sipsllc

I get connected when my skypebuddy called me on my ipphone but no voice from both side. Wondering why?

thanks.

what i get is incoming sip call on the uplink screen and nothing else after...

 

any idea ?

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Hi there,

I have follow your steps as below and got error in uplink program with error "register attempt for sip fail".

I am confusing whether we need to create a SIP user 5506 in sip.conf or as an extension in a@home?

To do the testing, what number from IP phone connect from asterisk to call skype user or from skype user to call internal extension in asterisk.

Thanks ...

Hi folks,

 

I know a few of you have been trying to get asterisk@home to work with uplink, here is a working example although there is a bug somewhere that causes a freeze when dialing skype out. I suspect this is an uplink bug in which case there is nothing we can do except hope for new release.

 

1. Setup an a@h trunk as follows:-

 

Outgoing settings

 

Trunk Name: skype

 

PEER Details:

 

allow=all

caninvite=yes

canreinvite=yes

context=extensions

host=dynamic

insecure=very

nat=yes

secret=5506

type=friend

username=5506

 

Incoming settings

 

USER Context: 5506

 

USER Details:

 

allow=all

callerid="Incoming Skype"

caninvite=yes

canreinvite=yes

context=ext-queues

host=dynamic

insecure=very

nat=yes

secret=5506

type=friend

username=5506

 

Registration

 

Register String:

 

5506:5506@192.168.1.2/5506

 

The format of the register string is username:password@uplinkhost/username, ie my machine with the uplink and skype client running is 192.168.1.2, adjust to suit your own network.

 

Note that 5506 is an arbitrary number that is based on the asterisk setup in this forum. Also I am using a context=ext-queues on incoming as I am routing my skype calls into call queue 123. If you just want to ring an extension use context=from-internal etc.

 

2. Define an outbound route.

 

Route name: skype

 

Dial patters: 8|X. This routes all calls prefixed with an 8 via this route. eg to dial 00441234567890

 

via skype dial 800441234567890 etc.

 

Trunk Sequence: SIP/skype

 

That completes the basic setup for a@h but we then need to add some bits to the config files, note that these will be overwritten each you time you change settings from within a@h so do these last.

 

Go to Tools -> Config Edit

 

Click on extensions_additional.conf

 

Find the skype outroute, eg outrt-xxx-skype

 

Change the dialout line to read:-

 

exten => _8X.,1,Macro(UpLink,${EXTEN:1},,)

 

If you want to map some fixed extensions to call skype users by their skypename then add some arbitrary extensions in the the [ext-local] section, eg

 

exten => 1977,1,Macro(UpLink,skypename)

exten => 1977,hint,SIP/1977

 

This will call skype user 'skypename' when you dial 1977, note you cannot dial your own skypename to test it, I tried and it doesn't work.

 

Click update.

 

Now open extensions.conf

 

Add this to the end of the file:-

 

[macro-UpLink]

exten => s,1,Dial(SIP/${ARG1}@5506,60,)

exten => s,2,Congestion

 

If you want to follow my example completely setup a call queue, number 123 before you edit the config files.

 

Uplink setup.

 

General

 

When SIP Calls: Use the dialled number

 

When Skype calls SIP dial the following number: 123

 

Note 123 is my call queue.

 

SIP

 

Full friendly display name: skype

 

SIP account number (or user): 5506

 

Server: 192.168.1.5

 

Use the ip address of your a@h server

 

Password: 5506

 

Network

 

Local SIP Port to listen on: 5060

 

Local RTP ports....: 8000

 

Use static IP address

 

Public IP address:- xxx.xxx.xxx.xxx eg put in your public IP address

 

External SIP Port: 5060

 

External RTP port starting from: 8000

 

Restart uplink.

If everything is ok then uplink should register as sip:5506@your-a@h-ip

 

That's it.

 

This all works for me with one problem, when I dial out through skype only the first call works and then the Skype client hangs and does not disconnect, it requies a ctrl-alt-delete to kill the uplink process before the skype client will disconnect and respond. With calls to skype users this does not happen, it only happens when dialing skype out, not skype to skype. I don't know whether this is an uplink bug??? I am dialing UK numbers of the format 00441234567890

 

If I've missed anything please let me know, good luck!

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