search for: xivo

Dear asterisk users, We would like to announce 2 VoIP activites organised by XiVO. For those who've never heard about the project: XiVO is a free/open source telephone system under the GPLv3 that has been based on asterisk since 2005. More info on our website: http://xivo.io [3]. We have an IRC channel, #xivo on freenode and a forum [0]. Feedback and contributions are always...

...s updated skills based routing patch for asterisk queue >> please test if you have time >> >> https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 >> >> > > You can find the latest version we maintain here : > https://github.com/xivo-pbx/asterisk/blob/master/debian/patches/xivo_skill_queues > (asterisk 13.5) > > We originally wrote this patch for xivo and it's included by default. > > Sylvain > that's great! do you have plan merge it to the asterisk master? -- --------------------------------------...

Hello, I did some tests because i'm interesting to transfer a non stasis bridge to a stasis bridge and i found a strange situation. A call B B answer You have a bridge On my asterisk CLI: xivo*CLI> bridge show b1d8fb21-ec6d-469a-9dde-bb6bfd5618cc Id: b1d8fb21-ec6d-469a-9dde-bb6bfd5618cc Type: basic Technology: simple_bridge Num-Channels: 2 Channel: SIP/tcu9tz-00000032 Channel: SIP/c4i2zd-00000031 Now with swagger UI on bridges section. POST /bridges/b1d8fb21-ec6d-469a-9dde-bb6bfd561...

hi, there is updated skills based routing patch for asterisk queue please test if you have time https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 -- --------------------------------------- Marek Cervenka =======================================

Hi, Le ven. 10 juil. 2020 à 16:56, Sylvain Boily <sylvain at wazo.io> a écrit : > > It probably can help you: > > https://github.com/wazo-platform/wazo-provd-plugins/blob/master/plugins/xivo-jitsi/1/templates/base.tpl > > Sylvain > > Yes, provided example was exactly what I was after ! Thank you very much ! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200721/d1da4ebd/attach...

hi all, Can someone help me with a kind of howto build call center around asterisk with all the necessary features like CTI, call recordings, call spying, real time monitoring etc? I will be glad if it is an open source code. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, 1. I'm looking for an (anomized) example of a Jitsi Desktop provisioning file compliant with Asterisk ? Jitsi Doc mentions it should adhere to Java properties file syntax (see [1]) but a working example would help. If this example file included the following settings, it would be perfect: - sip username, secret, authname, ... - codec G711, RFC2833 - BLF - local or remote phonebook -

hello all, I am looking for an implementation of a 10 man call center. low cost license or GPL will be preferred. I will be glad for your help. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160622/8e2e5ab8/attachment.html>

...n the AGI side ? The asterisk version we've been using for a long time is 1.4 and my current attempts are done on 1.8 branch. Thanks, -- Corentin LE GALL Proformatique (Groupe Avencall) - 10bis rue Lucien Voilin - F-92800 Puteaux Tel (+33/0)1.41.38.99.60 - Fax (+33/0)1.41.38.99.70 http://wiki.xivo.fr

Good afteroon all, First of all: thanks for everybody who is willing to think this through with me: I'm having some issues regarding call quality between some calls. Let me try to explain the situation first We have a Asterisk 11.16 server based on the Xivo distribution. There are 2 servers running in cluster (Active Passive), both virtual with the following config: Quadcore CPU 8 GB ram About 50Gb of diskspace which is used for about 15% (Let's call this Asterisk cluster 001 for clarity) The Asterisk server has a trunk to a cisco call manager wh...

...tps://desktop.jitsi.org/provisioning.html > > 2. Have ever tried passing some setting to installer (see [2]). Comments ? > > [2] https://desktop.jitsi.org/Documentation/MSIParameters.html It probably can help you: https://github.com/wazo-platform/wazo-provd-plugins/blob/master/plugins/xivo-jitsi/1/templates/base.tpl Sylvain

...2020-07-21 3:57 a.m., Olivier wrote: > Hi, > Le ven. 10 juil. 2020 à 16:56, Sylvain Boily <sylvain at wazo.io > <mailto:sylvain at wazo.io>> a écrit : > > > It probably can help you: > https://github.com/wazo-platform/wazo-provd-plugins/blob/master/plugins/xivo-jitsi/1/templates/base.tpl > > Sylvain > > Yes, provided example was exactly what I was after ! > Thank you very much ! You're welcome. Sylvain -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-user...

...best way it is done today. I think most people use FreePBX for this kind of task, but that may be historical inertia. You may want to take a look at nerdvittles.com. I think their Incredible PBX package uses FreePBX or some new GUI whose name escapes me. They also have a couple of articles on XiVO which they seem to be excited about. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

hi all, can anyone give me a guide on any asterisk admin solution / interface for config management, and monitoring? No database use is intended and I prefer open source. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

Hi every all, since a few weeks I came back to asterisk and tried to install version 1.6. The installation went fine so I decided to buy new dids on Voxbone. I have added the sip peers of Voxbone Belgium1 like this in the sip.conf [81.201.82.39] host=dynamic type=friend insecure=very context=your_context canreinvite=no qualify=no deny=0.0.0.0/0.0.0.0 permit=81.201.82.39/255.255.255.255 but

I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland < kvandenouweland

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but