Comments on: Initial investigations on hooking up Skype to SIP with Asterisk: NCH Uplink, ChanSkype, and PSGw

By: Luca

Luca — Mon, 05 May 2008 15:42:44 +0000

Another valid solution that supports up to 30 concurrent Skype calls is Skip2PBX (www.skip2pbx.com).
Fully support SIP with G711 Codec.
Just for your info.

By: z_smurf

z_smurf — Wed, 16 Apr 2008 19:15:57 +0000

Regarding problems with missing /dev/ivcsfifo* entries.

I found out that the reason for theese not being created comes from the script “ivcs_load” which fails to insmod ivcs.ko.

I recompiled ivcs.ko with the same version of GCC as the kernel is compiled with, and now it works.

By: raj

raj — Tue, 21 Aug 2007 04:43:35 +0000

This blog gave a brief understanding of how to connect skypetosip.
I have tested Uplink Skype2Sip and had no problems .Sound quality was good when i tried to connect x-lite sip phone to a skype user using asterisk.
But iam wondering how could i call a sip phone from skype using asterisk.
Please any pointers on this.

By: Alex Ayala

Alex Ayala — Mon, 09 Jul 2007 03:03:25 +0000

I tested Uplink Skype2Sip and had no problems at all. Tested the following:
1) Sip Phone (Cisco) connected to Asterisk to Skype, sound quality was perfect.
2) Skype into Sip Phone, same result
3) Skype to Asterisk which route to a cell phone (going through a SIP Gateway), quality wasn’t as good as the other but I think thats because is doing alot of routing eheh..if I routed through PSTN using FXO Card I probably get a lot better sound.

Since I’m using Freebsd I don’t have ChanSkype as an option.

By: Michael Martin

Michael Martin — Thu, 14 Jun 2007 16:19:41 +0000

Could you post more explicit configuration instructions for Asterisk newbies?
Thanks

By: Georges Hannelais

Georges Hannelais — Mon, 05 Feb 2007 14:15:26 +0000

I am testing Uplink Skype2Sip with great success (V1.30).
Uplink is on our Windows server and Asterisk on the Linux one.
Pluses:
Instalation is straight forward.
Sound quality is good for Skype to Skype (incoming & outgoing) and Skype Out using Linksys SPA942 SIP phones as Asterisk extentions. Skype In has not been tested yet.
Minuses:
DTMF does not pass thru and it is a problem as far as Digital Receptionist is concerned on the Asterisk machine.

By: Jason

Jason — Fri, 08 Dec 2006 23:03:53 +0000

Have you resolved the device creation /dev/ivcsfifo*? I installed the chan_skype and the device did not get created. I sent email to chanckype support. I have not got any response. How is the chanckype support? Or basically, you are on your own once you get the software.

By: Don

Don — Mon, 27 Nov 2006 04:27:52 +0000

I haven’t been happy with chanskype…because right now when I hangup a call in asterisk the skype instance doesn’t hangup…it takes multiple vnc desktops…1 per every channel you want to run…And when you have a problem it takes a week to get a reply from someone…

By: Sharad Ahlawat

Sharad Ahlawat — Wed, 15 Nov 2006 06:57:21 +0000

Hello, thank you for posting this comparison. I am using asterisk on gentoo and am trying to configure chanskype. How did you create the /dev/ivcsfifo* entries.

Thanks in advance

By: Matthew Gast

Matthew Gast — Tue, 31 Oct 2006 16:19:48 +0000

Phil,

Absolutely! Go right ahead.

Now you’re going to motivate me to do the second part of the investigation on the current version of ChanSkype.

Matthew