similar to: Remote Agents

My client wants to use my service that I will host. It is an IVR application. I have the solution worked out on the server side. I will port his current POTS line phone number to a DID service where I can control it via SIP. Question relates to his current phones. Forgive me as I am new. Does he need his current phones? How will they ring if I port the number? Should I simply have him remove

I have two servers each with one TDM card from Digium and three POTS lines going into each server (each POTS line is one individual number). and I want to have all incoming calls into server B be directed to the IVR on server A and to be able to fail over to Server B in case Server A has an issue (it can be a manual failover and I actually prefer it to be a manual process). I've

I am sure this is simple, but have been struggling. I want to create a call file that dials out a particular Dahdi channel to enable call forwarding on a POTS line. I have this in extensions.conf: [custom-callfwd] exten => s,1,Answer exten => s,n,Dial(DAHDI/4-1/*717157750) exten => s,n,Verbose(${DIALSTATUS}) exten => s,n,Hangup [custom-callfwdcanc] exten => s,1,Answer exten

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>

Hello, I have a fairly standard call center. I'm running 1.4.23.1. I am trying to get a mechanism where : 1- Employee A can have the phone at his desk ring when a call comes in the queue 2- When already on a call, employee A does not hear a "beep" in his phone because another call is trying to come in It's fairly simple. I tried a few different things: in queues.conf [559]

Hi guys, I hope I get your help for this problem ... at least point me through the right direction .... I configured a queue that when the people are tired of waiting they can press 1 and I direct them to voicemail ... but I'd like to make a callback solution at this point, because during busy hours it is hard to check the voicemail ... I'd like to make my system to call back this

I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk" My question is, what is the benefit of using Lua? I recently

A lot of people are going for the "VOIP only" approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com

Hi, I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. Regards Smir

On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote: > For a very basic setup it would work, but I would suggest POE at a > minimum, and vlan support if possible. > > Gigabit uplinks, 10/100 for the poe ports > > http://www.amazon.com/NETGEAR-ProSAFE-M4100-D10-POE-Ethernet-Managed/dp/ B00AUEYX0Y/ref=sr_1_3?ie=UTF8&qid=1424462577&sr=8-3&keywords=netgear+poe

Hi All, I have installed astriesk 6 and am able to make calls using sip x-lite. Its working as I expected. Now I want to make call from sipx-lite to PSTN using asterisk. can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.). 2) what is that I need to do after buying the card to make it talk to the real world PSTN network?

I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance,

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call

I have a client that wants to put a phone on their web page for customers to call them via their Asterisk server. ) A keypad is needed to enter credit card details. ) "Speed dial" buttons like "Tech Support," "Sales," etc. are a requirement. Actually, passing the SIP address in the HTTP link would work with a bit of arm twisting. ) Free is preferred, but not a

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

After i rebuilt my server i did default install of Asterisk using the steps off freepbx site. i used these steps before without any issues. this time i have to start Asterisk manually every time the server reboots. if i start it by using ./start_asterisk script in the freepbx directory i get this from grep root 3840 0.0 0.0 4480 544 pts/1 S 12:13 0:00 /bin/sh

I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the