Anyone have any recent experience with openfire and asterisk integration, perhaps with the spark IM client?About to dive into this and would appreciate any advice on gotcha..
Author : Jeff LaCoursiere
Home » Articles posted by Jeff LaCoursiere
Apologies if this is considered off-topic; I suspect the information might benefit a portion of the list.Can anyone point me in a direction to start implementation of E-911 services?Is this just something my upstream should supply, or can I connect..
I have two upstream providers we use for US termination.The dialplan sends calls out the primary and if that fails for specific reasons, it sends the same call out the secondary. This has worked well for us when we are lazy about keeping balances ..
I have played around with iaxmodem and hylafax and have a few working installations where PRIs are involved.I have a new customer that will be sending inbound fax calls via a new (SIP) DID provider we are working with (yup, same one from the last mess..
We are trying to work with a new DID provider and I find myself confused.Their standard integration is to send the call with no authentication.I am expected to whitelist all their possible gateways, and accept their calls I guess with no peer definitio..
I am dealing with a telco that has recently upgraded from a DMS100 switch to a Metaswitch, and our PRI no longer passes foreign caller ID information, i.e. if I place an outbound call with specific caller ID information not associated with the PRI,..
Has anyone created any docker images I might be able to use on EC2 for load testing an asterisk platform?I started an instance this morning and was about to load sipp and other tools, and then thought surely someone must have done this already.Id l..
Here is a funny story.We mostly do hotels in the Caribbean, and one of our first clients (going on ten years now) used the sample weather.agi that used to be shipped with… asterisk@home? Trixbox?Cant even recall where we originally got it from.T..
Our custom application sets some SIP headers that we want passed to the called party via asterisk in a simple proxy setup.It works fine for voice calls, but we also use SIP to send outofcall messages.I notice I cant use SIP_HEADER() to get those cus..
I have a need to pass through SIP headers that start with a particular prefix, without knowing beforehand what the full name of the header actually is.For example I need to test for any headers on an inbound channel that start with FOO_ and then ..