12-09-2013 04:58 AM - edited 03-16-2019 08:46 PM
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.
Solved! Go to Solution.
12-12-2013 12:58 PM
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
12-12-2013 12:58 PM
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
12-13-2013 12:16 AM
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
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide