Archives : January-2017
I need to make attended transfer work via an AMI request.Based on data from a Cisco trace from another system which successfully does an attended transfer, the Refer-To header requires the followi..
For years, I used to configure SIP phone VLAN membership through a DHCPserver.Here are the details:- I dedicate a LAN port on a switch to voice VLAN- somewhere else, I configure a DHCP server to serve LAN addresses within voice VLAN- any other swi..
I want to develop an Asterisk module, but I have several doubts when building Asterisk plus my module – which is inside src/apps directory.-How to compile and link additional .c programs, like chan_sip does?-How to compile and link libpq inside an Aster..
Dep.FinanceiroPrezado cliente, Nosso sistema não foi identificado o pagamento até o momento, por favor verificar. Protocolo de Atendimento:Clique aqui e visualize a segunda via. Dep.Financeiron_2364741225290646450105851366867094n_236474122529064645010585..
We are generating AOC messages via the AMI and trying to deliver them to various brands of SIP phone.When snom_aoc_enabled = yes in sip.conf then a message is set in the Snom format correctly.Were not having much luck sending any other type of AOC mess..
I googled about a bit without success, so…Is there a version matrix available?Something that would say: for kernel version w, you can run up to version x of Asterisk, DAHDI version y, and libpri version z?For example, I have a bunch of remote ho..
One common mean to remotely configure a phone is to send it some XML data using HTTP. Of course, this XML data is vendor specific but thanks to Asterisk multiple tools, it is quite easy to customize your dialplan to both build and send this speci..
Dear, I have Asterisk 1.8 (installed with Elastix 2.4) and I want to integrate a Vtiger 6.5 server.In my PBX I have Asterisk 1.8, Java 1.4 and I have not Java Jetty.What are the requirements in the Asterisk server in order to install the VtigerAsteriskConnec..
This was asked many years ago but I thought I would check to see if things have changed.Is it possible to take over a call in progress – using a replacement Asterisk server? In other words, if 2 user agents are connected through an Asterisk PBX, ..
If I understand correctly,Asterisk 14 introduced support for some new SRTP ciphers (including some 256 bit ones), previously only two 128 bit ciphers were supported. Using Asterisk 14, I was able to make a call from a softphone (Groundwire) with a ..