[asterisk-users] any video applications available
We found this URL: http://sourceforge.net/projects/asteriskvideo/ But these applications seem too old for Asterisk 11. Are there any video applications for Asterisk 11? We need these applications to implement IVVR. Or any other solution is to be appreciated. Thanks in advance. -- _ -- 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] any video applications available
hI Check WebRTC http://www.youtube.com/watch?v=E8C8ouiXHHk 2013/2/28 Jimmy Chang(Gmail) chang33...@gmail.com We found this URL: http://sourceforge.net/**projects/asteriskvideo/http://sourceforge.net/projects/asteriskvideo/ But these applications seem too old for Asterisk 11. Are there any video applications for Asterisk 11? We need these applications to implement IVVR. Or any other solution is to be appreciated. Thanks in advance. -- __**__**_ -- 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 -- *Alan Sánchez Torres* *Telefono:* 940484090 *Email: *asanc...@vtelcom.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
Re: [asterisk-users] Point a Digium phone to a configuration URL using mDNS without DPMA or DHCP option 66
On Wednesday 27 February 2013, Alex Villacís Lasso wrote: I have the following scenario. A small network has DHCP but does not publish option 66. An Asterisk server is on the network, but the Asterisk version does not support DPMA and it is hard to switch the version. However, there is a possibility to have a web server and an mDNS (Avahi) server. I have been reading about provisioning Digium phones without DPMA, and it mentions that option 66 can specify the URL prefix to use for the XML configuration. Is there a way to specify the same through Avahi mDNS? Something along the lines of this (of course this does not work, but gives an idea of what I am looking for): DPMA is proprietary and toxic, so you are quite right not to use it. If you add a line like options tftp-server-name 10.0.0.145 to your DHCP server's /etc/dhcp/dhcpd.conf , then this will give you the option 66 you need. And note that even although it's advertised as a TFTP server, the Digium phones actually put out a HTTP request aot a TFTP request (that took some head-scratching to figure out). So you will need to have Apache running on 10.0.0.145; and a bunch of configuration files named like 000fd300.cfg where the digits of the filename are the MAC address of the phone, in the web server's default document root (/var/www/ on Debian systems). -- 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] Point a Digium phone to a configuration URL using mDNS without DPMA or DHCP option 66
El 28/02/13 08:40, A J Stiles escribió: On Wednesday 27 February 2013, Alex Villacís Lasso wrote: I have the following scenario. A small network has DHCP but does not publish option 66. An Asterisk server is on the network, but the Asterisk version does not support DPMA and it is hard to switch the version. However, there is a possibility to have a web server and an mDNS (Avahi) server. I have been reading about provisioning Digium phones without DPMA, and it mentions that option 66 can specify the URL prefix to use for the XML configuration. Is there a way to specify the same through Avahi mDNS? Something along the lines of this (of course this does not work, but gives an idea of what I am looking for): DPMA is proprietary and toxic, so you are quite right not to use it. If you add a line like options tftp-server-name 10.0.0.145 to your DHCP server's /etc/dhcp/dhcpd.conf , then this will give you the option 66 you need. And note that even although it's advertised as a TFTP server, the Digium phones actually put out a HTTP request aot a TFTP request (that took some head-scratching to figure out). So you will need to have Apache running on 10.0.0.145; and a bunch of configuration files named like 000fd300.cfg where the digits of the filename are the MAC address of the phone, in the web server's default document root (/var/www/ on Debian systems). I forgot to point out that in my scenario, I may not have any administrative control over the DHCP server. Otherwise, the way you indicate would be obvious. Anyway, I received an email saying that mDNS cannot support pointing a Digium phone to anything other than a DPMA-enabled 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
[asterisk-users] Consolidated Signup for Community Services
We felt that it would be good to let you know about some minor changes happening with our community services. For quite some time, we've had a consolidated authentication server for most of our community services. This means that you use the same username and password for issues.asterisk.org, wiki.asterisk.org and code.asterisk.org. ReviewBoard (reviewboard.asterisk.org) still uses its own internal authentecation, but we plan to migrate it some day. To make this more obvious, and to streamline account creation, we now have a single place for creating asterisk.org community accounts: signup.asterisk.org. Existing accounts will be unaffected; we've only changed how you sign up for a new account. If you have any issues with the new signup service, please contact us at asteriskt...@digium.com. Thank you for your support! -- Digium's Asterisk Development Team -- _ -- 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] Getting compilation error while installing Dhadi
Thanks Steve and Russ. It worked. From: Steve Edwards asterisk@sedwards.com Subject: Re: [asterisk-users] Getting compilation error while installing Dhadi To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: alpine.DEB.2.02.1302271337180.3668@ws Content-Type: text/plain; charset=iso-8859-1; Format=flowed On Wed, 27 Feb 2013, Ahmed Munir wrote: I'm getting compilation error as trying to install latest version of dahdi /usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xdefs.h:152: error: conflicting types for ?bool? include/linux/types.h:36: error: previous declaration of ?bool? was here Don't let a little thing like a compilation error stop you :) Just comment out line 152 in xdefs.h There may be a 'proper' way to do this, but this should work. I had the same issue compiling zaptel-1.2.27 on CentOS 5.9 yesterday. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- From: Russ Meyerriecks rmeyerrie...@digium.com Subject: Re: [asterisk-users] Getting compilation error while installing Dhadi To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 20130227215108.GA17504@blackmagic Content-Type: text/plain; charset=iso-8859-1 error: conflicting types for ?bool? include/linux/types.h:36: error: previous declaration of ?bool? was here This issue is resolved by the latest dahdi-linux release 2.6.2-rc1. You can download a tarball of the release here: http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz Or you can check out the v2.6.2-rc1 tag from git: git clone git.asterisk.org/dahdi/linux dahdi-linux cd dahdi-linux git checkout v2.6.2-rc1 -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- Regards, Ahmed Munir Chohan -- _ -- 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] Dynamic Agents in a queue
Hi, We have a queue running with dynamic agents in asterisk 1.8.12.0 and FreePBX 2.10. We are using the linear ring style. Calls are going to the agents in the order in which they log in. Is there a way to send calls to an agents in a specific listed order and not in the order that they log in? That is (assuming agent logged in). Agent1 Agent3 Agent5. So calls would always go to Agent1 if he's logged in, than down to 3, and finally to agent 5? Thanks David P.S. This seems to work fine with static agents, just not dynamic agents. -- _ -- 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] Dynamic Agents in a queue
From: David Wessell da...@ringfree.biz To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 02/28/2013 04:34 PM Subject:[asterisk-users] Dynamic Agents in a queue Sent by:asterisk-users-boun...@lists.digium.com Hi, We have a queue running with dynamic agents in asterisk 1.8.12.0 and FreePBX 2.10. We are using the linear ring style. Calls are going to the agents in the order in which they log in. Is there a way to send calls to an agents in a specific listed order and not in the order that they log in? That is (assuming agent logged in). Agent1 Agent3 Agent5. So calls would always go to Agent1 if he's logged in, than down to 3, and finally to agent 5? Thanks David P.S. This seems to work fine with static agents, just not dynamic agents. I can interpret your question in one of two ways, so if my explanation below is not what you want, please forgive me. I think your best bet is in your code where you add the queue member you want to assign a penalty to that queue member. For instance, Using your example above, suppose you had Agent 1 through Agents 5. Assign queue penalties of 10, 20, 30, 40, and 50 to them, respectively. Now, using your example of 1, 3, and 5 logged in, as long as he is available, agent 1 will get a call. If he is not, it goes to agent 3. If 1 and 3 are not available, it goes to 5. However, this does mean that Agent 1 will take more calls as anytime he is available it will ring him without attempting the other agents. We use this as some agents are primary on one queue, but secondary or tertiary on other queues. That way if the primary people are all busy, they will fall through to their backups, but if they are available it will always prefer them. We store the queue levels in the asterisk database and when the agents log in, it looks them up in the database and applies the appropriate penalty to the AddQueueMember command. Hope this helps. -- _ -- 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] Transcoding issues with siren14
Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and slin16 to slin32. So why is slin used as the intermediate instead of slin16? -- _ -- 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
