I'm sitting in the Emerging Telephony Conference, so this seems a
particularly apt place to pre-announce this...
I've wanted to be able to gateway calls between Skype and Asterisk
for a while, which of course would require some type of protocol
converter (IAX or SIP to Skype, probably.) This of course is
directly not in Skype's interest, since they would like to keep the
network closed (boo!) so that users are forced to use their PSTN
gateway and other revenue-generating systems. On the other hand,
I'm trying to crack this open so that any VoIP channel can talk to
any other VoIP channel. Asterisk provides the ideal platform for
this type of conversion, if only Skype were accessible...
Please hold flames about how Skype is the enemy of open telephony
standards. I don't disagree. However, for a small sub-set of users
that I work with, Skype is a channel that is preferred for audio in
some circumstances, and I feel that it's worthwhile to have some
ability to connect with users who have expressed that preference.
There exists a commercial program called "PSGW"
(http://www.rsdevs.com/) which runs on (booo!) Windows and does SIP
to Skype conversion. It's about $29 USD. It uses the Skype API to
create calls in both directions, and then uses somewhat of a kludge
using software audio "cables" between a SIP/RTP driver system and
the Skype API. It works reasonably well, but to date has been
somewhat limited because it will only terminate calls to a specific
Skype user on the far end which is mapped in the program itself.
This has been somewhat limiting, since that means I can't
arbitrarily specify a user in the SIP invite to whom I want to
communicate.
I have contacted the company (programmer) that sells this software,
and I've negotiated a payment to him to patch the code such that
PSGW will allow arbitrary specification of Skype-side user choice,
as I've asked that this be released as part of the general
distribution of this commercial software. He says that this should
be ready within the next week or two for testing by me, and then
I've asked that the code is released into the next versions of PSGW.
So basically, I'm putting out a press release about someone else's
commercial software, but I think it's worth noting because of the
usefulness of this when used in conjunction with Asterisk.
I'll keep the list updated with the progress of the code and tests
with Asterisk.
JT
Update:
I have the code here, and I've been testing for a day or so. It
does work as requested, so now I have at least one-way many-to-many
communications into the Skype network. The developer has indicated
that a revised version of the PSGW (http://www.rsdevs.com/) code will
be available for sale shortly with the changes.
I haven't had much luck getting calls from Skype->SIP yet, but that
is probably a codec problem and I'm waiting on word of what the magic
incantation is to make everything match up. I've tried unlimiting my
codec choices, but it still seems that the PSGW software is unhappy
with the
list and sends a BYE at the moment the call is connected. I know
that this can be made to work, but I just don't have the right trick.
Synopsis of use:
The SIP gateway running on the Windows machine is configured as any
other peer/trunk. I have created a "dummy" Skype user, which is used
only for outbound calls into the Skype network. People will get
accustomed to seeing the "dummy" account when the office PBX needs to
get in touch with them.
[sip-to-skype]
type=friend
secret=blahpasswordhere
host=dynamic
context=intern
canreinvite=no
dtmfmode=rfc2833
nat=no
disallow=all
allow=ulaw
allow=alaw
Then, my dialplan segments look something like this:
; Call Jane on all her contact methods
; SIP/4454 = her Desk phone
; 12125551212 = her cell phone
; janedoe = her Skype ID
;
exten => 4454,1,Dial(SIP/4454&Zap/g1/12125551212&SIP/[EMAIL PROTECTED],100)
exten => 4454,n,Congestion
JT
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
