[Asterisk-Users] Skype-to-Asterisk(SIP): progress

[John Todd]

>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/janedoe at sip-to-skype,100)
exten => 4454,n,Congestion


JT