Re: [asterisk-dev] confbridge feature request
Is there a way of crowd sourcing the development of features? I sure there are many people who would find this type of feature very useful. After thinking about the the use of same = n,Set(TIMEOUT(absolute)= perhaps an update to this would be better to set a marker or group number to all that have the same marker. For example same = n,Set(TIMEOUT(absolute)=time,warning,warning-message,group, As usual the time denotes the time when the call will expire, warning is the time before the warning message is played, warning message is the audio file which will be played and group puts the caller into a group. This will allow you to update the timeout absolute for all that are in the same group via dial plan. couple this with the ability to execute dialplan externally would work for all sorts of applications not just conferencing. Best regards Jonathan -Original Message- From: Matthew Jordan Sent: Tuesday, December 16, 2014 8:53 PM To: Asterisk Developers Mailing List Subject: Re: [asterisk-dev] confbridge feature request On Tue, Dec 16, 2014 at 11:10 AM, jonathan white j...@uvacity.com wrote: On 15 Dec 2014 22:49, Jonathan White uvac...@googlemail.com wrote: Good morning. I was thinking it would be very useful if we could have a feature added to confbridge which would limit the life of a bridge. For example in dialplan you could set the bridge life with the following command same = n,Set(CONFBRIDGE(bridge,max_time)= and or same = n,Set(CONFBRIDGE(bridge,max_datetime)= The function would overwrite the previous setting allowing the bridge length to be dynamically changed during the length of the call. For example the call will last for an hour from when the last caller joins that particular bridge. I’ve looked at using same = n,Set(TIMEOUT(absolute)= but this is per channel not per conference or per group of callers If this feature was implemented there maybe a second or third setting or feature required which sets and plays hangup messages 60 seconds before the end of the call and at the end of the call. same = n,Set(CONFBRIDGE(bridge,hangup_warning)= same = n,Set(CONFBRIDGE(bridge,hangup_message)= This will point to a sound file which will be played at the bridge is being closed. Sadly I have no budget or personal skill to write this patch. My contribution to the project will have to be my ideas. Best regards Jonathan That is an interesting feature idea, and would be a good addition to ConfBridge. I will say that without contributing code to the project - or hiring someone to do so - it is unlikely (although not outside the realm of possibility) that anything will come of your idea. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] asterisknow-version
I’m not sure this is the correct channel to use but can I suggest the asterisknow-version rpm is not a dependency for asterisk13 and above? I understand the reason why this was done initially but it courses more problems that it solves First asterisknow-version overwrites the issues file which we use for displaying system information Second if you de install or mark asterisknow-version not to be installed in kickstart, you can not update your asterisk installation due to asterisknow-version being a dependency Is there a way to stop asterisknow-version from overwriting the issues file? Lastly I think there is a difference with the way repository data is compiled for asterisk 12 and above. I use revisor to help generate installation media. Revisor happily collects the repo data and downloads the required RPM’s when I use the Asterisk 11 repos but Asterisk 12 and above I have download the packages locally, compile my own repo data and use a local repository rather than the official site. Is there a difference in the way the repo data is generated in asterisks 12 and above? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] confbridge feature request
Has anyone thought of using some sort of crowd funding method for new features? I'm sure I'm not the only one who could use a good feature of this type. Regards -Original Message- From: Matthew Jordan Sent: Tuesday, December 16, 2014 8:53 PM To: Asterisk Developers Mailing List Subject: Re: [asterisk-dev] confbridge feature request On Tue, Dec 16, 2014 at 11:10 AM, jonathan white j...@uvacity.com wrote: On 15 Dec 2014 22:49, Jonathan White uvac...@googlemail.com wrote: Good morning. I was thinking it would be very useful if we could have a feature added to confbridge which would limit the life of a bridge. For example in dialplan you could set the bridge life with the following command same = n,Set(CONFBRIDGE(bridge,max_time)= and or same = n,Set(CONFBRIDGE(bridge,max_datetime)= The function would overwrite the previous setting allowing the bridge length to be dynamically changed during the length of the call. For example the call will last for an hour from when the last caller joins that particular bridge. I’ve looked at using same = n,Set(TIMEOUT(absolute)= but this is per channel not per conference or per group of callers If this feature was implemented there maybe a second or third setting or feature required which sets and plays hangup messages 60 seconds before the end of the call and at the end of the call. same = n,Set(CONFBRIDGE(bridge,hangup_warning)= same = n,Set(CONFBRIDGE(bridge,hangup_message)= This will point to a sound file which will be played at the bridge is being closed. Sadly I have no budget or personal skill to write this patch. My contribution to the project will have to be my ideas. Best regards Jonathan That is an interesting feature idea, and would be a good addition to ConfBridge. I will say that without contributing code to the project - or hiring someone to do so - it is unlikely (although not outside the realm of possibility) that anything will come of your idea. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] ARI Extending Existing Feature: bridge control
This would be a very useful feature to have available in dialplan. To be able vary the length of a call, during the call. I just emailed the dev list with the thought of adding this feature to confbridge with the following type of commands: same = n,Set(CONFBRIDGE(bridge,max_time)= or same = n,Set(CONFBRIDGE(bridge,max_datetime)= or same = n,Set(CONFBRIDGE(bridge,start_datetime)= same = n,Set(CONFBRIDGE(bridge,stop_datetime)= with the following to add some notifications same = n,Set(CONFBRIDGE(bridge,hangup_warning)= same = n,Set(CONFBRIDGE(bridge,hangup_message)= The idea was that if you have some call credit it will reduce by the variable number of callers which will join the same bridge. Using same = n,Set(TIMEOUT(absolute)= only sets the time out for an individual call and not the bridge. Perhaps something that could be used for all call types would be a better approach rather than modifying individual apps. Regards From: Nir Simionovich Sent: Wednesday, December 17, 2014 8:16 PM To: Asterisk Developers Mailing List Subject: [asterisk-dev] ARI Extending Existing Feature: bridge control Hi All, After shipping out my first patch to ARI, I became hungry :-) So, now I've set up a slightly higher goal, adding a much required feature for ARI. I'll describe the problem first, then I have some questions. The Asterisk dial application enables us to limit the duration of the call and play a warning sound. Once this had been set, as far as I know, you can't modify these values externally. When the feature was originally introduced, over 10 years ago, the goal was simple: enable a calling card system to limit the call according to a credit line. As time went by, people realized that this feature is useful, however limited. In today's mobile application era, when a VoIP phone can actually purchase credit while on the actual call, we need a way to control this from an external source. Now, I've started digging into the code, and I've managed to understand that following (feel free to bash me if I'm wrong): 1. The time limits are maintained at the bridge structure, not at the channel - using the ast_bridge_config data structure 2. The ARI bridges GET method only retrieves a list of bridges and their associated channels, not their configurations So, assuming that I'm reading the ast_ari_bridges_list function from resouce_bridges.c correctly, we retrieve a snapshot of all active bridges via the snapshots variable (if you can call it that). The output is built by iterating through it. Now, my questions: 1. Is there a way to obtain the information in ast_bridge_config for each of the iterated bridges, then output it via the JSON response? 2. Is there a way to manipulate the configuration of the bridge, via modifying the associated bridge configuration? The floor is now open :-) Nir S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] confbridge feature request
On 15 Dec 2014 22:49, Jonathan White uvac...@googlemail.com wrote: Good morning. I was thinking it would be very useful if we could have a feature added to confbridge which would limit the life of a bridge. For example in dialplan you could set the bridge life with the following command same = n,Set(CONFBRIDGE(bridge,max_time)= and or same = n,Set(CONFBRIDGE(bridge,max_datetime)= The function would overwrite the previous setting allowing the bridge length to be dynamically changed during the length of the call. For example the call will last for an hour from when the last caller joins that particular bridge. I’ve looked at using same = n,Set(TIMEOUT(absolute)= but this is per channel not per conference or per group of callers If this feature was implemented there maybe a second or third setting or feature required which sets and plays hangup messages 60 seconds before the end of the call and at the end of the call. same = n,Set(CONFBRIDGE(bridge,hangup_warning)= same = n,Set(CONFBRIDGE(bridge,hangup_message)= This will point to a sound file which will be played at the bridge is being closed. Sadly I have no budget or personal skill to write this patch. My contribution to the project will have to be my ideas. Best regards Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Add new option to Queue function
Just something I know which may restrict what can be done. Avaya have many patents for call distribution. This includes call distribution to agents who have spent the least amount of time on the phone and taken the lowest number of calls. On 25 Apr 2014 15:00, Nguyen Hoang Son nh...@vasc.com.vn wrote: Hi all, I'm using Queue function of Asterisk to arrange calls which is coming to my agents. I want to customize the way asterisk arrange coming call, in other word, is it possible to create a new option instead of using the existing: RINGALL, ROUNDROBIN,... . For example: The incoming call should come to the argent who has the most waiting time (idle time). I find out that the algorithm of each option of Queue is defined in app_queue.c in the source code but I don't know how to change, how to add the waiting time as a new option to sort by. This question is quite related to the development of asterisk, so please help if you have any idea or experience on that. Thank you very much. --- *NGUYỄN HOÀNG SƠN* M-Commerce Center VASC Software and Media Company - VNPT Addr: No.97 Nguyen Chi Thanh Street, Dong Da District, Hanoi, Vietnam Cell phone: +84 912998101 Skype: hoangsonk49 E-mail: nh...@vasc.com.vn -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Function Read - Timeout
Just a quick suggestion to enhance function Read. I am using function read in places to provide options to skip announcements or provide hidden features. However the minimum timeout is 1 second which puts an unnatural pause in the flow of announcements when not skipping. It would be great if there was a parameter not to wait for digits. Possible? Best regards J --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Function Read - Timeout
Yes that's a good idea. This fixes one of my issues however it doesn't when
I have two reads one after the other.
It would still be good if there was a parameter to have no delay.
J
-Original Message-
From: Eric Wieling
Sent: Friday, April 25, 2014 8:54 PM
To: Jonathan White ; Asterisk Developers Mailing List
Subject: RE: [asterisk-dev] Function Read - Timeout
You're holding it wrong.
There are several ways to accomplish this, the easiest is to play all sound
files with one Read, like:
Read(fwdto,call-fwd-unconditionalplease-enter-thedigits/11digit/igc/sounds/destinationtelephone-number,11,,1,6)
If you can't play all the sound files with one Read, then use WaitExten and
Background:
exten = s,1,Noop(Switch Manager IVR)
same = n,Answer
same = n,Ringing
same = n,Wait(1)
same = n,Set(LOCAL(count)=0)
same = n,While($[${count} 4])
same = n,Set(count=$[${count}+1])
same = n,Background(please-enter-the/igc/sounds/destinationnumber)
same = n,WaitExten(5)
same = n,EndWhile()
same = n,Playback(goodbye)
-Original Message-
From: asterisk-dev-boun...@lists.digium.com
[mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Jonathan White
Sent: Friday, April 25, 2014 3:12 PM
To: asterisk-dev@lists.digium.com
Subject: [asterisk-dev] Function Read - Timeout
Just a quick suggestion to enhance function Read.
I am using function read in places to provide options to skip announcements
or provide hidden features. However the minimum timeout is 1 second which
puts an unnatural pause in the flow of announcements when not skipping.
It would be great if there was a parameter not to wait for digits. Possible?
Best regards
J
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Re: [asterisk-dev] CentOS packaging
I’m not sure this is the right place for me to make a request but is it totally necessary to have asterisknow-version a dependency? During my kickstart the package is installed and I can’t exclude it and thus I have to uninstall it afterwards. It does look like it does much. Thanks From: David M. Lee Sent: Friday, February 28, 2014 5:30 PM To: Asterisk Developers Mailing List Subject: Re: [asterisk-dev] CentOS packaging On Feb 28, 2014, at 10:23 AM, David M. Lee d...@digium.com wrote: On Feb 27, 2014, at 3:05 PM, Jared Smith jaredsm...@jaredsmith.net wrote: On Wed, Feb 26, 2014 at 3:51 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: This is funny. In 11 pjproject is bundled (though patching it out is not an issue). In 12 it is not included in the tree anymore. I know, and the humor isn't lost on me. :-) That being said, I'm hoping to get the Asterisk fork of pjproject packaged up in Fedora/EPEL shortly, so that we can begin getting Asterisk 12 packaged. Now that PJSIP 2.2 have been released, I hope to update our fork shortly. Since all of our patches were either accepted upstream, or backported from trunk, I hope that means that the Asterisk fork will be identical to the PJSIP 2.2 release. That went smoother than I thought it would. Compiled, installed and test call worked without issue. Props to file for keeping up getting our stuff working with PJSIP trunk a few months back. The master branch on github.com/asterisk/pjproject is now identical to PJSIP 2.2. -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [r400723-400741] ConfBridge now has the ability to set the language of announcements
Thanks for the reply. Currently this is expected behaviour. It would be better from a user perspective to hear a prompt on the language the user selected not that of the bridge. I set the variable in dialplan so maybe different for each international user who joins. If a German user joins before an English or Spanish user all users hear prompts for joiners and leavers in German. It would sound more professional if each user heard prompts in their own language. I the the overhead for this would be quite high depending on how many languages there are. Perhaps an additional setting to set a default language if multiple languages are requested would work rather than fixing it to the first joiner. This way we can default back to a more common language of English. Regards On 7 Feb 2014 02:06, Richard Mudgett rmudg...@digium.com wrote: On Thu, Feb 6, 2014 at 6:40 PM, Jonathan White j...@uvacity.com wrote: Good afternoon. Thanks for adding this feature. I have been testing it today and notice some unexpected behaviour. When multiple users call in and set different languages they will only hear the language set by the first caller to join the conference. Is this the expected and desired behaviour? I would have expected the desired result would be for each caller who sets their own language to hear the prompts they selected not the language of the first caller. That option sets the language of the conference bridge itself so any prompts played to the bridge get played in the selected language of the bridge. Prompts played to the bridge are heard by everyone in the bridge at the same time. Prompts played to a specific user would be in that users language and are heard by that user only. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [r400723-400741] ConfBridge now has the ability to set the language of announcements
Good afternoon. Thanks for adding this feature. I have been testing it today and notice some unexpected behaviour.When multiple users call in and set different languages they will only hear the language set by the first caller to join the conference.Is this the expected and desired behaviour?I would have expected the desired result would be for each caller who sets their own language to hear the prompts they selected not the language of the first caller.Below is are a copy of the relating release notesMany thanks 2013-10-08 20:14 + [r400723-400741] Richard Mudgett rmudg...@digium.com * UPGRADE.txt, apps/app_confbridge.c, apps/confbridge/conf_config_parser.c, configs/confbridge.conf.sample, apps/confbridge/include/confbridge.h: app_confbridge: Can now set the language used for announcements to the conference. ConfBridge now has the ability to set the language of announcements to the conference. The language can be set on a bridge profile in confbridge.conf or by the dialplan function CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983) Reported by: Jonathan White Patches: M19983_rev2.diff (license #5138) patch uploaded by junky (modified) Tested by: rmudgett * apps/confbridge/conf_config_parser.c: app_confbridge: Fix duplicate default_user profile. * Fixed looking in the wrong profiles container to see if the default_user profile is already created in verify_default_profiles(). The bridge profile container is never going to hold user profiles. :) --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
