Skype to SIP Software From Australia

Reply from jcgalvezv on Apr 03, 2006 - 02:17 AM
I did and they replied. They are working on this issue. Lets wait for a while. I guess they will release a fix soon.

Juan C.
Reply from pekkis on Apr 06, 2006 - 09:31 AM
has anybody tested both Uplink and PSGW? impressions? I cannot get Uplink running because of this recording device error but I would hate to buy PSGW 3.0 since 2.0 shows many problems (crashing W2000 from time to time, strange target address definition method, problems with SIP port number setting, etc.)

br

pekkis
Reply from mazilo on Apr 06, 2006 - 02:27 PM
Will this software turn itself into a super node to server Skype?
Reply from mathew on Apr 19, 2006 - 09:43 AM
the new version of uplink works quite nice.
the new ver is 1.1 you can download it from http://www.nch.com.au/skypetosip/index.html
in the new version you can define the SIP and RTP ports. that makes it easy to work with.
it is 244k while the old one was 234k
Reply from pekkis on Apr 19, 2006 - 01:18 PM
in v1.1 no improvement as to recording device error

--pekkis--
Reply from pekkis on Apr 24, 2006 - 09:41 AM
no improvement with skype 2.0.0.103 either
Reply from yiannis on Apr 27, 2006 - 12:34 AM
Specs:
Windows XP Professional
Skype 2.0.0.97
Uplink 1.1

Everything is working fine. I tried on a Windows 2000 Advanced Server but Skype is complaining about Audio Device. Truth is that I have disabled the on-board Sound Card. I will enable it, install drivers and try again. Not before next week though. I'll post results as soon as.

Yiannis
Reply from woodpecker on May 08, 2006 - 11:38 AM
Hi folks,

I am trying to get uplink 1.1 to work with asterisk@home, it seems close to working, here is what I have done based on some settings from the nch forum that another asterisk user had used.

I have setup an incoming trunk as follows:-

[5506]
username=5506
host=dynamic
nat=yes
secret=5506
allow=all
type=friend
insecure=very
context=extensions
canreinvite=yes
caninvite=yes
callerid="Incoming Skype"

In the register string I have 5506:5506@192.168.1.4/5506 and it appears to be registered ok.

I have tried setting extensions and also DIDs with the number 5506 but this is where I am confused, do I need an extension or inbound route with a number of 5506?

My asterisk curently has two extensions which I use with other voip services, 200 and 201. I also have one inbound route with a DID for each PSTN to VOIP number I have.

I am also confused about the uplink options, I have to tell uplink what to call when it gets a skype call, I have tried telling it to call 5506 or 200 and 201 but nothing will work.

When I run uplink it gives this info:-

Uplink Skype2Sip started.
SIP Private IP is 192.168.1.4:5070
SIP Public IP is 81.86.xxx.xxx:5070
SIP Number 5506 [!at] 81.86.xxx.xxx (replace the [!at] with a @):5070
Attempting to register sip:5506@192.168.1.5
Registered as sip:5506@192.168.1.5

Receiving call...
Initiated sip call to:5506
Error returned:Not Found

I have also tried routing the call to 200, 201 and 5506 [!at] 192.168.1.5 (replace the [!at] with a @) (the ip of the asterisk server) but they all return a Not Found error from uplink. I cannot figure out what it cannot find or why it cannot find it, is it the extension it cannot find?

Any suggestions please?
Thanks,
Andy.
Reply from liorh on Aug 07, 2006 - 01:41 PM
Hello
just tested Uplink and its work fine to me. its a much beter application then PSgw as it work without crashes and all sip messages are from one port only. on SPGw every registaration is from diffrent port and thier for we had 100 of contacts in our proxy.
i would recomned kindly to add outbound proxy configuration and registrar username to cover all registrations methodes.

Uplink:
-Nice and easy to operate.
-stable
-most sip signaling i tested are acording to standard.


Good work!!!
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