Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available
Hey, There is a way to use the new confbridg without installing the new version of Asterisk, I'm new in using asterisk, my is version 1.6.2 of Asterisk. I would not want just to get into the installation of a new version just for a one commend which I have very great need. Thank you for your help. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Development Team Sent: Thursday, November 10, 2011 6:39 PM To: Asterisk Development Team Subject: [asterisk-users] Asterisk 10.0.0-rc1 Now Available The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list. All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk. Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions A short list of features includes: * T.38 gateway functionality has been added to res_fax. * Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz (More information available at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found in the CHANGES file. http://svnview.digium.com/svn/asterisk/branches/10/CHANGES For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 10.0.0-rc1 Thank you for your continued support of Asterisk! -- _ -- 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
[asterisk-users] Does Asterisk Support SIP Video Call ?
Hi all, I tried making a video SIP call using Asterisk But it didnt workonly voice call works? Regards Faraj Khasib -- _ -- 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] Does Asterisk Support SIP Video Call ?
Hey, Did you try google.com for this! I've done this several times now. Video for one-to-one call works if H264 is supported at both end points. All you need to do is enable video in sip.conf and set allow=h264 in the sip peers with video capability. You may need to see if your asterisk has h264 compiled on it. Regards, Sammy On Wed, Nov 16, 2011 at 2:23 PM, Faraj Khasib fkha...@iconnecths.comwrote: Hi all, I tried making a video SIP call using Asterisk But it didnt workonly voice call works? Regards Faraj Khasib -- _ -- 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
[asterisk-users] Server-to-server BLF
Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Thanks! Regards, Ronald -- _ -- 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] Does Asterisk Support SIP Video Call ?
Le 16/11/2011 10:23, Faraj Khasib a écrit : Hi all, I tried making a video SIP call using Asterisk But it didnt workonly voice call works? Hi Faraj, Asterisk support H261, H263, H263+ and H264. Video calls are working since at least 1.4 version. You have to activate it by setting videosupport=yes in sip.conf -- Daniel -- _ -- 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
[asterisk-users] Skype For Asterisk (SFA)
Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- 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] Skype For Asterisk (SFA)
Yes, Skype was a good thing. R.I.P On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote: Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- 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] Does Asterisk Support SIP Video Call ?
Now I did, thank you for ur help and it works :D From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI [ad...@tootai.net] Sent: Wednesday, November 16, 2011 5:49 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk Support SIP Video Call ? Le 16/11/2011 10:23, Faraj Khasib a écrit : Hi all, I tried making a video SIP call using Asterisk But it didnt workonly voice call works? Hi Faraj, Asterisk support H261, H263, H263+ and H264. Video calls are working since at least 1.4 version. You have to activate it by setting videosupport=yes in sip.conf -- Daniel -- _ -- 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] Skype For Asterisk (SFA)
I can tell you that siptosis is workable but the support has been dropped recently as well. It is a great program and especially the paid version with trunk builder i.e. you can have multiple skype instances On Wed, Nov 16, 2011 at 8:01 PM, Abdul Basit basit.e...@gmail.com wrote: Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- 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
[asterisk-users] Limit monthly calls by context
Hello group, I have this situation: I have several contexts with a few extensions each one. I need to give every context a limited quantity of minutes they can use. All the extensions in the context will share the same bag of minutes. Meaning ext 101 use 1900 mins, ext 102 60 mins and ext 40 mins. The limit must be monthly. I guess some billing solution can do the trick, but I think it's too much for that little. I don't need any other feature. I was thinking something like checking the CDR before make the call, I know it may permit some extra minutes to be used, but it really doesn't need to be that exact. A couple of extra minutes won't hurt. Ideas, suggestions ? Hans Goossen Investigación Desarrollo Planet S.A. http://www.pla.net.py -- _ -- 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] Limit monthly calls by context
On Wed, Nov 16, 2011 at 10:46 AM, Hans Goossen goos...@planet.com.py wrote: Hello group, I have this situation: I have several contexts with a few extensions each one. I need to give every context a limited quantity of minutes they can use. All the extensions in the context will share the same bag of minutes. Meaning ext 101 use 1900 mins, ext 102 60 mins and ext 40 mins. The limit must be monthly. I guess some billing solution can do the trick, but I think it's too much for that little. I don't need any other feature. I was thinking something like checking the CDR before make the call, I know it may permit some extra minutes to be used, but it really doesn't need to be that exact. A couple of extra minutes won't hurt. Ideas, suggestions ? Hans Goossen Investigación Desarrollo Planet S.A. http://www.pla.net.py You can use the DB[1] to add a table with user and seconds. Then use the start and end seconds to do this. I know there are many ways of doing this with AGI, Manager, Realtime, etc... [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DB -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Limit monthly calls by context
Hello Hans, On 11-16-2011 14:46, Hans Goossen wrote: I guess some billing solution can do the trick, but I think it's too much for that little. I don't need any other feature. i would create a macro which calls an agi. The agi searches the CDR table (mine is in sql) and calculates if the call can go through. Then i'd call this macro from every extension in the dial plan just before the dial cmd. I was thinking something like checking the CDR before make the call, I know it may permit some extra minutes to be used, but it really doesn't need to be that exact. A couple of extra minutes won't hurt. It depends on the number of simultaneous calls from within the same context. The agi can return a number of seconds (calculated from sql) which the dial cmd can use as an absolute limit and after that amount of seconds it can hang up the call (see S or L flags). regards adam -- _ -- 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] Becoming a CLEC
But what is the correct physical setup of a CLEC. Do you get rack space at a carrier hotel and equipment in there? Do you get rack space at the local ILEC CO?; which is Verizon here. What are the types of voice platforms used by CLECs? Thanks, --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Tuesday, November 15, 2011 12:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Becoming a CLEC There are clever ways to be a CLEC, and keen reasons for becoming so. But cheaper stuff ain't one of them. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 10:02 PM, Nick Khamis sym...@gmail.com wrote: Yeah! That is what I was thinking... Bringing Voice and Video under one umbrella, things like that... I actually come from a speech recognition and natural language processing background. Trying to build the voice network, and seeing how I can bring it all together. P.S. I started by getting acquainted with the proxies of course ;) Nick On Mon, Nov 14, 2011 at 9:42 PM, Alex Balashov abalas...@evaristesys.com wrote: Only through new, innovative applications. They will always deliver transport and dialtone cheaper than you. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 9:15 PM, Nick Khamis sym...@gmail.com wrote: Hahah! Yeah it does doesn't it? What do we do? How do we stay a float, It almost seems like the ILECs will drop their rates to a penny once the people in this, and Kamailio lists ;) actually put a dent in their underline. Nick On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Skype For Asterisk (SFA)
On Wednesday 16 November 2011, Abdul Basit wrote: Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. You would be better off persuading Skype users to transition to something else. Skype is the absolute antithesis of the whole point of telephony, which is to connect people together. This includes, implicitly, the ability for subscribers on one telecommunications provider's network to call subscribers on another network. Imagine if, say, Vodafone subscribers were unable to call up BT subscribers? Well, this is *exactly* what Skype are trying to create by keeping their protocols proprietary. Of course there remains a small but finite probability that Skype will be successfully reverse-engineered, the Source Code leaked, or Skype's owners forced to publish its communications protocols before the 2013 deadline. But it would be extreme folly to bet the family farm on this happening. It's time to start seriously evaluating Asterisk-compatible alternatives to Skype. -- AJS Answers come *after* questions. -- _ -- 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] Becoming a CLEC
On 11/16/2011 10:30 AM, eherr wrote: But what is the correct physical setup of a CLEC. Do you get rack space at a carrier hotel and equipment in there? Do you get rack space at the local ILEC CO?; which is Verizon here. What are the types of voice platforms used by CLECs? Just as a point of reference, the CLEC I have experience with had 2-3 racks in a colo facility and a handful of circuits coming in from the access tandem and SS7 provider. The core switch was a SUMMA4, but I forget who made the SS7 hardware (it may have been SUMMA as well). -- _ -- 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] Skype For Asterisk (SFA)
On Wed, 16 Nov 2011, A J Stiles wrote: You would be better off persuading Skype users to transition to something else. Skype is the absolute antithesis of the whole point of telephony, which is to connect people together. This includes, implicitly, the ability for subscribers on one telecommunications provider's network to call subscribers on another network. Imagine if, say, Vodafone subscribers were unable to call up BT subscribers? Well, this is *exactly* what Skype are trying to create by keeping their protocols proprietary. Of course there remains a small but finite probability that Skype will be successfully reverse-engineered, the Source Code leaked, or Skype's owners forced to publish its communications protocols before the 2013 deadline. But it would be extreme folly to bet the family farm on this happening. It's time to start seriously evaluating Asterisk-compatible alternatives to Skype. Sadly, my experience in the SOHO environment is that Skype wins. I tried to get my family to all use SIP videophones - and it worked for a couple of years - mostly. The downside was that they're mostly using crap domestic quality broadband and trying to use a videophone, or even a soft-phone on a PC just seemed too hard for them to grasp. They *all* moved to Skype recently - and I have to say I've been totally blown away at the ease of use and the quality of the calls - both sound and video. (And I'm using Linux too) The other thing - LAN to LAN calls STAY ON THE LAN! So I can Skype my wife next door and it doesn't use up any of my own broadband bandwidth wheras if I use a hosted SIP service, calls go out come back in again. Skype also seems to be able to run the lines at max. rate too - some sort of adaptive bandwidth - we get large and high resolution video calls from one end of the country to the other with the output bandwidth running at near max (800Kb sec in our case) And now I'm seeing some of my smaller business customers using Skype. For serious business calls too. It's free. They get video. It just works. No fiddling with NAT, port forwarding, never any hint of one-way audio. I really was skeptical at first, but Skype is here to stay - mostly because it just works. Even a complete computer idiot can install it and make it work. Give them a SIP phone, or SIP softphone and tell them to set it up and they'll just leave it alone as too complicated. As for interoerability - well there's Skype-Out. It works, it's set at a reasonable price level, so what more do you need? Once upon a time I would block Skype from working inside a corporate LAN and would recomend against it's use - now I'm told to explicitly allow it. Times are changing and I'm finding it harder to persuade small businesses to use SIP phones - and why should they... Gordon -- _ -- 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] Skype For Asterisk (SFA)
I would agree, unfortunately. However, I still see it as a glorified webcam chat and not a telecommunication device like a SIP/soft phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Wednesday, November 16, 2011 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Skype For Asterisk (SFA) On Wed, 16 Nov 2011, A J Stiles wrote: You would be better off persuading Skype users to transition to something else. Skype is the absolute antithesis of the whole point of telephony, which is to connect people together. This includes, implicitly, the ability for subscribers on one telecommunications provider's network to call subscribers on another network. Imagine if, say, Vodafone subscribers were unable to call up BT subscribers? Well, this is *exactly* what Skype are trying to create by keeping their protocols proprietary. Of course there remains a small but finite probability that Skype will be successfully reverse-engineered, the Source Code leaked, or Skype's owners forced to publish its communications protocols before the 2013 deadline. But it would be extreme folly to bet the family farm on this happening. It's time to start seriously evaluating Asterisk-compatible alternatives to Skype. Sadly, my experience in the SOHO environment is that Skype wins. I tried to get my family to all use SIP videophones - and it worked for a couple of years - mostly. The downside was that they're mostly using crap domestic quality broadband and trying to use a videophone, or even a soft-phone on a PC just seemed too hard for them to grasp. They *all* moved to Skype recently - and I have to say I've been totally blown away at the ease of use and the quality of the calls - both sound and video. (And I'm using Linux too) The other thing - LAN to LAN calls STAY ON THE LAN! So I can Skype my wife next door and it doesn't use up any of my own broadband bandwidth wheras if I use a hosted SIP service, calls go out come back in again. Skype also seems to be able to run the lines at max. rate too - some sort of adaptive bandwidth - we get large and high resolution video calls from one end of the country to the other with the output bandwidth running at near max (800Kb sec in our case) And now I'm seeing some of my smaller business customers using Skype. For serious business calls too. It's free. They get video. It just works. No fiddling with NAT, port forwarding, never any hint of one-way audio. I really was skeptical at first, but Skype is here to stay - mostly because it just works. Even a complete computer idiot can install it and make it work. Give them a SIP phone, or SIP softphone and tell them to set it up and they'll just leave it alone as too complicated. As for interoerability - well there's Skype-Out. It works, it's set at a reasonable price level, so what more do you need? Once upon a time I would block Skype from working inside a corporate LAN and would recomend against it's use - now I'm told to explicitly allow it. Times are changing and I'm finding it harder to persuade small businesses to use SIP phones - and why should they... Gordon -- _ -- 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] Skype For Asterisk (SFA)
On 11/16/2011 10:44 AM, Gordon Henderson wrote: The other thing - LAN to LAN calls STAY ON THE LAN! So I can Skype my wife next door and it doesn't use up any of my own broadband bandwidth wheras if I use a hosted SIP service, calls go out come back in again. Skype also seems to be able to run the lines at max. rate too - some sort of adaptive bandwidth - we get large and high resolution video calls from one end of the country to the other with the output bandwidth running at near max (800Kb sec in our case) As SIP endpoints (servers, phones, etc.) get upgraded to support the ICE NAT traversal mechanism, this will start happening for regular SIP calls as well. This *should* already happen with the Blink softphone, for example, since it fully supports ICE. Also note that you are using the term 'calls' when you really mean 'media streams'; in all of the cases you outlined, the 'call' signaling still follows the same path it did originally, but the media stream path can be shortened if the two endpoints are able to exchange media directly. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- 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] Asterisk 10.0.0-rc1 Now Available
On 11/16/2011 02:17 AM, Eyal wrote: Hey, There is a way to use the new confbridg without installing the new version of Asterisk, I'm new in using asterisk, my is version 1.6.2 of Asterisk. I would not want just to get into the installation of a new version just for a one commend which I have very great need. It's not 'one command', it's an entire application that relies on new media descriptions and other core pieces of Asterisk itself. The simple answer is no; the app_confbridge from Asterisk 10 cannot be practically backported or used with previous releases of Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- 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] Server-to-server BLF
On 11/16/2011 04:18 AM, Ronald Cepres wrote: Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Here is one way: https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+AIS There are other methods documented on the wiki as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- 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] Skype For Asterisk (SFA)
On Wed, Nov 16, 2011 at 8:56 AM, Kevin P. Fleming kpflem...@digium.comwrote: As SIP endpoints (servers, phones, etc.) get upgraded to support the ICE NAT traversal mechanism, this will start happening for regular SIP calls as well. This *should* already happen with the Blink softphone, for example, since it fully supports ICE. Hi Kevin, Just curious on when we should expect to see the manufactures get on board with the ICE NAT? Does any particular manufacture stand out in implementing ICE NAT in their endpoints currently? Also what is Digium doing to promote it? -- _ -- 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] Becoming a CLEC
On 11/16/2011 10:30 AM, eherr wrote: But what is the correct physical setup of a CLEC. There is no correct physical setup. The setups vary as much as anything else does, and are shaped mainly by the purpose of the CLEC and the range of products it provides. Do you get rack space at a carrier hotel and equipment in there? CLECs that provide a substantial range of business-class voice and data services usually have quite a bit of equipment and either end up building out their own telco-grade data center somewhere (which can be synergistic for many of them since they are also data center operators in general), or renting a cage in a carrier hotel. There are CLECs whose equipment can functionally fit into a single rack, or even less, but those are the specialised, single-track ones that mainly exist to support the back side of some VoIP product. In cases where only one or two racks are involved, a carrier hotel is indeed a common venue. Do you get rack space at the local ILEC CO?; which is Verizon here. Yes, but _only_ for the purpose of colocating equipment that is related to backhaul and CFA, i.e. to providing services out of that CO and dragging the last-mile loops to the customer out of the CO and onto your private network. A CO and the equipment allowed it is a very restrictive and regulated environment full of equipment certification criteria and obscure rules. It will seem especially restrictive if you're used to working with commodity PC hardware and open-source; virtually nothing of the sort is allowed to be colocated in a CO. Also, keep in mind that COs generally have 23 telco racks (not 19 data racks) and supply -48V DC, or, at best, 220V AC. Space in a busy metro CO is very expensive. You really don't want to think of it as a general-purpose colocation facility. That's not what it's for. What are the types of voice platforms used by CLECs? The answer to that varies a great deal depending on the services being provided. But in general, CLECs use converged softswitches that offer them the combination of 1) TDM facilities and Class 4 routing features they need, along with (obviously) SS7 support and support for more obscure protocols that become very important in CLEC land, such as H.248/MEGACO, MGCP, etc. and 2) Class 5 subscriber features and applications so they can sell business lines, hosted PBX, etc. CLECs generally are looking for all of that in one chassis, with the obvious redundancy implications as well. They want something that they can connect to the ILEC tandems while simultaneously supporting constructs as high-level as voicemail or find-me-follow-me. Common platforms in the wild: - MetaSwitch (Class 4/5) - Sonus (rather Class 4 and IP-oriented) - Lucent Compact Switch - formerly Telica (quite Class 4) - Taqua - Excel - Tekelec Broadsoft and Cisco BTS (not so much anymore) figures every heavily into this, but they're slightly different animals than the rest. That's just the formulaic stuff. The big CLECs have all sorts of custom stuff, such as Level3's famed Lucent TNT Max-based Viper network and corresponding media gateway control/signaling gateways. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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
[asterisk-users] polycom soundpint ip650 question
On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -- _ -- 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] Limit monthly calls by context
I'd say try a2billing- thats abit of an overkill for just this functionality but you'll get lot or options to play with there. On Wed, Nov 16, 2011 at 7:02 PM, ad...@3a.hu wrote: Hello Hans, On 11-16-2011 14:46, Hans Goossen wrote: I guess some billing solution can do the trick, but I think it's too much for that little. I don't need any other feature. i would create a macro which calls an agi. The agi searches the CDR table (mine is in sql) and calculates if the call can go through. Then i'd call this macro from every extension in the dial plan just before the dial cmd. I was thinking something like checking the CDR before make the call, I know it may permit some extra minutes to be used, but it really doesn't need to be that exact. A couple of extra minutes won't hurt. It depends on the number of simultaneous calls from within the same context. The agi can return a number of seconds (calculated from sql) which the dial cmd can use as an absolute limit and after that amount of seconds it can hang up the call (see S or L flags). regards adam -- __**__**_ -- 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-**usershttp://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] Skype For Asterisk (SFA)
On Wednesday 16 November 2011, Gordon Henderson wrote: On Wed, 16 Nov 2011, A J Stiles wrote: You would be better off persuading Skype users to transition to something else. Sadly, my experience in the SOHO environment is that Skype wins. [stuff deleted] And now I'm seeing some of my smaller business customers using Skype. For serious business calls too. It's free. They get video. It just works. No fiddling with NAT, port forwarding, never any hint of one-way audio. [stuff deleted] As for interoerability - well there's Skype-Out. It works, it's set at a reasonable price level, so what more do you need? I need a rock-solid guarantee that nobody can pick up the ball and go home, leaving all former users effectively stranded. A single-vendor proprietary solution is a *massive* single-point failure. Multiple, competing but mutually-compatible proprietary solutions slightly less so. If there is even just one Open Source implementation out there, then this sort of thing can never happen. Once upon a time I would block Skype from working inside a corporate LAN and would recomend against it's use - now I'm told to explicitly allow it. Nothing goes near my company's LAN without Source Code. I figure if they don't want you to see it, there must be something in it that they wouldn't expect you to like it if you saw it. The level of paranoia on Skype's part only reinforces that impression. -- AJS Answers come *after* questions. -- _ -- 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] polycom soundpint ip650 question
Core show channels verbose - if you do asterisk -rx cscv from bash From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, November 16, 2011 12:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] polycom soundpint ip650 question On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -- _ -- 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
[asterisk-users] Polycom Attended Transfer
When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E -- _ -- 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] polycom soundpint ip650 question
Thanks for the response. What you described is for the CLI. I am asking is there a way on the phone itself or is there a phone that does have this capability. Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, November 16, 2011 1:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] polycom soundpint ip650 question Core show channels verbose - if you do asterisk -rx cscv from bash From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, November 16, 2011 12:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] polycom soundpint ip650 question On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -- _ -- 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] Polycom Attended Transfer
Upgrade to 10.0 - this isn't available in any of the 1.X flavors because they had to re-invent the background stuff for it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, November 16, 2011 12:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom Attended Transfer When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E -- _ -- 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] polycom soundpint ip650 question
You might be able to use hints/buddies. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, November 16, 2011 12:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] polycom soundpint ip650 question Thanks for the response. What you described is for the CLI. I am asking is there a way on the phone itself or is there a phone that does have this capability. Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, November 16, 2011 1:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] polycom soundpint ip650 question Core show channels verbose - if you do asterisk -rx cscv from bash From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, November 16, 2011 12:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] polycom soundpint ip650 question On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -- _ -- 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] Limit monthly calls by context
You can do this with a global variable and a dedicated context. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Wednesday, November 16, 2011 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Limit monthly calls by context I'd say try a2billing- thats abit of an overkill for just this functionality but you'll get lot or options to play with there. On Wed, Nov 16, 2011 at 7:02 PM, ad...@3a.hu wrote: Hello Hans, On 11-16-2011 14:46, Hans Goossen wrote: I guess some billing solution can do the trick, but I think it's too much for that little. I don't need any other feature. i would create a macro which calls an agi. The agi searches the CDR table (mine is in sql) and calculates if the call can go through. Then i'd call this macro from every extension in the dial plan just before the dial cmd. I was thinking something like checking the CDR before make the call, I know it may permit some extra minutes to be used, but it really doesn't need to be that exact. A couple of extra minutes won't hurt. It depends on the number of simultaneous calls from within the same context. The agi can return a number of seconds (calculated from sql) which the dial cmd can use as an absolute limit and after that amount of seconds it can hang up the call (see S or L flags). regards adam -- _ -- 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] Polycom Attended Transfer
When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller’s callerID during a blind transfer? Upgrade to 10.0 – this isn’t available in any of the 1.X flavors because they had to re-invent the background stuff for it. Asterisk v1.8 added the connected line support not Asterisk v10. Richard -- _ -- 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] Polycom Attended Transfer
So there is no way to do these with programming. For instance, setting a variable in the DB and grabbing it to override the field when transferring? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett Sent: Wednesday, November 16, 2011 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Attended Transfer When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller’s callerID during a blind transfer? Upgrade to 10.0 – this isn’t available in any of the 1.X flavors because they had to re-invent the background stuff for it. Asterisk v1.8 added the connected line support not Asterisk v10. Richard -- _ -- 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] Limit monthly calls by context
On Wednesday 16 November 2011, Hans Goossen wrote: Hello group, I have this situation: I have several contexts with a few extensions each one. I need to give every context a limited quantity of minutes they can use. All the extensions in the context will share the same bag of minutes. Meaning ext 101 use 1900 mins, ext 102 60 mins and ext 40 mins. The limit must be monthly. I guess some billing solution can do the trick, but I think it's too much for that little. I don't need any other feature. I was thinking something like checking the CDR before make the call, I know it may permit some extra minutes to be used, but it really doesn't need to be that exact. A couple of extra minutes won't hurt. Ideas, suggestions ? Set a global variable or several in the beginning of the dialplan with the monthly allowance for each context. In the h extension of each time-limited context, subtract the duration of the call just made from the global. In each extension which is time-limited, replace Dial() with a call to a macro which either actually executes the Dial() if there is any time remaining (and uses that as an absolute timeout value), otherwise plays a message. If you need the remaining allowances to persist across reboots / dialplan reloads, you'll have to have an AGI write them to a file which you can #include in the [globals] section of the dialplan. A cron job can then be used to reset the allowances at the beginning of each month, by copying a pristine file over it. Note it's still possible to go over with this method, because the remaining time is only recalculated at hangup, but you said it didn't have to be exact. -- AJS Answers come *after* questions. -- _ -- 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] Polycom Attended Transfer
When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller’s callerID during a blind transfer? Upgrade to 10.0 – this isn’t available in any of the 1.X flavors because they had to re-invent the background stuff for it. Asterisk v1.8 added the connected line support not Asterisk v10. So there is no way to do these with programming. For instance, setting a variable in the DB and grabbing it to override the field when transferring? See https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information This applies to v1.8 and later. Richard -- _ -- 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] Limit monthly calls by context
If you store the Global in DB and read it back from DB, it can persist across reboots and reloads. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Wednesday, November 16, 2011 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Limit monthly calls by context On Wednesday 16 November 2011, Hans Goossen wrote: Hello group, I have this situation: I have several contexts with a few extensions each one. I need to give every context a limited quantity of minutes they can use. All the extensions in the context will share the same bag of minutes. Meaning ext 101 use 1900 mins, ext 102 60 mins and ext 40 mins. The limit must be monthly. I guess some billing solution can do the trick, but I think it's too much for that little. I don't need any other feature. I was thinking something like checking the CDR before make the call, I know it may permit some extra minutes to be used, but it really doesn't need to be that exact. A couple of extra minutes won't hurt. Ideas, suggestions ? Set a global variable or several in the beginning of the dialplan with the monthly allowance for each context. In the h extension of each time-limited context, subtract the duration of the call just made from the global. In each extension which is time-limited, replace Dial() with a call to a macro which either actually executes the Dial() if there is any time remaining (and uses that as an absolute timeout value), otherwise plays a message. If you need the remaining allowances to persist across reboots / dialplan reloads, you'll have to have an AGI write them to a file which you can #include in the [globals] section of the dialplan. A cron job can then be used to reset the allowances at the beginning of each month, by copying a pristine file over it. Note it's still possible to go over with this method, because the remaining time is only recalculated at hangup, but you said it didn't have to be exact. -- AJS Answers come *after* questions. -- _ -- 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] Polycom Attended Transfer
Unfortunately, I only have 1.4.26 installed. What's the next stable version? Should I go to 1.6, 1.8, or 10 Thanks, --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett Sent: Wednesday, November 16, 2011 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Attended Transfer When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller’s callerID during a blind transfer? Upgrade to 10.0 – this isn’t available in any of the 1.X flavors because they had to re-invent the background stuff for it. Asterisk v1.8 added the connected line support not Asterisk v10. So there is no way to do these with programming. For instance, setting a variable in the DB and grabbing it to override the field when transferring? See https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information This applies to v1.8 and later. Richard -- _ -- 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] Polycom Attended Transfer
On Wed, Nov 16, 2011 at 1:49 PM, eherr email.eherr9...@gmail.com wrote: When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller’s callerID during a blind transfer? Thanks, --E The point of an attended transfer is to announce the calling party. When you hit transfer on the Polycom you have the option to select Blind on the screen. A blind transfer will use the caller id of the incoming call, not the person making the transfer. Ryan -- _ -- 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] Polycom Attended Transfer
I understand and agree. There is one client who prefers having the attended transfer still display the original caller ID because some users still just hit transfer and hangup. The boss has a few times got caught saying What when he thought it was an internal call but really wasn't. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: Wednesday, November 16, 2011 3:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Attended Transfer On Wed, Nov 16, 2011 at 1:49 PM, eherr email.eherr9...@gmail.com wrote: When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E The point of an attended transfer is to announce the calling party. When you hit transfer on the Polycom you have the option to select Blind on the screen. A blind transfer will use the caller id of the incoming call, not the person making the transfer. Ryan -- _ -- 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] Polycom Attended Transfer
Have you tried the conferencing button? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Wednesday, November 16, 2011 2:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom Attended Transfer I understand and agree. There is one client who prefers having the attended transfer still display the original caller ID because some users still just hit transfer and hangup. The boss has a few times got caught saying What when he thought it was an internal call but really wasn't. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: Wednesday, November 16, 2011 3:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Attended Transfer On Wed, Nov 16, 2011 at 1:49 PM, eherr email.eherr9...@gmail.com wrote: When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E The point of an attended transfer is to announce the calling party. When you hit transfer on the Polycom you have the option to select Blind on the screen. A blind transfer will use the caller id of the incoming call, not the person making the transfer. Ryan -- _ -- 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] Standard UIDs, especially for asterisk?
On Tue, Nov 15, 2011 at 04:53:02PM +, Tony Mountifield wrote: In article 4ec296b9.8040...@digium.com, Jason Parker jpar...@digium.com wrote: On 11/15/2011 10:42 AM, Tony Mountifield wrote: Yes, I was hoping to use such a system user and group for asterisk, which would not conflict with any other system package I might install in the future, by virtue of being reserved for asterisk. There shouldn't be any conflict either way. (Properly written) packages don't specify a UID to use - they just get created sequentially, so the next available ID is used. If that were the case, I would expect different installations of the same distro (with varying package selections) to have different values for UIDs of specific system users. But examination of several different RH-based systems from FC1 through to CentOS 6 shows the same values being used. I would be reluctant to label all such packages as improperly written :-) RPM and DEB work differently in this case. With deb packages you just extract the package onto the system. If anything is non-root it is extracted by UID. However typically all the files are owned by root and if there's any need to change ownership, it is done by the post-install script. RPM has a different way: it keeps record of the user and group *name* of each packaged file. When the file is unpacked, it applies the UID and GIU of the user/group from the current system. Thus if a file belongs to 'mysql' and mysql's UID on the system is 56, it will be chown-ed to 56. This is why users and groups need to be created in a pre-install script: before the package is unpacked. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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
[asterisk-users] Grandstream HT503 colgado
Hola, estoy instalando un ATA HT503 de Grandstream conectado a Asterisk y todo funciona bastante bien (llamadas en entrada y salientes). El unico problema que tengo es con el colgado. Si la llamada entrante va al buzón de voz y la persona cuelga... el canal queda abierto. Será que alguien tiene o me puede indicar donde encontrar los parametros de colgado para una linea Telecom? Muchas gracias de antemano - Bakko -- _ -- 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
[asterisk-users] Use Polycom FX with Asterisk
Hi List, I have a Polycom FX video unit and I'm thinking maybe I can integrate it on our Asterisk Server to be able to do teleconference and video as well via Polycom FX. I already have oh323 configured on my Asterisk box and I just no idea on how to let them work.Any help please? Regards, Malvin -- _ -- 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
