Outbound Call through FXO terminated with code 47

[Remon Adel]

Dears ,
CUCM (11.5) ====H323 GW (router 4331with no PVDM  ) FXO card with it's own resources ===PSTN

when we make outbound call for example dial my cell Phone ,it is ringing but after 10s call  terminated ,
also if i answered the call ,it  disconnected after this period 10s.

the configured dial peer between gateway and CUCM is used default Codec g729. The region between IP phone and Gateway use codec g729 . So the codec of call is g729.

Also w try to force call to use g711 but same issue

No transcoder is configured on this GW ,we don't have PVDM. also in this scenario we don't need it as FXO modem use it's resources to decode medial . and calls between IP phone and GW will be g729 or g711 .

This Cause code (47) indicates that no resources found ,We need to clarify this issue as we have the same scenario on another site and calls succeed .

Kindly find attached output of these  debugs  during this issue

debug voip ccapi inout
debug h225 asn1
debug h245 asn1
debug cch323 all
debug ip tcp transaction

[R0g22 · Accepted solution]

Does not look like a DSP issue. I see the following -

*/ Connect sent by GW with H245 IP and port -

Apr 12 19:00:49.640 UTC: H225.0 OUTGOING PDU ::=

value H323_UserInformation ::=
{
h323-uu-pdu
{
h323-message-body connect :
{
protocolIdentifier { 0 0 8 2250 0 4 }
h245Address ipAddress :
{
ip '0A067D81'H
port 22007
}

*/ At this point, CUCM needs to establish a TCP connection with that IP and port that the GW sent previously.
You already had TCP logs enabled but I don't see any TCP handshake happening.

*/ CUCM sends a disconnect post that -

Apr 12 19:01:01.309 UTC: H225.0 INCOMING PDU ::=

value H323_UserInformation ::=
{
h323-uu-pdu
{
h323-message-body releaseComplete :
{
protocolIdentifier { 0 0 8 2250 0 5 }
callIdentifier
{
guid '0032A1CCDDACF1AC0100E4050A067D86'H
}
}
h245Tunneling FALSE
}
}

Since you see a disconnect at a stage where the IOS FSM was waiting for caps via H.245, you see a CV=47.

Can you please your config here ? Was this working fine before ? If yes, what changed ?

[Remon Adel]

Hello Nipun ,
kindly find attached current config of H323 section ,please note that it's fresh install not worked before .

Thanks

[Remon Adel]

As per below logs , TCP connection established  but CAP negotiation not happened  .

Please note that there's FW between GW and CUCM ,May this cause this issue ???

Apr 12 19:00:45.734 UTC: TCB7FD34E66F980 created

Apr 12 19:00:45.734 UTC: TCP0: state was LISTEN -> SYNRCVD [1720 -> X.X.X.X(49778)]

Apr 12 19:00:45.734 UTC: TCP: tcb 7FD34E66F980 connection to X.X.X.X:49778, peer MSS 1360, MSS is 516

Apr 12 19:00:45.734 UTC: TCP: Selective ack is disabled from the CLI

Apr 12 19:00:45.734 UTC: TCP: sending SYN, seq 4067474583, ack 3681303653

Apr 12 19:00:45.734 UTC: TCP0: Connection to X.X.X.X:49778, advertising MSS 536

Apr 12 19:00:45.737 UTC: TCP0: state was SYNRCVD -> ESTAB [1720 -> X.X.X.X(49778)]

Apr 12 19:00:45.737 UTC: //-1/xxxxxxxxxxxx/H323/cch323_ct_main: SOCK 0 Event 0x1

Apr 12 19:00:45.737 UTC: TCB7FD33C7FBA58 accepting 7FD34E66F980 from X.X.X.X.49778

[R0g22]

With H.323 there will be two TCP handshakes and both will remain active for the duration of the call.
This one is for destination port 1720 from CUCM sourcing using an ephemeral port 49778. After this you would see H.225 messages being exchanged.

What FW do you have ? Is it doing H.323 inspection ?

[Remon Adel]

Hello
I noticed that GW negotiates h245 with port 22007  but this out of ephemeral range (32768-65535).

So why the h.323 gw negotiates the h245 info with out of range ports ?

          h245Address ipAddress :

          {

            ip '0A067D81'H

            port 22007

          }

          callIdentifier

Unfortunately , i don't have info about FW but i think TCP ephemeral  ports have been opened .

[R0g22]

Where did you find that range from ? Last I remember the ranges are -

IOS voice gateway : from 11000 to 65535
Cisco Call Manger : from 1024 to 65535

The docs are incorrect and I remember a couple of doc defects that were filed to fix that. Not sure if they ever were.

You need to take access of the FW and do a packet dump on the ingress or egress.

[Remon Adel]

As per this cisco docs ,The ephemeral port range for the system (CUCM11.5) is 32768 to 61000

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1/sysConfig/CUCM_BK_SE5DAF88_00_cucm-system-configuration-guide-1151/CUCM_BK_SE5DAF88_00_cucm-system-configuration-guide-1151_chapter_01010100.pdf

But after review again ,i found this note ((For IOS gateways,the H.245 port range is from 11000 to   65535)) from CUCM to Gateway and Vice versa .
So what about the range you posted
Cisco Call Manger : from 1024 to 65535

Do you mean that the required ports for connection  from CUCM to GW  are (11000 to   65535 ) and ports for connection from GW to CUCM  are  (1024 to 65535) ???
Please confirm .
Thanks

[R0g22]

So basically these are source ports wherein CUCM and/or GW tells the other end what port each will use to listen on H.245
Effectively, even if the range does not overlap it should not cause issues because they are both locally generate and significant. If IOS or CUCM does not list support for a particular port range for H.245 does not mean that they would not establish a TCP/exchange messages. The logic would be the same as that of RTP ports. Locally significant only.

Now, I have not seen CUCM using 1024 or a port lower than 10000 being used. I would suggest you opening up the entire range i.e. 11000 - 65535 on your firewall for traffic b/w GW and CUCM.

Also, do you traverse a WAN link for this ? Is their a WAN optimiser present ?

[Remon Adel]

Hello Nipun Singh Raghav
The issue has been fixed after we permit these range (11000 - 65535 ) between CUCM and GW.
But now we need to convert GW from H323 to SIP gateway ,what is  port range will be required  between CUCM and SIP gateway ?

Thanks

[R0g22]

SIP is much easier. You only need to allow 5060 for both TCP and UDP. The media negotiation is carried out through SDP which are carried along with the same SIP message. No separate messages and/or port usage. But you might need to consider opening up the RTP port range on the firewall to avoid one way/no audio issues.

[Remon Adel]

Hello,
After converting gateway to sip ,Outbound calls works fine but incoming failed ,
kindly find attached of below debugs during incoming call issue
debug voip ccapi inout
debug ccsip all
debug vpm signal

Incomming voice port is 0/1/0 and configured with plar 1204

[R0g22]

I never see the router sending a INVITE to CUCM. The call gets disconnected at the port. Take these logs in a buffer rather on the monitor terminal. Use only "debug ccsip message" and "debug ccsip error" along with VPM signal, voip vtsp all and ccapi inout.

[Remon Adel]

WE GET THIS ISSUE FIXED

Hi
the issue was , CUCM reject anonymous calls as per debug ,SIP Header From info is (<sip:9[2-9].......@X.X.X.X.:5060>Expires:  3600)  so modify these by below config on sip GW and disable regect  anonymous calls on Phones line .
we faced these issue only with incoming call through  fxo line which not provide caller number from Teleco .

voice class sip-profiles 1

 request INVITE sip-header From modify "<sip:9*@" "<sip:0000000000@"

!

dial-peer voice 1 voip

 voice-class sip profiles 1