip message on your VG and see if you receive Option Ping messages from CUCM and verify the response from your VG.

If you can, please upload the result here.

Thanks

Regards

Carlo

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falomassor · ‎10-27-2023

Thanks for your response. Here is the output for the debug ccsip messages.

Carlo Poggiarelli · ‎10-27-2023

Perfect!

Thank you for the output.

As you can see, VG is not sending 200 OK as response.

Please post the entire VG config so we can look into it and check where we can fix it

Thanks again

Cheers

Carlo

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falomassor · ‎10-27-2023

Hello Carlo,

Here is the running config.

Thanks

Carlo Poggiarelli · ‎10-27-2023

Ok,

We are missing some important parts of the config

Let’s begin with

conf t

voce service voip

sip

bind all source-interface GigabitEthernet0/0/1

end

See if the trunk goes up on cucm side

Thanks

Cheers

Carlo

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falomassor · ‎10-27-2023

I made the change but the trunk did not go up. Here is my topology.

Roger Kallberg · ‎10-27-2023

There are a number of issues with your gateway configuration, I’ll get back on that in a later response once I’m in front of a computer so I can properly work with your shared configuration. But first off in CM you cannot have the IP that faces your service provider set as the destination/source IP on the trunk. You’ll need to use the IP of the inside interface of your gateway.

Can you even ping the CM from the gateway and vice versa from the CM? That’s number one in making this work.

Your whole topology is somewhat of a puzzle to me. For one why do you have the same IP network on the management interface and the interface to your service provider? Also for what reason are you using sub interfaces on the network facing your inside network?

Carlo Poggiarelli · ‎10-27-2023

Ok,

Let’s try a ping from the vg to the cucm using :

ping 10.1.10.202 source 192.168.1.3

than config session transport tcp on dialpeer 1

dial-peer voice 1 voip

session transport TCP

I noticed that you have a directly connected interface on the CUCM’s subnet…. why don’t you change the ip on the sip trunk configuration from 192168.1.3 to 10.1.10.1 ?

You can add the following line to dial-peer 1

voice—class sip bind all source-interface GigabitEthernet0/0/0.10

Please let me know

Regards

Carlo

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falomassor · ‎10-27-2023

Hello Carlo.

I got the same result. No service and reason is local=0.

Carlo Poggiarelli · ‎10-27-2023

Hi,

Please make these two changes

voice service voip
ip address trusted list
ipv4 10.1.10.202 255.255.255.255

dial-peer voice 1

session target ipv4:10.1.10.202:5060

Please attach here the output of a Show sip-ua status.

Thanks again

Regards

Carlo

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Roger Kallberg · ‎10-27-2023

Without seeing the configuration of your voice gateway it will be hard to give you a detailed answer, but a guess based on previous experience would be that it could be related to the built in security feature in IOS. It would not allow traffic from an IP that isn’t present in dial peer configuration. There are a couple of options for how to handle this, but without knowing how your configuration looks like it would only be a guess work. Can you please post your entire running configuration?