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Replies

Cisco CME config with Skype Connect error

DOUGLAS DRURY
Level 1
Level 1

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#

23 Replies 23

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.

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

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.

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

Please rate all helpful posts "The more you help the more you learn"

Thanks Carlo, I'll try and update the IOS as soon as i can.

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

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. 

//Suresh Please rate all the useful posts.

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?

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