SIP to H323 to SCCP Call are not connecting after IVR configured on Cisco Voice Gateway 3825

[ahmed-mujeeb]

Hi Everyone,

I have set a SIP trunk from a ITSP provider on Cisco 3825 router, It sas successfully registered with ITSP.

1) we can successfully made a call form internal network to PSTN.

2) But when trying to call from PSTN to internal network, IVR played but when we dialed the desired extension the ring continuous to go, hence the IP Phone rings but when anyone pick up there was no voice and that time the caller is continuously listening the ringing tone. IVR is ios based on Cisco router.

My call flow is as follows,

PSTN ==> SIP(ITSP) ==> H323(VG to CUCM) ==> SCCP(IP Phones)

Kindly Help me,

[Ayodeji Okanlawon]

My first advise to you is to use sip trunk between CUCM and gateway. SIP to SIP works better than SIP to H323 and its easier to troubleshoot.

We can troubleshoot your current setup, but I strongly advise to migrate your h323 gateway to sip

[ahmed-mujeeb]

Thanks for the advise Ayodeji,

But the we are using current setup with PRI and we take the SIP trunk for testing so I am testing on the production enviornment,

So can you kindly help me in the current setup..

[Ayodeji Okanlawon]

OK..

please do the ff (enable logging)

service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit

Then..

<Enable debugs, then test again.>

debug ccsip mesages

debug voip ccapi inout

debug h225 asn 1

debug h245 asn 1

<Enable session capture to txt file in terminal program.> (such as Putty)

then do the ff:

terminal length 0
show logging

2. Attach sh run of your gateway

Attach the logs and the sh run (include calling, called and time of call)

[ahmed-mujeeb]

Attached are the show running config and show logging..

calling number = 34536571

called number PSTN = 37131444

called number (ip extension) = 2625

time of call = Sep 28 16:25

[Ayodeji Okanlawon · Accepted solution]

Hi,

I have looked at your logs and here is my observation..

+++ Your ITSP sends call using EO (Early offer) ++

Since you have a SIP to H323 connection, the gateway is going to try and do FS (fast start) to CUCM. FS is not configured by default in CUCM and hence CUCM doesn't respond tot he FS request,,

+++Here we see the gateway sending FS to CUCM++

485572: Sep 28 16:25:25.406: //369488/6570095AA376/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=2002
485573: Sep 28 16:25:25.410: H245 FS OLC OUTGOING PDU ::=

value OpenLogicalChannel ::=
    {
      forwardLogicalChannelNumber 1
      forwardLogicalChannelParameters
      {
        dataType audioData : g711Ulaw64k : 20
        multiplexParameters h2250LogicalChannelParameters :
        {
          sessionID 1
          mediaControlChannel unicastAddress : iPAddress :
          {
            network 'AC140B32'H
            tsapIdentifier 18473
          }
          silenceSuppression TRUE

Further down since CUCM doesn't respond to FS, the gateway then attempts to send a new setup using slow start...But SIP early offer to h323 slow start is not supported..

To resolve this there are two possible solutions

1. Configure a separate dial-peer to cucm for test purposes as follows

dial-peer voice 2625 voip
 destination-pattern 2625
  session target ipv4:172.20.11.10

sess proto sip
 voice-class codec 1 
  dtmf-relay rtp-nte sip-k

no vad

1b.. Next configure a sip trunk between cucm and your gateway

1c. Test again..

2. The second solution will be service impacting. You will need to enable inbound FS on your h323 gateway in CUCM. You will need an MTP for this and you will need to select your originating codec to be g711u.

This is why I suggested using a sip trunk earlier. Due to the operability issues between sip and h323...

[ahmed-mujeeb]

hello Ayodeji,

I have tried the method 1 by creating dial-peer 2625 but the same issue persists, can you please elaborate the second method how can i enable inbound FS? I already have configured MTP in ios as you saw in the running config. do I need more to do?

[ahmed-mujeeb]

Thanks Ayodeji,

Issue has been resolved.

I only enable the inbound FastStart on My H323 Gateway and connection establishes..

Thanks again for all your efforts and support Ayodeji,

[ahmed-mujeeb]

Hi Ayodeji,

enable fastStart resolves my issue but now i am facing random call disconnect issue.

[Ayodeji Okanlawon]

Is the disconnect happening on inbound or outbound calls? 

[ahmed-mujeeb]

Disconnect is happening in inbound calls.

[Ayodeji Okanlawon]

Okay..Can we do this..

I am proposing this because its much easier to troubleshoot.

We need to go back to the SIP option..

1. Configure the dial-peer I suggested earlier.

dial-peer voice 2625 voip
destination-pattern 2625
session target ipv4:172.20.11.10
sess proto sip
voice-class codec 1
dtmf-relay rtp-nte sip-k
no vad

2.Ensure you have your sip trunk configured..

3. On the sip trunk, there is a sip profile applied.. Go to device>settings>sip profile..Click on find and then click on the (standard sip profile..if this is the one used)

Scroll down to trunk specific configuration and change the SIP rel1xx to send PRACK if 1f 1xx contains SDP as shown below..Save it..Then go back to the sip trunk and reset it

Turn on debugs..

debug ccsip messages

Now do a test call and transfer to 2625..

If it doesn't work..Please send me the debugs (attach here)

[ahmed-mujeeb]

Hi Ayodeji,

Sorry for the late reply,

Actually i confirmed from the organization, they are experiencing call disconnects on outbound calls.

[Ayodeji Okanlawon]

Did you enable outbound fast start on your h323 gateway. If you did, then you need to enable early offer on your cube gateway. 

On the dial peer to your ITSP add

Voice-class sip early offer forced. 

[ahmed-mujeeb]

I just configured the sip early offer on dial peer, need some time to confirm this.

Second issue we have an operator where all calls land, the issue is happening frequently that when the operator transfer the inbound call to any extension it dropped, and these calls are coming from our PRI (POTS).