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CUE greetings error for Multi site

Unanswered Question
Aug 26th, 2013
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Hi,



I have a multi site(Site A & site B) IP Telephony set up with Call manger Express and voice mail for CUE.


I have forwarded the calls to voice mail when no answer and busy.


Its working fine when an external call comes and if the user is busy or no answer its going to voice mail and fine with internal calls.


But when site A user tries to make calls to Site B  if the user is busy or no answer its going to voice mail but we cannot hear the greetings after the ring its blank and as we can see that its going to the Voicemail number.


Do I need to enable anything on unity express for this?


Thanks in advance


Nithin Louis.

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ronpatel Mon, 08/26/2013 - 03:39
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Is it happening in other way call . I mean if you call from site b to site a phone , what is the status for other way call

Sent from Cisco Technical Support iPad App

nithin louis Mon, 09/02/2013 - 06:38
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Hi Ronak,


Yes both side we are facing the same issue.


Site A

=====


dial-peer voice 7777 voip

description ***SIP dial peer- Unity express***

destination-pattern 777.

session protocol sipv2

session target ipv4:192.168.0.3

dtmf-relay sip-notify

voice-class codec 183

no vad


voice class codec 183

codec preference 1 g711alaw

codec preference 2 g711ulaw

codec preference 3 g729r8


dial-peer voice 767676 voip

description ***Dial-Peer to Site B***

destination-pattern 4[1,5]..

translate-outgoing called 321

session target ipv4:192.168.32.100

voice-class codec 183 

dtmf-relay h245-alphanumeric

no vad



Site B

======

dial-peer voice 6666 voip

description "Unity Express Dial Peer"

destination-pattern 666.

session protocol sipv2

session target ipv4:192.168.32.101

dtmf-relay sip-notify

voice-class codec 183

no vad


voice class codec 183

codec preference 1 g711alaw

codec preference 2 g711ulaw

codec preference 3 g729r8


dial-peer voice 1 voip

destination-pattern 913..

translate-outgoing called 123

session target ipv4:192.168.0.2

voice-class codec 183 

dtmf-relay h245-alphanumeric

no vad


Thanks

Nithin Louis

nithin louis Wed, 09/04/2013 - 06:29
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Hi ,


When I am calling from Site A(Extn.100) to Site B (Extn.500) and Site B's phone is offline then Site A 's phone can hear the greetings from CUE.But When Site B 's phone is on line then the same issue is happening like cannot hear the greetings.


We are connected the 2 site using Site to site VPN with sufficient bandwidth.


Thanks in advance


Nithin Louis.

brmeade Wed, 09/04/2013 - 09:55
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Collect the following for one of these calls:


debug h225 asn1

debug h245 asn1

debug ccsip messages


Disable console logging and use a logging buffer:

no logging console

logging buffered 5000000


Then to view the log, do the following:

term len 0

show log

brmeade Sat, 09/07/2013 - 14:51
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Immediately after the initial setup is complete with CUE, the gateway is sending a Re-Invite setting the call to receive only:


Sep  5 19:37:45.582: //6288/78FCE004BFEF/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:[email protected]:5060;transport=udp SIP/2.0



Via: SIP/2.0/UDP 192.168.32.100:5060;branch=z9hG4bK5EC65C



From: "Javed Jabber" ;tag=B62AB014-2202



To: ;tag=dsc6941c0f



Date: Thu, 05 Sep 2013 19:37:45 GMT



Call-ID: [email protected]



Supported: 100rel,timer,resource-priority,replaces,sdp-anat



Min-SE:  1800



Cisco-Guid: 2029838340-0362353123-3220164084-0968885464



User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2



Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER



CSeq: 102 INVITE



Max-Forwards: 70



Timestamp: 1378409865



Contact:



Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=2000"



Diversion: ;privacy=off;reason=no-answer;counter=1;screen=no



Expires: 180



Allow-Events: telephone-event



Content-Type: application/sdp



Content-Length: 208







v=0



o=CiscoSystemsSIP-GW-UserAgent 656 9782 IN IP4 192.168.32.100



s=SIP Call



c=IN IP4 192.168.32.100



t=0 0



m=audio 18282 RTP/AVP 0



c=IN IP4 192.168.32.100



a=recvonly



a=rtpmap:0 PCMU/8000



a=ptime:20



Seems like CUE just interprets that as inactive media:

Sep  5 19:37:45.590: //6288/78FCE004BFEF/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 Ok



Via: SIP/2.0/UDP 192.168.32.100:5060;branch=z9hG4bK5EC65C



To: ;tag=dsc6941c0f



From: "Javed Jabber" ;tag=B62AB014-2202



Call-ID: [email protected]



CSeq: 102 INVITE



Content-Length: 128



Content-Type: application/sdp



Contact:



Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=2000"



Allow-Events: telephone-event



Allow: INVITE, BYE, CANCEL, ACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO



Cisco-Gcid: [email protected]







v=0



o=CUE 3641731 3 IN IP4 192.168.32.101



s=SIP Call



c=IN IP4 0.0.0.0



t=0 0



m=audio 20856 RTP/AVP 0



a=rtpmap:0 PCMU/8000


This results in silence for 14 seconds until the user hangs up:

Sep  5 19:37:59.178: H225.0 INCOMING PDU ::=



value H323_UserInformation ::=

    {

      h323-uu-pdu

      {

        h323-message-body releaseComplete :

        {

          protocolIdentifier { 0 0 8 2250 0 4 }

          callIdentifier

          {

            guid '9D737A05157711E39EE0A3E57C8B164F'H

          }

        }

        h245Tunneling TRUE

      }

    }


Sep  5 19:37:59.182: //6288/78FCE004BFEF/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:[email protected]:5060;transport=udp SIP/2.0



Via: SIP/2.0/UDP 192.168.32.100:5060;branch=z9hG4bK5EE1346



From: "Javed Jabber" ;tag=B62AB014-2202



To: ;tag=dsc6941c0f



Date: Thu, 05 Sep 2013 19:37:45 GMT



Call-ID: [email protected]



User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2



Max-Forwards: 70



Timestamp: 1378409879



CSeq: 103 BYE



Reason: Q.850;cause=16



P-RTP-Stat: PS=0,OS=0,PR=2,OR=161,PL=0,JI=0,LA=0,DU=13



Content-Length: 0



So we need to find out what is causing that recvonly event.


At the beginning of the call, we see the inbound and outbound OpenLogicalChannel but I don't see any OpenLogicalChannelAcks.  I wonder if that may be the problem.


Can you post your full configs?

brmeade Sat, 09/07/2013 - 15:09
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It looks like your SiteB doesn't have an incoming voip dial-peer that these calls would match so they may be matching dial-peer 0 which is G.729 only.  Try adding this on SB to your first voip dial-peer:


dial-peer voice 1 voip

incoming called-number .

nithin louis Tue, 09/10/2013 - 05:51
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Hi Brian,


I have added the incoming called-number . under the dial-peer voice 1 voip. But result is same


Thanks & Regards

Nithin Louis.

brmeade Tue, 09/10/2013 - 05:53
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Can you re-attach a fresh copy of your config and the same debugs?

nithin louis Sat, 09/07/2013 - 15:00
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Hi Brian,


Please note the below points..


Site A

=======

CME Ip addres is : 192.168.0.2 & CUE is 192.168.0..3


Site B

=====

CME IP addres is : 192.168.32.100 & CUE is 192.168.32.101


Regards

Nithin Louis.

Michael Gerrard Wed, 09/04/2013 - 10:09
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It smells like routing to me, with no-way-voice. Ensure reachability to the IP address of the CUE module IP address from everywhere.


Also, I'd add in incoming called-number . into your VoIP dial-peers to ensure that you don't hit dial-peer 0 when dialling between sites. CUE is obviously a bit funny about codec choices so best to end up using the right one.


Mike.

nithin louis Tue, 09/10/2013 - 05:50
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Hi Mike,


I have added the incoming called-number . in the Voip Dial-peer but result is same


Regards

Nithin Louis.

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