Transfer to DN that is CFA to voicemail fails

[Daniel Castimore]

I am running CME 9.1 on a 3925 router. I have CUE 8.6.6 installed on the ISM. My problem exists with several different phone models as well as internal and external calls.

1. User on extension 8236 sets call forward all to voicemail.
   
2. Someone calls another extension (8244).
   
3. The original call is transfered to 8236.
   
4. During the attended transfer portion, the user on 8244 hears the voicemail message begin to play, and presses transfer a second time to complete the transfer.
   
5. The original caller hears MOH while the transfer is in progress, but when the transfer is complete, all they hear is silence.
   

See configs below:

telephony-service

sdspfarm units 2

sdspfarm tag 1 conference

conference hardware

no auto-reg-ephone

authentication credential REDACTED

!

max-ephones 265

max-dn 600

ip source-address 10.0.5.2 port 2000

system message REDACTED

url services http://10.10.20.1/voiceview/common/login.do

url authentication http://10.10.20.1/voiceview/authentication/authenticate.do

cnf-file location flash:

cnf-file perphone

load 7937 apps37sccp.1-4-4-0

time-zone 4

voicemail 7999

max-conferences 12 gain -6

call-park system application

moh "flash:music-on-hold/music-on-hold.au"

multicast moh 239.10.16.4 port 2000

web admin system name REDACTED

dn-webedit

time-webedit

transfer-system full-consult

transfer-pattern 6.......

transfer-pattern ....

secondary-dialtone 6

create cnf-files version-stamp REDACTED

voice register global

mode cme

source-address 10.0.5.2 port 5060

max-dn 300

max-pool 265

load 9971 sip9971.9-3-2SR1-1

load 8961 sip8961.9-2-3-27.loads

load 8945 SIP8941_8945.9-3-1-18

load ATA-187 ATA187.9-2-3-1

load 6945 SIP8941_8945.9-3-1-18

authenticate realm all

timezone 4

voicemail 7999

tftp-path flash:

create profile sync

dial-peer voice 7999 voip

destination-pattern 7999

session protocol sipv2

session target ipv4:10.10.20.1

dtmf-relay sip-notify

codec g711ulaw

no vad

voice register dn 87

number 8236

call-forward b2bua all 7999

call-forward b2bua busy 7999

call-forward b2bua noan 7999 timeout 20

allow watch

pickup-group 3

mwi

voice register dn 92

number 8244

call-forward b2bua busy 7999

call-forward b2bua noan 7999 timeout 20

allow watch

pickup-group 3

mwi

[bharatk]

Hi Daniel,

I would want you to apply the following under the VOIP dial-peer pointing towards the CUE.

Configure voice class sip-profiles as below and apply
this "voice-class sip profile 1" to the voip dial-peer towards CUE.

voice class sip-profiles 1
request REINVITE sdp-header Audio-Attribute modify "sendonly" "sendrecv"

dial-peer voice 5299 voip<<<<<<<(dial-peer of the CUE)

 voice-class sip profiles 1

There is a known bug and below are its details:

http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCth96879

No media  after Consult Call transfer from CME  SIP phone to CUE-AA

Symptom:- One way audio, only when SIP CME IP Phone transfers a call to CUEConditions:- Transferring party is SIP phoneWorkaround:Configure voice class sip-profiles as below and apply this "voice-class sip profile 1" to the voip dial-peer towards CUE. voice class sip-profiles 1 request REINVITE sdp-header Audio-Attribute modify "sendonly" "sendrecv" -RG, Bharat Kumar

[jcp408ADP]

Rood issue is SIP phones on CME.  If you don't create and apply the sip profile below then the transfer of the call to CUE will not have the RTP stream re-addressed to the original call.    The command bolded below is specifically what was missing from my configuration.

voice class sip-profiles 1

  request REINVITE sdp-header Audio-Attribute modify "sendonly" "sendrecv"

voice service voip

  sip

    sip-profiles 1