01-22-2012 04:17 AM - edited 03-16-2019 09:08 AM
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.
Solved! Go to Solution.
01-22-2012 07:52 AM
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
01-23-2012 10:02 AM
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.
01-22-2012 07:52 AM
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
01-22-2012 08:46 AM
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?
01-22-2012 11:22 AM
I never tried connecting to Skype, but I doubt they would allow connecting without authenticating.
Chris
01-22-2012 02:02 PM
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
Principal Engineer at Logicalis UK
Please rate helpful posts...
01-22-2012 07:16 PM
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.
01-23-2012 03:08 AM
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?
01-23-2012 06:44 AM
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.
01-23-2012 09:48 AM
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"
01-23-2012 10:02 AM
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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