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No Audio with Call Transfer in CUE Script

Answered Question
Dec 9th, 2013
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Hello,


I have a CUE script set up to dial several extensions and transfer them over to a MeetMe conference hosted on the same 2911 router where the CUE ISM is installed. The call flow is as follows:


CUCM <---> 2911 <---> CUE


with the calls to the script being initiated by phones registered with the CUCM.


Upon dialing the script pilot number, the calls to the specified extensions are initiated and transferred to the MeetMe DN (7070) successfully, but there is no audio on the calls. The script seems to complete successfully as well, because all the extensions together with the initiating one are transferred to the MeetMe DN. The "show call active voice compact" command shows all the participants connected to 7070 and the "show ephone-dn conf" output displays the correct number of active sessions to 7070


Direct calls to the MeetMe DN (without going through the script) work fine.


Logging in the atrace.log seems to show that unmuting the calls by the CUE fails. I have tested with all three SIP call transfer methods in CUE to no avail.


CUE and 2911 configs as well as the atrace.log file from CUE are attached herewith.


Any ideas will be highly appreciated.

Correct Answer by Alexander Maroukian about 3 years 8 months ago

Hi Miroslav,


You may check the transfer method in CUE - please set to bye-also and without h450 services under voice service voip in CME.


HTH,

Alex

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Correct Answer
Alexander Maroukian Thu, 12/12/2013 - 12:58
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  • Cisco Employee,

Hi Miroslav,


You may check the transfer method in CUE - please set to bye-also and without h450 services under voice service voip in CME.


HTH,

Alex

MiroslavSG Fri, 12/13/2013 - 00:16
User Badges:

Hello Alexander,


Thank you for the reply and the suggestion. Setting the CUE transfer to "blind bye-also" and disabling h450. services on the router was indeed the correct combination of settings to get the audio running with this script.


Below are the settings in the "voice service voip" for anyone who might run into similar issues:


voice service voip

no ip address trusted authenticate

ip address trusted call-block cause call-reject

no lpcor incoming

voice call rate monitor

no notify redirect ip2ip

no callmonitor

no rtcp keepalive

rtp-port range 0 0

no rtp-ssrc multiplex

no dtmf-interworking

cpa

cpa timing silent 375

cpa timing live-person 2500

cpa timing timeout 3000

cpa timing noise-period 100

cpa timing valid-speech 112

cpa timing term-tone 15000

cpa threshold noise-level max -50dBm0

cpa threshold noise-level min -60dBm0

cpa threshold active-signal 15db

no mode border-element

media flow-through

cti timeout make-call-prompt 60

cti shutdown

allow-connections sip to sip

no redundancy

no supplementary-service h450.2

no supplementary-service h450.3

no supplementary-service h450.7

no supplementary-service h450.12 advertise-only

no supplementary-service h450.12

no supplementary-service h225-notify cid-update

supplementary-service sip moved-temporarily

no supplementary-service sip refer

supplementary-service sip handle-replaces

no supplementary-service media-renegotiate

no supplementary-service ringback h225-info

redirect ip2ip

text relay modulation baudot45.45 autobaud-on

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

fax-relay sg3-to-g3

  no cause-code legacy

modem relay latency 200

modem relay gateway-xid dictionary 1024 string-length 32 compress backward

modem relay sse redundancy interval 20

modem relay sse redundancy packet 3

modem relay sse t1 1000

modem relay sse retries 3

modem relay sprt retries 12

modem relay sprt v14 receive playback hold-time 50

modem relay sprt v14 transmit hold-time 20

modem relay sprt v14 transmit maximum hold-count 16

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