You are still focusing on ONE of the choices given when that isn't your only option. It is simply untrue to say that the answer to "it's broken" was "pay us". You were (now on multiple occasions) told how it would come to pass that a resolution will come about. You choose to ignore precisely two-thirds of the options available to you in order to continue to grind your axe.
I am convinced you are either trolling or simply myopic. You have choices, they are yours to make. Stop trying to say that the entire Asterisk development community is simply in it for money, because that is patently false. - Brad From: [email protected] [mailto:[email protected]] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 1:28 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) 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 <[email protected]<mailto:[email protected]>> 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: [email protected]<mailto:[email protected]> [mailto:[email protected]<mailto:[email protected]>] 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 <[email protected]<mailto:[email protected]>> 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: [email protected]<mailto:[email protected]> [mailto:[email protected]<mailto:[email protected]>] 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 <[email protected]<mailto:[email protected]>> 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: [email protected]<mailto:[email protected]> [mailto:[email protected]<mailto:[email protected]>] 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 <[email protected]<mailto:[email protected]>> 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: [email protected]<mailto:[email protected]> [mailto:[email protected]<mailto:[email protected]>] 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 <[email protected]<mailto:[email protected]>> wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 ________________________________ From: [email protected]<mailto:[email protected]> [mailto:[email protected]<mailto:[email protected]>] 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<http://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 <[email protected]<mailto:[email protected]>> wrote: ________________________________ From: [email protected]<mailto:[email protected]> [mailto:[email protected]<mailto:[email protected]>] 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
