%IP-3-DESTHOST : how to fix it?

iantra123 · ‎07-20-2010

Hi

I experience a problem about transferring calls from an CUE 7.1 auto attendant console to CCM4.

Calls are comming in a 1st gateway (GW-1),

and it is forwarded to the Autoattendant (aa.aef) that is in the CUE in another CUCME router.(GW2)

The operator is in the CUCM 4.2.

Scenario :

0. caller A make call trhoug an E1 PRI.

1. call arrive int GW-1 (that has PRI)

2. Call is Transfered to CUE (GW-2) auto attendant

3. from CUE calle A push 0 to call the operator

4. Operator is in a CUCM, and phone ring

5. Operator want to trransfert to a destination B (eg using transfer full consult)

5.1 When transfer : the end destination (B) hears the operator

5.2 after pushing the buttun "transfer", the call seems be transfered.

5.3 Caller A cannot hear caller B. and call is disconnected.

Is there any solution?

CUCME router (GW2) give the error :

=========================

Jul 20 08:39:17.443: %IP-3-DESTHOST: src=10.11.4.2, dst=0.0.0.0, NULL desthost -Process= "VOIP_RTCP", ipl= 0, pid= 308, -Traceback= 0x418B162C 0x41FD331C 0x41FD4334 0x41FD491C
Jul 20 08:39:21.711: %IP-3-DESTHOST: src=10.11.4.2, dst=0.0.0.0, NULL desthost -Process= "VOIP_RTCP", ipl= 0, pid= 308, -Traceback= 0x418B162C 0x41FD331C 0x41FD4334 0x41FD491C
Jul 20 08:39:25.763: %IP-3-DESTHOST: src=10.11.4.2, dst=0.0.0.0, NULL desthost -Process= "VOIP_RTCP", ipl= 0, pid= 308, -Traceback= 0x418B162C 0x41FD331C 0x41FD4334 0x41FD491C
Jul 20 08:39:30.575: %IP-3-DESTHOST: src=10.11.4.2, dst=0.0.0.0, NULL desthost -Process= "VOIP_RTCP", ipl= 0, pid= 308, -Traceback= 0x418B162C 0x41FD331C 0x41FD4334 0x41FD491C
Jul 20 08:39:33.295: %IP-3-DESTHOST: src=10.11.4.2, dst=0.0.0.0, NULL desthost -Process= "VOIP_RTCP", ipl= 0, pid= 308, -Traceback= 0x418B162C 0x41FD331C 0x41FD4334 0x41FD491C
Jul 20 08:39:39.855: %IP-3-DESTHOST: src=10.11.4.2, dst=0.0.0.0, NULL desthost -Process= "VOIP_RTCP", ipl= 0, pid= 308, -Traceback= 0x418B162C 0x41FD331C 0x41FD4334 0x41FD491C
========================

The result of error decoder is in this link :

http://www.cisco.com/cgi-bin/Support/Errordecoder/index.cgi?action=search&locale=en&index=all&query=%25IP-3-DESTHOST&counter=0&paging=5&links=reference&sa=Submit

==============
! GW2 IOS is: Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version 12.4(22)YB, RELEASE SOFTWARE (fc2)

===============
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
!

dial-peer voice 2672 voip

description to AAttendant
destination-pattern 2672
session protocol sipv2
session target ipv4:10.11.4.3
dtmf-relay rtp-nte
codec g711ulaw
!

There is a transcoder, MTP, CFB

===================

regards,

Antra

jsliwinski · ‎07-20-2010

This can be IP connectivity issue between your gateway and IP phone B. As a result RTP stream from phone B to Gateway does not make it.

"5.3 Caller A cannot hear caller B."

1. What is the IP address of GigabitEthernet0/0

Try to do extended ping from the gateway using GigabitEthernet0/0 address and the IP address of the phone B. If you do not get reply this is your answer you need to fix IP connectivity.

-- j

iantra123 · ‎07-21-2010

Hi,

I call directly the CUCME to phone B.

Everything is ok.

I called directly from a phone A(PRI) to phone B : ok

But if the call goes from A ->CUA-AA -> Operator -> B :call is cuted when A is transfered to B.