It's simple, if a product is broken shouldn't it be fixed? In this case the answer is "for a price" which is absurd because it is an open source product. If there was a decent community of developers surrounding this "open source project", it would be fixed simply because it's broken, no questions asked.
On Wed, Feb 23, 2011 at 1:19 PM, Watkins, Bradley < bradley.watk...@compuware.com> wrote: > Implying that the Asterisk developers (which is itself a fairly nebulous > statement since those who contribute to Asterisk are many and come from > different companies/countries/etc.) are “not in it to make a good product” > but to make a “profit” is not only highly insulting but a complete > mischaracterization of what you were told on IRC. > > > > What you were told was that there are essentially three choices (and this > goes for pretty much any open source software, as already stated). > > > > You may either fix it yourself (if you have the skills), pay someone to fix > it for you (if you can or must trade money for expediency), or wait for > someone else with the skills and/or money necessary to fix it. > > > > Regards, > > - Brad > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa > *Sent:* Wednesday, February 23, 2011 1:05 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented > inchan_sip.c on line 11951) > > > > Yes, they want money, they've told me that several times...it's unfortunate > that asterisk's dev community is not in it to make a good product but a > profit > > On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas <da...@debsinc.com> wrote: > > My bad – “natively” means using the Queue command from the dialplan. Since > the “powers that be” are aware of this problem, I suppose it will get fixed > when somebody either has some spare time or a sufficient bounty is offered. > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa > *Sent:* Wednesday, February 23, 2011 11:57 AM > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented > inchan_sip.c on line 11951) > > > > I'm sorry i don't know what you mean by natively. I'm almost certain the > queue is handled via AGI and not using asterisk's queue. > > On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas <da...@debsinc.com> > wrote: > > Do you use the Queue command “natively” or from the AGI? In the example > you gave, if you did a “core show channels”, I assume that Agent007 would be > idle, but ineligible for Queue activity. > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa > *Sent:* Wednesday, February 23, 2011 11:37 AM > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented > inchan_sip.c on line 11951) > > > > Sure, it really manifests itself whenever using AGI for call flow, but this > is how it affects us... > > incoming call -> queue -> agent007 -> xfer -> pussygalore > > now the AGI/dialplan thinks agent007 is on phone with pussygalore until > that xfered call terminates so if another call comes into queue while > pussygalore is on the phone w/ that xfered call, agent007 will not even be > attempted by queue > > > > I'm sure there could be other scenarios in which this REFER issue could > pose a problem but this is the most consequential scenario which we have to > deal with everyday. > > > > > > On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas <da...@debsinc.com> > wrote: > > I use Polycom 501’s and use the Transfer Key to send inbound calls to other > extensions. Can you give me an A-B-C example of how this problem manifests > itself? > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa > *Sent:* Wednesday, February 23, 2011 11:11 AM > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented > inchan_sip.c on line 11951) > > > > Interesting but the issue I'm having relates to Inbound and Outbound REFERs > since I'm using Polycom's Transfer softkey (which allows for both Inbound > and Outbound Transfers). I know this is not an issue when using Asterisk's > built-in transfer (only allows Inbound transfers). > > > > On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas <da...@debsinc.com> > wrote: > > Have you read this thread? > > http://forums.digium.com/viewtopic.php?t=74418 > > > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa > *Sent:* Wednesday, February 23, 2011 10:36 AM > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented > inchan_sip.c on line 11951) > > > > I did not see this issue anywhere on issues.asterisk.org > > Can you give me a reference number to the issue? Also, it is a problem with > all releases of asterisk. > > On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas <da...@debsinc.com> > wrote: > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa > *Sent:* Wednesday, February 23, 2011 10:11 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] REFER and dialplan broken (as documented > inchan_sip.c on line 11951) > > > > There is a problem when transferring calls using REFER, asterisk does not > notify dialplan. I've been told to use AMI as a workaround to notify my > dialplan/routing program but that would require a huge change to our > software. I was wondering if there is any intention of fixing this problem. > > Here is issue as stated in chan_sip.c > > "this is currently broken as we have no way of telling the dialplan engine > whether a transfer succeeds or fails." > > Thanks. > > > > I’m quite certain that this problem is being considered (for reference, > this is a 1.8.X issue). If you aren’t satisfied with the progress being > made, you should research your own solution and/or offer a bounty. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users