[CUCM 9.x] No dial tone when calling in

[Dmitry Golovenkin]

I have a fairly new setup with CUCM and CUC running 9.1 version. Calls are going to the CUBE via SIP trunk from the provider, then they get sent via another SIP trunk to the CUCM, which then forwards it to the Call Handler in CUC (for business hours and holiday reasons). CUC then redirects the calls to Hunt Group 600 to ring the required phones.

Now when the some makes a call to the system from outside does not hear any ringing. However if the person is to call a someone's phone directly then the tone is there. DDI and main number use the same Route Pattern for translation and I have the outside dialtone ticked on the CUBE config as well as the Hunt Group.

Am I missing something?

[kelvin.blair]

What does the cube dialpeers look like?  You might be needing this on the voip dialpeer, but without seeing your dialpeer configuration for these calls it might be hard to tell.  You can try this and see if it works.

progress_ind setup enable 3

progress_ind progress enable 8

[Dmitry Golovenkin]

Hi Kevin,

This is my config of the dial-peers.

dial-peer voice 1 voip
 description **Incoming Calls**
 destination-pattern 441525......
 session protocol sipv2
 session target ipv4:192.168.13.101
 voice-class codec 1  
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
dial-peer voice 10 voip
 description **Outgoing Calls**
 destination-pattern 0.T
 session protocol sipv2
 session target sip-server
 voice-class codec 1  
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
dial-peer voice 2 voip
 description **Incoming Calls**
 destination-pattern 441908787...
 session protocol sipv2
 session target ipv4:192.168.13.101
 voice-class codec 1  
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
dial-peer voice 100 voip
 description **Inbound dial-peer for Outbound calls**
 session protocol sipv2
 session target sip-server
 incoming called-number 0.T
 voice-class codec 1  
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
dial-peer voice 101 voip
 description **Outbound Emergency calls**
 session protocol sipv2
 session target sip-server
 incoming called-number 0...
 voice-class codec 1  
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad

Also, what you suggested made no difference.

[kelvin.blair]

Try using the commands I suggested on your incoming dialpeer.  If that is unsuccessful, please provide a sho run excluding any passwords.  Also, please provide the following debug's:

debug ccsip messages

debug ccsip calls

debug voip dialpeer inout

Thanks,

[Dmitry Golovenkin]

Please see attached debug. Thanks for your help so far but the commands for DPs didn't do any difference.

[OberonWnv]

This is my config of the dial-peers.

Покажи dial-peers pots

[Dmitry Golovenkin]

У меня только sip. Проблема устронена, спасибо.

[Dmitry Golovenkin]

Here is the show run with sensitive info removed.

[kelvin.blair]

First Suggestion:

Your matching incoming dialpeer of 10 and outgoing dialpeer of 2. 

Dialpeer 2 add

incoming called XXXXXX <<<--should match destination-pattern

progress_ind setup enable 3

progress_ind progress enable 8

see if that helps..

provide debugs again if no go..

[Dmitry Golovenkin]

Sorry, makes no difference.

[Dmitry Golovenkin]

Here is the debug. Many thanks

[kelvin.blair]

Going out on a limb considering i can't look at your CUCM trunk configuration..

On your SIP profile in CUCM, do you have disable early media 180 checked?  try checking this to see if it helps or unchecking it. One thing that is really bothering me as well is this statement on the dialpeer to CUCM.. By default CUCM is delayed offer. 

voice-class sip early-offer forced is not needed to CUCM.  It wasn't until 9.X that early offer is supported, but you need Early Offer support for voice and video calls (insert MTP if needed) enabled to do otherwise you do it at the cube level as you are doing for outbound calls..

[Dmitry Golovenkin]

Hi Kevin,

Sorry but none of these seem to make any difference.

1) I tried checking "Disable Early Media on 180" and making a call in, nothing - just silence but the internal phones ring.

2) Then I tried taking out "early-offer forced" as well which made no difference either.

Do I need to have 2 dial-peers on the CUBE, one for accepting the call from the VoiP provider and the second one for passing this call to the CUCM? At the moment I only have the one.

[kelvin.blair]

Do you have an MRGL assigned to the trunk that contains a ANN..? 

Also, bind you sip traffic to a single interface.  this can cause issues as well. 

For example:

voice service voip

sip

     bind control source-interface GigabitEthernetx/x

     bind media source-interface GigabitEthernetx/x

[Dmitry Golovenkin]

Sorry, no clue what the top bit means?

When I bind sip to Port-Channel 1.1 (which is my VLAN1), I get no audio when making a call.