PSTN call to MCU through CUCM / H323 gw drops after 10s

[Jean-Christophe Majchrzak]

Hi,

I've an Auto Attendant number configured on a MCU MSE 8510 for conferencing.

The architecture is

                    PSTN
                        |
                        |
                   H323 GW 15.3.3M
                        |
                        |

IP PHONE----CUCM 8.6.2
                        |
                        |
                   SIP TRUNK
                        |
                        |
                    VCS-C v7.1
                        |
                        |
                     MCU

I've other stuff around the VCS-C but they're not involved here.

I've created on the MCU a conference number 7006500 that is registered as H323 and SIP onto the VCS-C (Gatekeeper and SIP Registrar)

I've set up a route pattern on my CUCM that points to the SIP Trunk to VCS-C for that number.

For internal calls it works perfectly. We can have video on IP Phones but I've set the region settings to no video for the moment.

But when a call comes from PSTN it's disconnected after 10s. I can hear the conference prompt asking for conference number and PIN, if I'm quick enough I can enter the conference also, so DTMF works. Then hang up.

Something related to codec for sure. Disconnect cause 47 in debug isdn q931.

When I check "Media Termination Point Required" on the SIP Trunk then it works for calls from PSTN. But - there is a but - if I let this "Media Termination Point Required" checked, then internal calls to MCU (that were working before) are almost unable to use the DTMF to enter conf code and Pin. MCU does not understand or detects with the utmost difficulties the keys pressed.

On MCU all the possible codecs are checked.

We advertise G722 internally and incoming calls from PSTN have a voice class codec applied for the dial-peers with G711 A and mu.

Here are the debugs

debug h225 asn1

debug h245 asn1

debug voip ccapi inout

Any idea would be appreciated.

JC.

[Dennis Mink]

Jean,

Looking at the debugs the disconnect cause code is 47, which indicate issues with resources not being available (MTP/Xcoder) and or a codec mismatch. H323 uses fast start, looking at your debugs, comparable to SIP's early offer, which is why it sort of started working when you ticked MTP Required, on your SIP trunk configuration.

I can see from the debugs that the H245 set up messages are trying for  G711alaw

004865: Nov  7 22:53:37.124 CET: H245 FS OLC OUTGOING PDU ::=

value OpenLogicalChannel ::=
    {
      forwardLogicalChannelNumber 2
      forwardLogicalChannelParameters
      {
        dataType nullData : NULL
        multiplexParameters none : NULL
      }
      reverseLogicalChannelParameters
      {
        dataType audioData : g711Alaw64k : 20

have you verified the region in which the h323 gateway is and the region, in which the SIP trunk is, and what codec is being tried, logically, and also If there is XCODE available?

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[Jean-Christophe Majchrzak]

Hi Minkdennis,

Thanks for according time to my issue.

I confirm that Device Pool of H323 gw and SIP Trunk to VCS-C have these resources in this order : XCODE,  Harware MTP , Software MTP , Hardware CFB , Software CFB , MOH.

The Device Pools have different names and difference resources lists but for administrative purpose only. They target the same resources using PVDM3 / soft from cluster. These SCCP resources are acquired for sure.

Do you think H323 Fast Start can cause this issue ?

By the way there is no configured IOS command for H323 Slow or Fast Start. But on CUCM GW page "Enable Inbound FastStart" is checked.

If I set under Voice Service Voip -> h323 slow start , which one takes precedence over the other ?

I've also seen this document https://supportforums.cisco.com/docs/DOC-28882 that is somehow similar to my issue. About slow and fast start.

Would you suggest me to try with Slow Start ?

JC.

[Ayodeji Okanlawon]

Jean,

First of all you cant have fast start on h323 to Delayed offer on SIP...You need to have either slow start to delayed offer or fast start to EO on sip

Secondly, to get a clearer picture of whats going on you need to enable detailed traces on cucm and send them over. We need to see the H323-h2245 negotiation between the gateway and cucm and the sip signalling between the cucm and the VCS.

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[Jean-Christophe Majchrzak]

Hi

What kind of traces must I activate and extract ?

I suppose Call Manager detailed traces.

But for the export what is the selection ? I've a lack of knowledge in the debug part.

JC

[Jean-Christophe Majchrzak]

I succeeded in collecting CUCM SIP traces, this was explained there https://supportforums.cisco.com/docs/DOC-29963

I've also followed your document here https://supportforums.cisco.com/docs/DOC-27105

I've also used the http://translatorx.cisco.com/ to display the flow.

What's evident is that the incoming call arrives through my H323 GW1 (with fast start as you noticed), and later the ACK w/ SDP from CUCM tells the VCS-C to open the Media Stream between MCU and H323 GW2 (another PSTN gw not involved into the call setup)

Moreover how can the MCU (not known by CUCM, only VCS_C is configured) / H323 gw trust each other with a direct p2p stream ? Does CUCM allow it through its signaling ?

JC

[Jagpreet Singh]

Hi Jean,

          To answer your question:

> The VCS will tell the CUCM over SIP about the IP where the media needs to be terminated ( MU ) and the same will be forwarded over H.245 to the gateway.

Now, you said you were able to get the CCM traces, can you please share them here with the calling and the called number.

Regards,

Jagpreet

[Jean-Christophe Majchrzak]

Here are the traces.

Called number 7001511

Calling number is external and CLID restricted.

VCS has 10.3.239.2

MCU has 10.3.239.4

CUCM has 10.22.108.2

And for my question can someone clarify it (see below) ?

By the way there is no configured IOS command for H323 Slow or Fast Start. But on CUCM GW page "Enable Inbound FastStart" is checked.

If I set under Voice Service Voip -> h323 slow start , which one takes precedence over the other ?

JC

[Ayodeji Okanlawon · Accepted solution]

I have looked at the edited trace file you sent and this has confirmed that you are doing fast start on H323 (even though I didnt see the OLC before the h225 setup. I assume its because you have edited the trace file) to Delayed offer on SIP. You need to either remove the inbound fast start on CUCM and configure slow start on your gateway or enable Early offer on the SIP trunk to the VCS (NB you will need an MTP for this)

To answer your question....

The default on H323 gateway is fast start. To enable fslow start on the gateway you will  need to configure it under voice service voip or at the dial-peer level...

voice service voip

call start slow

or

voice class h323 1

h225 timeout tcp establish 3

call start slow

You then apply the voice class h323 command to the dial-peer pointing to cucm.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Jean-Christophe Majchrzak]

I've done 2 tests

1) Work on SIP config

SIP Trunk -> Media Termination Point Required checked + DTMF Signaling Method -> No Preference

SIP Profile -> Early Offer support for voice and video calls (insert MTP if needed) checked

Result : now it works for incoming calls through H323 gw (fast start), but internal calls are failing entering DTMF for audioconf number and Pin. I'm using 89xx SCCP and 99xx SIP phones with up to date fw. One issue is solved but another appears...

I'll have a look at your doc (you again !) https://supportforums.cisco.com/docs/DOC-33944 and try to find out why.

Have you any idea of the root cause ?

1) Work on H323 gw config

Keep SIP Trunk with Delayed offer.

call start slow applied on H323.

And...IT WORKS !!

Thanks.

JC