Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
_ 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
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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Mea Culpa - I see this in my 1.4.37 source as well (line 8401 in this release chan_sip.c). Hopefully someone like Tilghman will address this; a simple hack would be to create a C daemon that did a core show channels and transmit to appropriate results back for referral. _ 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
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
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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
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
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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
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
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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
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 --
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. --
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Un-top-posting... 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. On Wed, 23 Feb 2011, vip killa wrote: 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 And what are you 'in it' for? The developer community is populated by many kinds of people. Some do it because it's their job, some do it for the challenge, some do it 'for the greater good' and some do it as 'a gun for hire.' Whatever their motivation, are you receiving more than you give? My guess is 'yes' which makes it 'fortunate' for you and me. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
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.commailto: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.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto: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.commailto: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.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto: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.commailto: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.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto: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.commailto:da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto: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
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 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
snip Asterisk is (IMO) a very good product. It is NOT a perfect product, but I'm sure that most if not all of the Commercial PBX products available are not either. You get what you pay for; In this case, you pay in time instead of actual cash (unless you use the commercial flavor of Asterisk). It all boils down to what you need and what you are willing to do/pay to get that. -- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Actually from what I understand Asterisk is the only product that has this REFER problem. I know for a fact FreeSWITCH (open-source) handles REFERs fine. On Wed, Feb 23, 2011 at 1:28 PM, Danny Nicholas da...@debsinc.com wrote: snip Asterisk is (IMO) a very good product. It is NOT a perfect product, but I’m sure that most if not all of the Commercial PBX products available are not either. You get what you pay for; In this case, you pay in time instead of actual cash (unless you use the commercial flavor of Asterisk). It all boils down to what you need and what you are willing to do/pay to get that. -- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
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: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] 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 bradley.watk...@compuware.commailto: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.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto: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.commailto: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.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto: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.commailto: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.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto: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.commailto:da...@debsinc.com wrote: I use Polycom 501's and
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? On Wed, Feb 23, 2011 at 1:38 PM, Watkins, Bradley bradley.watk...@compuware.com wrote: 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:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *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 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Sorry for the top post - this is from my phone. Sounds like the issue may actually be with the AGI that is handling your ACD queue. I've used the built-in Queue() command to handle situations like you describe without running into the issues you detailed. And that's with Polycom phones, too. Without more details, I'm not sure how much help you're going to get. Show us some console output of the issue, capture the proper debug logs, etc, and perhaps you'll find more help. Thanks, --Warren Selby, dCAP On Feb 23, 2011, at 11:57 AM, vip killa vipki...@gmail.com wrote: 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On 02/23/2011 12:43 PM, vip killa wrote: I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? Option 3 was wait for someone else with the skills and/or money necessary to fix it. Demanding that somebody fix an issue will not work in any community, open source or otherwise. You'll only be labeled a nuisance and ignored. -- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 12:44 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) I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? snip From what I see, the source fix on the Asterisk level would indeed be a major undertaking. But since you are using an AGI to control the Queue command instead of using it from the dialplan, you have more control over this problem than you realize. For simplicity of illustration, let's say your AGI simply wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the Polycom transfer from Agent007 to pussygalore, Agent007 is marked as busy in the queue although the call is no longer active for 007. One possible workaround would be to have a duplicate bail queue set up the same way. If my AGI does a core show channels and sees that 007 is not on the phone, I can do queue(bail) instead of queue(normal). -- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? In a commercial product, you have two options when you find a bug: (1) Pay for it to be fixed. (2) Live with the bug. In an open-sourced product, you have those same two options, plus an additional one: (3) Fix it yourself. Those are the only three you have to choose from. -- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On Wed, 23 Feb 2011, Danny Nicholas 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 12:44 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) I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? snip From what I see, the “source fix” on the Asterisk level would indeed be a major undertaking. But since you are using an AGI to control the Queue command instead of using it from the dialplan, you have more control over this problem than you realize. For simplicity of illustration, let’s say your AGI simply wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the Polycom transfer from Agent007 to pussygalore, Agent007 is marked as busy in the queue although the call is no longer active for 007. One possible workaround would be to have a duplicate “bail queue” set up the same way. If my AGI does a “core show channels” and sees that 007 is not on the phone, I can do queue(bail) instead of queue(normal). Watch out for race conditions doing things like this... j-- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. On Wed, Feb 23, 2011 at 2:21 PM, Richard Kenner ken...@gnat.com wrote: I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? In a commercial product, you have two options when you find a bug: (1) Pay for it to be fixed. (2) Live with the bug. In an open-sourced product, you have those same two options, plus an additional one: (3) Fix it yourself. Those are the only three you have to choose from. -- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On Wed, 23 Feb 2011, vip killa wrote: I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. You 'effing' kill me :) You have to be a troll. You can't be this stupid. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On Wed, Feb 23, 2011 at 3:43 PM, vip killa vipki...@gmail.com wrote: I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. If you're really interested in trying to resolve your issue, as opposed to just complaining about it, perhaps you can post the requested debug information[1] from earlier. [1] - https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information With the requested information, you may be surprised at the type and level of help you may get from this mailing list. If all else fails, you can always open a new issue on the bug tracker and it will get looked at. It's a pretty painless procedure. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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
