01-18-2014 09:16 AM - edited 03-16-2019 09:19 PM
Hi,
I've configured my CME with a sip connection from Skype but i can't seem to figure out why i can't make outgoing or incomming calles. Would sombody be able to tell me where i have gone wrong please.
Thanks
in advance. Usernames and passwords in the config have been changed.
Internal calls work
no ip domain lookup
ip domain name TEK-SERV.local
multilink bundle-name authenticated
!
!
!
!
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
no update-callerid
!
!
!
!
!
voice class sip-profiles 1000
request ANY sdp-header Connection-Info remove
response ANY sdp-header Connection-Info remove
!
!
!
!
!
!
!
!
!
!
voice translation-rule 1111
rule 15 /^...$/ /88051000228349/
!
voice translation-rule 2222
rule 2 /^80/ //
rule 3 /^80/ /0044/
!
!
voice translation-profile skype
translate calling 1111
translate called 2222
!
!
voice-card 0
no dspfarm
!
!
!
!
username phoneeng privilege 15 secret 5 $1$vWm6$7HiP3sqDZ7a5D94tmw9xd.
archive
log config
hidekeys
!
!
no ip ftp passive
!
!
!
dial-peer voice 101 voip
description skype outgoing
translation-profile outgoing skype
destination-pattern 8.T
voice-class sip profiles 1000
session protocol sipv2
session target dns:sip.skype.com
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
!
sip-ua
credentials username 88051000228349 password 0 password realm sip.skype.com
authentication username 88051000228349 password 0 password
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sip.skype.com expires 3600
sip-server dns:sip.skype.com
connection-reuse
host-registrar
g729-annexb override
!
!
telephony-service
no auto-reg-ephone
load 7914 S00104000100
load 7902 CP7902060000SCCP050124A.sbin
load 7905 CP7905060000SCCP050124A.sbin
load 7906 SCCP11.8-3-3S.loads
load 7911 SCCP11.8-3-3S.loads
load 7912 CP7912060000SCCP050124A.sbin
load 7920 cmterm_7920.4.0-02-00.bin
load 7960-7940 P00307020200
max-ephones 42
max-dn 42
ip source-address 192.168.16.1 port 2000
system message TEK SERV
url messages http://192.168.16.1:80/messages
cnf-file location flash:
date-format dd-mm-yy
voicemail 3099
max-conferences 8 gain -6
hunt-group logout HLog
moh flash:/music-on-hold.au
web admin system name ADMIN secret 5 $1$vo27$4gASd5vT0G4S68DVHmzBE1
dn-webedit
time-webedit
transfer-system full-consult
directory entry 1 901771622598 name Test
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-template 1
softkeys hold Join Newcall Resume Select
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd Login
softkeys seized Redial Endcall Cfwdall Pickup Gpickup Callback
softkeys alerting Acct Callback Endcall
softkeys connected Hold Endcall Trnsfer Confrn Acct Park
!
!
ephone-dn 1 dual-line
number 3001
pickup-group 3097
label 3001
description IS
name I S
!
!
ephone-dn 2 dual-line
number 3002
pickup-group 3097
label DDI-3002
name Jon Dow
!
!
ephone-dn 3 dual-line
number 3003
pickup-group 3097
label 3003
description Douglas Drury
name Douglas Drury
!
!
ephone-dn 4 dual-line
number 3004
pickup-group 3097
!
!
ephone 1
mac-address 0012.0101.0C75
username "IS"
type 7940
button 1:1
!
!
!
ephone 2
mac-address 001E.ECE9.E966
username "jdow"
type CIPC
button 1:2
!
!
!
ephone 3
mac-address 0012.00EC.B345
username "ddrury" password 3003
fastdial 1 3099 name c
type 7940
button 1:3
pin 3003
!
!
ephone-hunt 1 sequential
pilot 3098
list 3001, 3002, 3003, 3004
no-reg pilot
description Service Desk
!
!
!
line con 0
logging synchronous
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
login local
transport input telnet
!
scheduler allocate 20000 1000
end
TEK-SERV-CME#
01-27-2014 12:56 PM
Douglas, can you try to test your translation rule? To see if your translation works fine for the caller id that Skype is expecting to receive.
TEK-SERV-CME#test voice translation-rule 1111 3003
------
You need to modify your translation rule to:
voice translation-rule 1111
rule 15 /^....$/ /99051000228349/
instead of
voice translation-rule 1111
rule 15 /^...$/ /99051000228349/
HTH
Gabriel.
01-27-2014 01:16 PM
Hi Gabriel,
I ajusted the translation rule (see below) I tried to make an outgoing call but it failed. Is there a debug command i can run?
TEK-SERV-CME#test voice translation-rule 1111 3003
Matched with rule 15
Original number: 3003 Translated number: 99051000228349
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
01-27-2014 02:27 PM
The debug command is the one you have been used "debug ccsip messages" . Another question, Skype expects to receive as called number the country code + rest of the number. Are you dialing that way?
For example to call a USA number, Skype needs to receive 1+7147773456
Gabriel.
01-27-2014 12:43 PM
Hi Douglas.
Before we go on troubleshooting, I suggest to upgrade your IOS to the latest available version that should be 15.1.4M
HTH
Regards
Carlo
Sent from Cisco Technical Support iPhone App
01-27-2014 01:17 PM
Thanks Carlo, I'll try and update the IOS as soon as i can.
11-25-2014 05:05 PM
douglas did u get this running? im currently testing a set up.. my calls are going out. my incoming calls are reaching my router but but the cucm.. so i cant recieve to my headset.. would appreatiate some help
11-25-2014 09:50 PM
Hi Macazarpros,
You may need to check the inbound CSS on the sip trunk configuration. If that doesn't help, I suggest you to open a new thread with debug ccsip message collected for a inbound call.
11-28-2014 04:57 PM
thanks it as indeed the css.. .. calling works fine from the dial peer directly to a phone.. when the directed to the attendent and a pilot number hands off the call to a phone in a hunt group the party on the other end ccant hear me.. any thoughts?
11-26-2014 02:30 AM
Hi Douglas,
Try this:
voice service voip
sip
bind all source-interface <IP address of interface>
Enter the IP address of your voice interface or loopback interface.
If it still doesn't work then please share the output of debug ccsip messages.
Thanks
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