08-26-2013 03:30 AM - edited 03-16-2019 07:02 PM
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.
08-26-2013 03:39 AM
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
09-02-2013 06:38 AM
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
09-04-2013 06:29 AM
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.
09-04-2013 08:57 AM
Hi,
Any update guys....
Regards
Nithin Louis.
09-04-2013 09:55 AM
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
09-07-2013 02:23 PM
09-07-2013 02:51 PM
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:6666@192.168.32.101:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.32.100:5060;branch=z9hG4bK5EC65C
From: "Javed Jabber" <17101334>;tag=B62AB014-220217101334>
To: <6666>;tag=dsc6941c0f6666>
Date: Thu, 05 Sep 2013 19:37:45 GMT
Call-ID: 78FCE004-159911E3-BFF4CDF4-39C004D8@192.168.32.100
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: <17101334>17101334>
Call-Info: <192.168.32.100:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"192.168.32.100:5060>
Diversion: <112>;privacy=off;reason=no-answer;counter=1;screen=no112>
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: <6666>;tag=dsc6941c0f6666>
From: "Javed Jabber" <17101334>;tag=B62AB014-220217101334>
Call-ID: 78FCE004-159911E3-BFF4CDF4-39C004D8@192.168.32.100
CSeq: 102 INVITE
Content-Length: 128
Content-Type: application/sdp
Contact: <6666>6666>
Call-Info: <192.168.32.101:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"192.168.32.101:5060>
Allow-Events: telephone-event
Allow: INVITE, BYE, CANCEL, ACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO
Cisco-Gcid: 78FCE004-159911E3-BFF4CDF4-39C004D8@192.168.32.100
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:6666@192.168.32.101:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.32.100:5060;branch=z9hG4bK5EE1346
From: "Javed Jabber" <17101334>;tag=B62AB014-220217101334>
To: <6666>;tag=dsc6941c0f6666>
Date: Thu, 05 Sep 2013 19:37:45 GMT
Call-ID: 78FCE004-159911E3-BFF4CDF4-39C004D8@192.168.32.100
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?
09-07-2013 02:58 PM
09-07-2013 03:09 PM
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 .
09-10-2013 05:51 AM
Hi Brian,
I have added the incoming called-number . under the dial-peer voice 1 voip. But result is same
Thanks & Regards
Nithin Louis.
09-10-2013 05:53 AM
Can you re-attach a fresh copy of your config and the same debugs?
09-10-2013 07:55 AM
09-07-2013 03:00 PM
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.
09-04-2013 10:09 AM
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.
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