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CME Register to Skype Fail

rock981119
Level 1
Level 1

Hi All,

  I Have Create Skype sip accout, and follow some doc to configure CME to regsiter, but fail, ask for help.

Router config:

sip-ua

credentials username 99051000154290 password 7 033E6D063E210F67543A3427161303 realm sip.skype.com

authentication username 99051000154290 password 7 003E250B3C75252D1512616C08180D realm sip.skype.com

no remote-party-id

retry invite 2

retry register 10

timers connect 100

no timers hold

registrar dns:sip.skype.com expires 3600

sip-server dns:sip.skype.com

connection-reuse

host-registrar

handle-replaces

==============================================================

E2921#sh sip-ua register status

Line                             peer       expires(sec) registered P-Associ-URI

================================ ========== ============ ========== ============

99051000154290                   -1         51           no

==============================================================

debug ccsip massage was in attach, thx for guidance.

Rock.

2 Accepted Solutions

Accepted Solutions

Chris Deren
Hall of Fame
Hall of Fame

As seen in your debug the connection does not get established and is not authorized:

401 Unauthorized

Are you sure you have corrent username and password in your sip-ua config?

Chris

View solution in original post

It doesn't change the sip header directly, it just change your calling id number, for example from my extension 1306 to

99051000001534, that way when your call match your skype dial-peer, Skype will accept your call because your are using the number (example: 99051000001534) as your CLID, that is what Skype is expecting to receive.

Gabriel.

View solution in original post

9 Replies 9

Chris Deren
Hall of Fame
Hall of Fame

As seen in your debug the connection does not get established and is not authorized:

401 Unauthorized

Are you sure you have corrent username and password in your sip-ua config?

Chris

Hi Chris,

  Thx to reply, maybe credentials is not correct, I debug ccsip info saw that:

//36/000000000000/SIP/Info/sipSPIHandleAuthChallenge: Invalid Username or Password

In your opinion that config is OK without User/Pass?

I never tried connecting to Skype, but I doubt they would allow connecting without authenticating.

Chris

Hi

I read (well... scanned over half-heartedly) a document from Microsft the other day that described how to set up a connection from MS Lync to Skype.

The solution basically involved putting in an Asterisk server, which ran a custom developed interface of some sort to Skype. So call went from Lync to Asterisk, to custom code to Skype. Suggests to me that connecting to skype is a little more complex than a simple SIP trunk.

Have a read: http://www.microsoft.com/download/en/details.aspx?id=22644

Aaron Harrison

Principal Engineer at Logicalis UK

Please rate helpful posts...

Aaron Please remember to rate helpful posts to identify useful responses, and mark 'Answered' if appropriate!

Hi,

This can work with a normal SIP trunk in your voice gateway, I configured this for two of our customers. So, first of all, can you ping sip.skype.com?..Also, can you check in your skype manager that you are using the username/password authentication method (also check if your password and SIP user is the same configure in your router).

Gabriel.

Hi Chris,

  User/Pass incorrect,  change another accout it can registered.

99051000154290@sip,skype.com, that URI i think is regular to make a call.

What if behind CBUE, has a CallManager of IP Phone something, how  can i modification the sip header make behind CUBE device can use the  regular format outside call?

Hi, you can use translation rules to change your calling number ID. I use to change this way:

voice translation-rule 1111

rule 15 /^....$/ /99051000001534/

voice translation-profile Skype

translate calling 1111

dial-peer voice 101 voip

description Skype Outgoing

translation-profile outgoing Skype

destination-pattern 9T

voice-class codec 100

session protocol sipv2

session target dns:sip.skype.com

dtmf-relay rtp-nte

no vad

So everything that comes with a 4 length digit can be converted to the SIP user number

Gabriel.

Hi Gabriel,

  Did you mean translation-rule can change the sip request header like:

"From: <>99051000154xxx@sip.skype.com>"?

  Cuz i saw some doc using "voice class sip-profile xxx", some like that pattern:

voice class sip-profile xxx

  request INVITE sip-header From modify "sip:(.*>)"  "sip:99051000154xxx@sip.skype.com>"

And use in "voice service voip" or "dial-peer".

If translation-rule can do that it will be nice, cuz i am not skillful for "voice class sip-profile"

It doesn't change the sip header directly, it just change your calling id number, for example from my extension 1306 to

99051000001534, that way when your call match your skype dial-peer, Skype will accept your call because your are using the number (example: 99051000001534) as your CLID, that is what Skype is expecting to receive.

Gabriel.

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