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
Kudos!!! 'Nuf said!> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > John Todd > Sent: Thursday, January 26, 2006 2:44 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Skype-to-Asterisk(SIP): progress > > > 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 > _______________________________________________ > --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 >
> 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.Thanks for your post, John. I too encounter resistance when I ask subcontractors in other countries to use X-Ten or other clients to connect to our pbx. The invariable, "ya, I use Skype" doesn't inspire me, since I'd have to be a a computer to use it too. You almost have to mail them a hardware phone to get them to do it. (Then they're hooked by the way.) Cellphones have lessened the importance of being able to reach someone in the same city by using their pbx directly (alas, they'll always find you if your cell is on), but the story is still the same for people in different parts of the world. It'd be great if * could talk to Skype, especially natively. Maybe someday it will be an advantage to Skype but right now it's like you say - competition.
>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. > >JTUpdate: 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@sip-to-skype,100) exten => 4454,n,Congestion JT
> 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.Has the developer indicated to you whether this would be a free upgrade for existing clients or whether additional payments would be expected? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons
I want to try this http://www.rsdevs.com/psgw_sip3.shtml but is it worth spending $40? Feedback to the list from any one who tried would be useful. On 2/1/06, John Todd <jtodd@loligo.com> wrote:> > > 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. > > > >Has the developer indicated to you whether this would be a free upgrade for > >existing clients or whether additional payments would be expected? > > > >Regards, > > > >Chris > >-- > >C.M. Bagnall, Director, Minotaur I.T. Limited > >This email is made from 100% recycled electrons > > No, there has been no indication of this but it hasn't been > discussed, either. If you are a registered subscriber, perhaps it > may be best to ask. It sounds reasonable to _me_ at least... > > 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 >