Re: [Asterisk-Dev] SCM discussion
Steven wrote: A concern I have about the distributed development model is in getting the code back to Digium. In the current centralized development model, I submit patches to Digium for inclusion. The submission along with the disclaimer on file show an active interest in Digium using the code. In a distributed model, I wouldn't need to be actively submitting code back for it to be easily available. I don't know what new mechanism will need to be adopted to keep all that clear. Yes, that is an issue as well, and one of the big reasons that I want a central development server, to kind of keep things a little 'closer' to Digium. Also, I have in mind quite a bit of automation that the developer would get for free by using the development server. This would include: - public and private trees, with public trees automatically made visible via a web interface; any tree could have access granted to other developers, allowing for team development and testing of complex patches - commits to public trees posted to a common commits mailing list - commit triggers to notify developers when the changes in their public trees no longer merge cleanly with the HEAD branch (or whatever branch they are tracking) - links from Mantis into the development server so that issues can be tracked with the development tree they are based on Certainly anyone running their own public repository/tree could provide these things, but I think there is value in them being available at no cost to a member of the team, just by hosting their work on the 'official' development server. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Linux leaves Bitkeeper: quite a dustup
On Tue, 2005-04-12 at 20:28 -0700, Kevin P. Fleming wrote: > Steven Critchfield wrote: > > > For my office we have used subversion. We have been pretty happy with it > > as well. But the reason I suggested we look at arch is that it would be > > really nice to help merge in some of the foreign patch sets into an > > easier to get and maintain branch. > > Exactly... one of my goals (I'll post more details in the other thread) > is to allow members of the 'team' to have a public area on a development > server to make their patches available for testing/comments/etc, in a > form that people can easily download and automatically update (without > requiring monitoring a Mantis bug and downloading/applying new patches). > While everyone can do this now by setting up their own CVS/SVN/etc. > public server, that's a lot to ask (and manage) for someone who just > wants to help out with development and bug fixing. So that is another reason I think we may need arch. Instead of bogging Digium down further with the administration of the various repositories, let those of us who are willing to contribute back host our own repositories. One of the big attractive features is the ability to pick and choose from multiple repositories. Arch is supposed to break down the commits into smaller section than we are used to with CVS or SVN so you can even pick apart a commit and get only what you want from other people's repositories. I look forward to a time where we can have someone tracking Digum's HEAD, Steve Underwood's spandsp tree already merged with HEAD, and maybe someone needs Kapejod(sp?)'s ZapBRI code again already merged. In that example, the developers already are hosting their own distribution sites. They could create their own extensions to the Digium HEAD repository. From that, we as the users or other developers could pick and choose what we need to get what we want done. This type of management will splinter a few small portions out, but will keep those splinters close to the core group. I would hope it would allow people easier ways to explore new ideas and mix and merge all the best of breed functions. Hopefully we could remove some of the work required to organize the development from Digium as sole gatekeeper to Digium as primary gatekeeper. A concern I have about the distributed development model is in getting the code back to Digium. In the current centralized development model, I submit patches to Digium for inclusion. The submission along with the disclaimer on file show an active interest in Digium using the code. In a distributed model, I wouldn't need to be actively submitting code back for it to be easily available. I don't know what new mechanism will need to be adopted to keep all that clear. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Channels, jitterbuffer and PLC?
Steve Kann wrote: Also, not written here, is the alaw to ulaw case; Here, is the only transcoding that does not go through slinear, and PLC isn't applied in this case either, but, I think that if that transcoder would pass through INTERP frames, the xlaw PLC stuff described above would do it's magic then. Some recent changes to CVS may have changed this behavior behind your back :-) When Asterisk sets up a transcoding path now, it _always_ goes through slinear, unless the new optimization is turned off (via a setting in asterisk.conf). Yes, this means that xlaw-to-xlaw conversion is somewhat slower, but it also means that all applications that want to deal with slinear on a bridge can do so easily. It may also mean that xlaw-to-xlaw will, in fact, cause PLC to work its magic, but I can't say for sure. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Linux leaves Bitkeeper: quite a dustup
Steven Critchfield wrote: For my office we have used subversion. We have been pretty happy with it as well. But the reason I suggested we look at arch is that it would be really nice to help merge in some of the foreign patch sets into an easier to get and maintain branch. Exactly... one of my goals (I'll post more details in the other thread) is to allow members of the 'team' to have a public area on a development server to make their patches available for testing/comments/etc, in a form that people can easily download and automatically update (without requiring monitoring a Mantis bug and downloading/applying new patches). While everyone can do this now by setting up their own CVS/SVN/etc. public server, that's a lot to ask (and manage) for someone who just wants to help out with development and bug fixing. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Linux leaves Bitkeeper: quite a dustup
Paul Querna wrote: Huh? Subversion can have commit permissions per-branch, per-directory, or even per-file: That's news to me... Based on my reading of the SVN book at red-bean, the only way this can be accomplished is by performing all SVN access through the Apache DAV module. This means that 'regular users' can't add/remove access to their own branches, it requires editing Apache configuration files. If there's some other way SVN can do this, please enlighten me :-) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
RE: [Asterisk-Dev] GPL Violation http://www.asteriskwin32.com/
Or perhapse, as he says on his website: and AsteriskWin32 cygwin patches, avaible soon, come back later ! We should just come back later ;) > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Jerris, Michael MI > Sent: Tuesday, April 12, 2005 7:38 AM > To: Asterisk Developers Mailing List > Subject: RE: [Asterisk-Dev] GPL Violation > http://www.asteriskwin32.com/ > > As I have source for a semi working * on cygwin (but derived > from somone elses work, and not complete or distributed) and > I am working on a another windows port for donation to the > tree, I can say there are most definitely code changes > necessary. This guy promissed me the source would be > available several weeks ago, and has not responded to > subsiquent e-mails after that date. I think somone at digium > needs to give this guy a ring as he is making money off > breaking their copywrite. > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Derek > > Smithies > > Sent: Tuesday, April 12, 2005 12:58 AM > > To: Asterisk Developers Mailing List > > Subject: Re: [Asterisk-Dev] GPL Violation > > http://www.asteriskwin32.com/ > > > > Hi, > > yes/no. > > > > The question is: has the guy sold product? > > > > If he is still developing, but not sold product, then fine. > > There is no GPL violation. > > > > > > If he is still developing, and has sold product, then there is a > > possible violation. > > > > You see, suppose he has been required to write Makefiles > and configure > > commands to get it to work (but not changed the asterisk > source) he is > > not required to release. > > > > > > > > Derek. > > > > > > > > > > On Mon, 11 Apr 2005, Johnathan Corgan wrote: > > > > > Preston Garrison wrote: > > > > > > > Damn give the guy a break, if it was me i would tell you > > all where > > > > you could shove it. > > yeah - I really want to give him a break. I really hate these > > developer discussions that take up so much time... > > > > > > > > His promises notwithstanding, though, he actually really is > > infringing > > > on the Asterisk copyright at this point. > >Nope - we do not know this for certain. > > > > -- > > Derek Smithies Ph.D. > > IndraNet Technologies Ltd. > > Email: [EMAIL PROTECTED] > > ph +64 3 365 6485 > > Web: http://www.indranet-technologies.com/ > > > > ___ > > Asterisk-Dev mailing list > > Asterisk-Dev@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-dev > > > ___ > Asterisk-Dev mailing list > Asterisk-Dev@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] GPL Violation http://www.asteriskwin32.com/
On Tue, 12 Apr 2005, Derek Smithies wrote: > The question is: has the guy sold product? > > If he is still developing, but not sold product, then fine. > There is no GPL violation. > > If he is still developing, and has sold product, then there is a > possible violation. > > You see, suppose he has been required to write Makefiles and configure > commands to get it to work (but not changed the asterisk source) he is > not required to release. Wrong. This has absolutely nothing to do about software being 'sold'. And it also has nothing to do with 'additions vs modifications'. He distributes binaries. Binaries contain GPL'd code. Thus, he must provide *all* of the source code necessary to build his binaries, yes, that would include his Makefiles and whatever else. -alex ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] GPL Violation http://www.asteriskwin32.com/
On Mon, 11 Apr 2005, Preston Garrison wrote: > Damn give the guy a break, if it was me i would tell you all where you > could shove it. If this was you, me and FSF would already be filing lawsuits. ;) -alex ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
RE: [Asterisk-Dev] Re: AgentLogin to MeetMe conference?
On the wiki there is a bunch of extension logic for agent login that you could modify to do the trick. As for if this is -dev or -users, it does not look like this is anything you would want to do in code, so extension logic is probably better for your needs. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Tony Mountifield > Sent: Tuesday, April 12, 2005 4:16 AM > To: asterisk-dev@lists.digium.com > Subject: [Asterisk-Dev] Re: AgentLogin to MeetMe conference? > > In article <[EMAIL PROTECTED]>, > Steve Edwards <[EMAIL PROTECTED]> wrote: > > I posed this question on the user list, but nobody bit :) > > > > > How can I configure AgentLogin to connect the agent to a > MeetMe conference? > > > > > > Or, can I achieve similar functionality through other means? > > > > I wasn't really expecting it to just be a configuration issue, just > > wishing. > > > > I guess I'm going to need to dive into the code. > > > > Anybody care to suggest how to accomplish the following? > > > > I want the agent to log in (authenticate). > > > > I want to be able to control (via the queue.conf file or something > > similar) which conference the agent is sent to. Basically, > I'd like to > > be able to say an agent can handle calls from queue 1, 2, 3 > or, I want > > the agent to be sent to conference 4 upon login and stay there. > > > > While I can probably cobble up something with extension > logic, I think > > the agent concept would be cleaner and more consistent > > I think you need to define (or explain) the logic a bit more > completely. > > What I THINK you're saying in your above example is something like: > > a) The agent logs in. If there are calls waiting on the > queue(s), then the agent gets given those call(s) until the > queue(s) is/are empty. > > b) When the queue(s) is/are empty, the agent gets put into > the conference. > > c) What happens when the next call comes in on a queue? Does > the agent get pulled out of the conference automatically to > answer the call? If so, what if the agent was in the middle > of saying something important in the conference (assuming he > is in the conf to talk as well as listen)? > > It might help people to suggest approaches if you explain > what problem you are trying to solve - what you are > fundamentally trying to achieve. > > It may still be a -users topic, but we can't tell at the moment. > > Cheers > Tony > -- > Tony Mountifield > Work: [EMAIL PROTECTED] - http://www.softins.co.uk > Play: [EMAIL PROTECTED] - http://tony.mountifield.org > ___ > Asterisk-Dev mailing list > Asterisk-Dev@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
RE: [Asterisk-Dev] GPL Violation http://www.asteriskwin32.com/
As I have source for a semi working * on cygwin (but derived from somone elses work, and not complete or distributed) and I am working on a another windows port for donation to the tree, I can say there are most definitely code changes necessary. This guy promissed me the source would be available several weeks ago, and has not responded to subsiquent e-mails after that date. I think somone at digium needs to give this guy a ring as he is making money off breaking their copywrite. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Derek Smithies > Sent: Tuesday, April 12, 2005 12:58 AM > To: Asterisk Developers Mailing List > Subject: Re: [Asterisk-Dev] GPL Violation > http://www.asteriskwin32.com/ > > Hi, > yes/no. > > The question is: has the guy sold product? > > If he is still developing, but not sold product, then fine. > There is no GPL violation. > > > If he is still developing, and has sold product, then there > is a possible violation. > > You see, suppose he has been required to write Makefiles and > configure commands to get it to work (but not changed the > asterisk source) he is not required to release. > > > > Derek. > > > > > On Mon, 11 Apr 2005, Johnathan Corgan wrote: > > > Preston Garrison wrote: > > > > > Damn give the guy a break, if it was me i would tell you > all where > > > you could shove it. > yeah - I really want to give him a break. I really hate these > developer discussions that take up so much time... > > > > > His promises notwithstanding, though, he actually really is > infringing > > on the Asterisk copyright at this point. >Nope - we do not know this for certain. > > -- > Derek Smithies Ph.D. > IndraNet Technologies Ltd. > Email: [EMAIL PROTECTED] > ph +64 3 365 6485 > Web: http://www.indranet-technologies.com/ > > ___ > Asterisk-Dev mailing list > Asterisk-Dev@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] strange Q931 SETUP coming from Cisco VIC-2BRI
Hello Everyone, I am working with a J4BRI and it works ok with our national provider ( TelecomItalia ) and when connected to ISDN pabx ( Ericsson and Tenovis ). We now have switch to a new telco provider ( Fastweb ) which brings you S0 via Cisco 1760 VIC-2BRI interface. They say ( and Cisco says ) that VIC-2BRI emulate perfectly the national isdn standard, but I found that it is not true. First of all I have to switch 3-4 and 5-6 pin of the isdn cable, otherways is not going to work, not too bad but anyway... The critical thing is that I can not receive calls! The normal ( telco and pabx ) Q931 SETUP message arrives like this: 08 01 2c 05 a1 04 03 80 90 a3 where: 08 means Q931 message 01 length of call reference field 2c call reference field a1 sending complete 04 next is bearer cap 03 80 90 is bearer cap and so on when I connect the J4BRI to the VIC-2BRI I receive 08 01 2f 05 04 03 80 90 a3 ... a1 so the a1 ( sending complete ) arrives at the end of the SETUP message. After the bearer cap , the calling number and the called number. Asterisk does not recognize this byte stream as a SETUP message and so discards it. I post it here as in asterisk-users no one seems to be able to help me Best regards ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Manager Redirect - Possible bug ?
Hi there, I have being trying to use the Manager API redirect command. I have done so with initial sucess, however if I try and redirect an already redirected channel the far end (calling party) is never sent an update invite with updated SDP and as a consequence does not hear anything. I am using purely SIP and here is a bit more detail. 1. 123456 Calls and is put into a queue 2. I use the manager interface to redirect the call to a SIP phone, sayy 8339 3. Call gets redirected. If 8339 answers, all is well so far. 4. If I now use the manager API to put the caller on hold (transfer to an extension 876), the calling party heres no MOH Redirecting again to 8339 works fine. Looking at ethereal traces it appears that after step 4 although the cli shows that the call has been successfully redirected to extension 876 and HOH is started, the calling party (123456) is still sending RTP to 8339, this is because it never received intstructions to stop sending. Is this a bug, or am I missing something. Relevant parts of my extensions.conf are reproduced below. exten => 8339,1,Dial(SIP/8339) exten => 876,1,Answer exten => 876,2,MusicOnHold(Default) Note : If Initially redirect to 876 in the first place, that works and caller hears music on hold. Please reply with comments, so that if this is a bug I can raise it as such. Thanks Umar ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] GPL Violation http://www.asteriskwin32.com/
On Tue, 2005-04-12 at 15:51 +1200, Derek Smithies wrote: > There is a clear intention to release a working codebase, which is > commendable. That intention certainly isn't clear to _me_. I cannot see any reason why the release is considered stable enough for the _binaries_ to be made available but not the source -- that seems backwards to me. It sounds like just a convenient excuse, and repeatedly pushing the date back would seem to reinforce that interpretation. -- dwmw2 ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] GPL Violation http://www.asteriskwin32.com/
On Tue, 2005-04-12 at 16:57 +1200, Derek Smithies wrote: > The question is: has the guy sold product? > > If he is still developing, but not sold product, then fine. > There is no GPL violation. This is incorrect. The GPL comes into effect when you _distribute_ the program. Sales have absolutely nothing to do with it. > You see, suppose he has been required to write Makefiles and configure > commands to get it to work (but not changed the asterisk source) he is > not required to release. The source code for a work means the preferred form of the work for making modifications to it. For an executable work, complete source code means all the source code for all modules it contains, plus any associated interface definition files, plus the scripts used to control compilation and installation of the executable. -- dwmw2 ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Re: AgentLogin to MeetMe conference?
In article <[EMAIL PROTECTED]>, Steve Edwards <[EMAIL PROTECTED]> wrote: > I posed this question on the user list, but nobody bit :) > > > How can I configure AgentLogin to connect the agent to a MeetMe conference? > > > > Or, can I achieve similar functionality through other means? > > I wasn't really expecting it to just be a configuration issue, just > wishing. > > I guess I'm going to need to dive into the code. > > Anybody care to suggest how to accomplish the following? > > I want the agent to log in (authenticate). > > I want to be able to control (via the queue.conf file or something > similar) which conference the agent is sent to. Basically, I'd like to > be able to say an agent can handle calls from queue 1, 2, 3 or, I want > the agent to be sent to conference 4 upon login and stay there. > > While I can probably cobble up something with extension logic, I think > the agent concept would be cleaner and more consistent I think you need to define (or explain) the logic a bit more completely. What I THINK you're saying in your above example is something like: a) The agent logs in. If there are calls waiting on the queue(s), then the agent gets given those call(s) until the queue(s) is/are empty. b) When the queue(s) is/are empty, the agent gets put into the conference. c) What happens when the next call comes in on a queue? Does the agent get pulled out of the conference automatically to answer the call? If so, what if the agent was in the middle of saying something important in the conference (assuming he is in the conf to talk as well as listen)? It might help people to suggest approaches if you explain what problem you are trying to solve - what you are fundamentally trying to achieve. It may still be a -users topic, but we can't tell at the moment. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] AgentLogin to MeetMe conference?
I posed this question on the user list, but nobody bit :) How can I configure AgentLogin to connect the agent to a MeetMe conference? Or, can I achieve similar functionality through other means? I wasn't really expecting it to just be a configuration issue, just wishing. I guess I'm going to need to dive into the code. Anybody care to suggest how to accomplish the following? I want the agent to log in (authenticate). I want to be able to control (via the queue.conf file or something similar) which conference the agent is sent to. Basically, I'd like to be able to say an agent can handle calls from queue 1, 2, 3 or, I want the agent to be sent to conference 4 upon login and stay there. While I can probably cobble up something with extension logic, I think the agent concept would be cleaner and more consistent Suggestions welcomed :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
