SIP CUBE dial-in problems

[Paul Austin]

Hi Folks, I'm having a real problem with migrating to a new ISP for a customer. The original one works fine but he new one just doesnt want to play ball and I'm at a loss to be honest. So, outbound is fine but inbound is not.

Call made from PSTN, through CUBE and phone on UCM rings, if I answer it, the call is still ringing on the PSTN phone, the call duration time is increasing on the UCM phone then eventually it clears down.

If I make the same call from the PSTN but le it ring without answering, the UCM phone rings for a while, stops but I still get rininging on the PSTN phone for a while before that is cleared down.

What I can see from the CUBE - debug ccsip messages output is that I get a 504 Gateway Timeout

Any ideas?

I have attached the relevant debug file.

Thanks in advance

[Wilson Samuel]

Hi Paul,

just wondering if you have MTP configured (software or hardware based) and assigned to the correct MRG / MRGL for the SIP Trunk.

[Paul Austin]

Hi, Yes I have and have tried both varients.

Thanks

[Carlo Poggiarelli]

Hi Paul.

Under sip-ua try to add disable-early-media 180

HTH

Regards

Carlo

[Paul Austin]

No sorry.

[Mohammed al Baqari]

Hi,

You have 100Rel enabled in your network and your ITSP isn't responding with PRACK (probably not supported) which is causing the call leg with ITSP to timeout.

You need to disable 100rel feature wherever it is configured. If you share your config of the CUBE, it will be very useful for us to look.

Here are the steps on how to disable it.

In CUBE (at dialpeer level or globaly)

dial-peer voice 1 voip

no voice-class sip rel1xx (require | support) "100rel"

 session protocol sipv2

OR

voice service voip

 sip

 no rel1xx (require | support) "100rel"

[Paul Austin]

No sorry that didnt work

[Mohammed al Baqari]

In this case, disable it in your CUCM:

1. In the CUCM Administration interface, go to Device > Device Setting > SIP Profile
2. Open the SIP profile used by a given SIP trunk.
3. Choose Disabled from the SIP Rel1XX Options drop−down list.
4. Reset the SIP trunk profile for the given SIP trunk

[Paul Austin]

Nope sorry

Thanks though.

[Mohammed al Baqari]

Ok .. Please keep these parameters off as I suggested and send the latest debugs.

[Paul Austin]

Debug posted.

Thanks

[Paul Austin]

One thing that may be the problem is that on the initial invite from the ISP I have 1 M channel but on the invite from the CUBE to the UCM I have 2 ??

[Mohammed al Baqari]

I can see 100Rel is disabled now in UCM but not in the CUBE.

In you CUBE, please apply the following config:

voice service voip
 sip
  rel1xx disable

If still not working, please share the latest debug after applying the command.

[Paul Austin]

Nope not good

Thanks

[Mohammed al Baqari]

Hi Paul,

Now the 200OK is passing to your ITSP. So we resolved the problem of 100Rel. However, I see that timer header is required in 200OK message which isn't supported by your ITSP and it is getting a response.

I am looking to fix this and will get back to you.

On a side note, there is a mistake in the config. In dialpeer 5.

The dialpeer is using source interface as g0/1 and matching inbound calls from both UCM and ITSP which isn't correct. Your ITSP needs interface g0/2 as source.

A quick fix for this error is to take off the voice-class bind command from dialpeer 5.