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Skype Connect Inbound Dial Plan - RESOLVED
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KentC
Brekeke Guru


Joined: 09 Dec 2011
Posts: 108
Location: rw-rw-rw-

PostPosted: Sat Dec 13, 2014 3:45 pm    Post subject: Skype Connect Inbound Dial Plan - RESOLVED Reply with quote

1. Brekeke Product Name and Version:
Brekeke Sip Server Version 3
2. Java version:
1.6
3. OS type and the version:
CentOS 5.8
4. UA (phone), gateway or other hardware/software involved:
Skype Connect + Counterpath Bria 2.4 Professional
5. Your problem:

I would like to know how to build the best dial plan for using Skype Connect using "Registration" under Authentication Details. I can send calls and can bring in the Skype Number with the following rule, and want to send to a registered user like shown in the deploy patterns example below:

Matching Patterns
$request = ^INVITE
$addr = ^sip.skype.com$
$registered = true
To = sip:12027381550@
Deploy Patterns
$rtp = true
$nat = true
To = sip:99051000256359@MY.LAN.IP.ADDRESS


What would the better option to be so I can route this to my ext 99051000256359? Any suggestions, would be great as this would let me configure Skype fully working into Brekeke Sip Server without workaround.

Thanks,
Kent C.


Last edited by KentC on Mon Dec 15, 2014 2:31 pm; edited 1 time in total
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KentC
Brekeke Guru


Joined: 09 Dec 2011
Posts: 108
Location: rw-rw-rw-

PostPosted: Mon Dec 15, 2014 5:37 am    Post subject: Reply with quote

I can also try the following routing JUST the Skype Number and still not getting through when this is registered via LAN:

Here are the screenshots of what is happening:

Active Sessions:
http : //tinypic.com/r/rrnmgi/8

Session Details of Call:
http : //tinypic.com/r/10r0fiw/8

CURRENT DIAL-PLAN : [Another attempt]

Matching Patterns
$request = ^INVITE
$addr = ^sip.skype.com$
$registered = true
To = sip:12027381550@
Deploy Patterns
$rtp = true
$nat = true
To = sip:12027381550@MY.LAN.IP.ADDRESS


In this graph, the call sends ^INVITE from 63.209.x.x to LAN IP then LAN IP sends ^INVITE to the ROUTER IP, and the cycle repeats till CANCEL is sent from Skype IP.

Code:
|Time     | 63.209.144.201                        | WAN ROUTER IP                         |
|         |                   | LAN BREKEKE IP        |                   
|10.632   |         INVITE SDP (g711U g729 g711A telephone-eventRT...pe-101)          |                   |SIP From: <sip:12102740656@sip.skype.com:19000 To:<sip:12027381550@sip.skype.com
|         |(5060)   ------------------>  (5060)   |                   |
|10.632   |         100 Trying|                   |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|10.634   |                   |         INVITE SDP (g711U g729 g711A telephone-eventRT...pe-101)          |SIP Request
|         |                   |(5060)   ------------------>  (1029)   |
|11.136   |                   |         INVITE SDP (g711U g729 g711A telephone-eventRT...pe-101)          |SIP Request
|         |                   |(5060)   ------------------>  (1029)   |
|12.138   |                   |         INVITE SDP (g711U g729 g711A telephone-eventRT...pe-101)          |SIP Request
|         |                   |(5060)   ------------------>  (1029)   |
|14.138   |                   |         INVITE SDP (g711U g729 g711A telephone-eventRT...pe-101)          |SIP Request
|         |                   |(5060)   ------------------>  (1029)   |
|18.141   |                   |         INVITE SDP (g711U g729 g711A telephone-eventRT...pe-101)          |SIP Request
|         |                   |(5060)   ------------------>  (1029)   |
|26.142   |                   |         INVITE SDP (g711U g729 g711A telephone-eventRT...pe-101)          |SIP Request
|         |                   |(5060)   ------------------>  (1029)   |
|42.145   |                   |         INVITE SDP (g711U g729 g711A telephone-eventRT...pe-101)          |SIP Request
|         |                   |(5060)   ------------------>  (1029)   |
|47.773   |         CANCEL    |                   |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |                   |
|47.774   |         200 OK    |                   |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|47.774   |         487 Request Terminated          |                   |SIP Status
|         |(5060)   <------------------  (5060)   |                   |
|47.813   |         ACK       |                   |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |                   |


Let me know what you guys think...thanks!

Kent C.
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Laurie
Brekeke Master Guru


Joined: 07 Jan 2008
Posts: 242

PostPosted: Mon Dec 15, 2014 10:52 am    Post subject: Reply with quote

Hi Kent,

> $addr = ^sip.skype.com$

$addr's value should be an IP address. It doesn't indicate an FQDN.
http://wiki.brekeke.com/wiki/DialPlan-Matching-addr

If Skype's From-URI indicates the domain, you can define like this in the Matching Patterns.
From = sip.skype.com

The reason why your DialPlan rule doesn't match will be $addr.
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KentC
Brekeke Guru


Joined: 09 Dec 2011
Posts: 108
Location: rw-rw-rw-

PostPosted: Mon Dec 15, 2014 2:25 pm    Post subject: Reply with quote

Hi Laurie,

Thank you for the major tip. I knew someone here would clue me into what I was missing...

I tried what you suggested and noticed when the call was coming into the server, since this was using the Brekeke Sip Server directly for the registration and session details showed to-if as LAN IP and to-uri as router IP with sip.skype.com locked into place during routing...

I figured out maybe I needed to turn these two settings off together to allow the FQDN sip.skype.com adjust to IP [Your hint provided of course] and now the call is routing as expected!

HERE IS THE SETTINGS TURNED OFF IF YOU ARE USING UPPER REGISTRATION:

DNS SRV = off
DNS AAAA = off


Since the Skype ID is very unique to it's protocol, I think @best this would be the final answer.

This has been resolved as Skype DID Number purchased routed and called my Counterpath Bria 2.4 Pro Softphone as expected.

Thank you again for your assistance making this work under REGISTRATION settings on Authentication Details tab of Skype Connect Sip Profile.

Kent C.
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KentC
Brekeke Guru


Joined: 09 Dec 2011
Posts: 108
Location: rw-rw-rw-

PostPosted: Fri Dec 19, 2014 1:19 pm    Post subject: Reply with quote

All,

Here is the settings I used to make Skype Connect work both ways on the Brekeke Sip Server LIKE a PBX using UPPER REGISTRATION inbound/outbound working[ONLY USING sip.skype.com + brekeke registrars]

Skype Connect Register via Brekeke Sip Server UPPER REGISTRATION from softphone [WORKING/CONFIRMED IN SCREENSHOT]
http://tinypic.com/r/mtmnnp/8

DIAL-PLAN
http://tinypic.com/r/fp1ndi/8


THE TRICK TO MAKING THIS WORK!

MUST HAVE THIS SETUP UNDER MANUAL REGISTER:
Code:
99051000256359   sip : 99051000123456 : skype connect pw here @ sip.skype.com   

 Expires : 360000000
 Priority : 1000
 User Agent : Brekeke Admintool rev.333
 Requester : 127.0.0.1:43232
 Time Update : Sun Dec 14 12:47:26 CST 2014

12027381550   sip : 12027381550 @ 1XX.0.0.X:53468    Expires : 360000000
 Priority : 1000
 User Agent : Bria Professional release 2.4 stamp 49381
 Requester : 1XX.0.0.X:53468
 Time Update : Wed Dec 17 10:22:40 CST 2014
   

99051000123456   sip : 99051000123456 @ 1XX.0.0.X:32740    Expires : 45
 Priority : 1000
 User Agent : Bria Professional release 2.4 stamp 49381
 Requester : 1XX.0.0.X:32740
 Time Update : Fri Dec 19 14:26:02 CST 2014


-You will need to first use REGISTER [TICK ON] ONLY, build the dial plan [either listed for LAN will work or WAN if you prefer]
Register your skype number to Brekeke Sip Server via softphone/hardphone. Then after you register phone, then TURN ON UPPER REGISTRATION [TICK ON] and use sip.skype.com with sip uri for your Skype account is manually registered via REGISTERED CLIENTS tab. Then RE-REGISTER your SIP USER with password from Skype Connect account on softphone/hardphone which will register on Skype Connect under your Authentication Details. Once you do this, you should then be able to call inbound to your Skype number and see the call in Active Sessions! Has to take the dial-plan posted to work cleanly like real phone.

SKYPE INBOUND

Matching Patterns
$request = ^INVITE
$registered = true
To = sip:12027381550@
Deploy Patterns
$rtp = true
$nat = true
$b2bua = true
To = sip:12027381550@LAN.IP.ADDR.HERE

That makes the call delivery a little faster then without b2bua.

I'm not sure if there is a clear way to send IP Authentication to sip.skype.com outbound.

This is without the workaround I've posted in prior thread for Skype Connect/Skype Manager.

Thanks for your time viewing this,
Kent C.[/code]
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