01-20-2011 04:07 AM - edited 03-16-2019 02:58 AM
Hello all,
I've got a CME system on a 2811 router, connected to a SIP handoff from the carrier. Outbound calls work fine; inbound calls work SORT-OF, as in I can hear "something" ring and then the voicemail system picks up the call. However, the message that plays is "Enter your ID, followed by #", as if the caller had dialed a DiD, then pressed * to check VM messages.
I have looked everywhere and cannot find what might be causing this to happen. Another set of eyes would be greatly appreciated to help get the calls to go to the handset first, then to the regular VM message that indicates that the person just wasn't able to answer the call.
Relevant snippets of the config:
voice rtp send-recv
!
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
header-passing
error-passthru
early-offer forced
midcall-signaling passthru
<part of config config omitted>
voice translation-rule 1
rule 1 /\(^8\)\([2-9]......\)/ /\2/
rule 2 /\(^8\)\(1[2-9]..[2-9]......\)/ /\2/
rule 3 /^8911/ /911/
rule 4 /^911/ /911/
!
!
voice translation-profile OUTBOUND_e164
translate called 1
<part of config config omitted>
dial-peer voice 5000 voip (dial peer for voicemail)
destination-pattern 50..
b2bua
session protocol sipv2
session target ipv4:172.16.31.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 911 voip
translation-profile outgoing OUTBOUND_e164
destination-pattern 911
session protocol sipv2
session target sip-server
voice-class sip early-offer forced
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 8911 voip
translation-profile outgoing OUTBOUND_e164
destination-pattern 8911
session protocol sipv2
session target sip-server
voice-class sip early-offer forced
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 8001 voip
translation-profile outgoing OUTBOUND_e164
destination-pattern 8[2-9]......
session protocol sipv2
session target sip-server
voice-class sip early-offer forced
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 8002 voip
translation-profile outgoing OUTBOUND_e164
destination-pattern 81[2-9].........
session protocol sipv2
session target sip-server
voice-class sip early-offer forced
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 8011 voip
translation-profile outgoing OUTBOUND_e164
destination-pattern 8011T
session protocol sipv2
session target sip-server
voice-class sip early-offer forced
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 100 voip
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte
codec g711ulaw
<part of config config omitted>
sip-ua
authentication username 1-DX-1224 password <obfuscated>
sip-server ipv4:<obfuscated>:5060
presence enable
!
!
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 1
sdspfarm transcode sessions 2
sdspfarm tag 1 confprof1
conference hardware
video
maximum bit-rate 384
authentication credential admin <obfuscated>
max-ephones 42
max-dn 144
ip source-address 172.16.31.1 port 2000
max-redirect 20
system message <obfuscated>
url services http://172.16.31.2/voiceview/common/login.do
url authentication http://172.16.31.1/CCMCIP/authenticate.asp
cnf-file location flash:
load 7937 apps37sccp.1-4-2-0.bin
load 7960-7940 P00308010200
load 7942 SCCP42.9-1-1SR1S
time-zone 12
voicemail 5000
max-conferences 2 gain -6
moh music-on-hold.au
multicast moh 239.10.10.10 port 2000 route 172.16.31.1
web admin system name admin secret <obfuscated>
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 8
I'm not sure if I need to be setting up an inbound translation, and if so, what does that look like, or am I missing something simple that I can't see the forest for the trees on. As I said, another set of eyes would be greatly appreciated here.
Thanks in advance!
Kevin
01-24-2011 10:44 AM
From problem description it seems the incoming call has DNIS 50XX
and is using dialpeer 5000 for outbound leg to CUE (:172.16.31.2 ??)
where its being handed over to VM app instead of AA.(VM trigger matching
with DNIS) That's all speculation based on info provided.
For us to see what is going on, we'd need some debugs
Capture following debugs for one such call in a logging buffer and then dump it
in a text file:
deb ccsip mess
deb ccsip err
deb voip ccapi inout
Pl. note calling and called #
Include sh runn from CUE as well.
Thx.
DK
01-31-2011 04:50 AM
This has been solved.
I had to add "no supplementary-service sip moved-temporarily" under voice service voip and that cleared things up.
Thanks for the help though!
Kevin
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