similar to: A unique 'click to call' project - Could usesomeadvice

Colin, Thanks for your assistance. Reading over your advice I seem to still be a bit confused. My agents are not on the Asterisk server; it appears in your advice that my the call will travel this path: WWW interface --> agent enters their DID, platform to use, and termination DID --> AST calls agent --> Agent calls termination DID If my agents are not on the Asterisk server

Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a

In the example I posted previous, there is an obvious gaping security hole, it would be trivial for someone to read the querystring and exploit it to make free phone calls, spoof caller ID (if you allow the CallerID to be set with a QueryString value), etc. You want to make damn sure that the URL is not publicly accessible or somehow obsfucate the querystring, or use POST. In my case, I

Why don't you use Local and router functionality to find a route to PSTN based agents? W _____ From: Aloi, Christopher [mailto:caloi@usadatanet.com] Sent: Friday, February 17, 2006 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] A unique 'click to call' project - Could use someadvice Hello List, I work for an IP

Hi all, I'm getting contradictory results from bptest and ncvTest on a model calculated by GLS as: olslm = lm(log(rr)~log(aloi)*reg*inv, data) varlm = lm(I(residuals(olslm)^2)~log(aloi)*reg*inv, data) glslm = lm(log(rr)~log(aloi)*reg*inv, data, weights=1/fitted(varlm)) Testing both olslm and glslm with both ncvTest and bptest gives: > ncvTest(olslm) Non-constant Variance Score Test

Hello List - I've got a question regarding POS terminal transactions (credit card machines, ATM, etc...). Currently we setup customers in the following manner: Customer Location --> Data T1 --> DataCenter -> PSTN Termination We are currently using Mediatrix gear for fax transmissions from the customer location, but they don't seem to handle POS modem sales very well. Does

Hello List, Question: Has anyone been able to create multiple queue_log files in /var/log/asterisk for multiple queues? We are designing a multi-tenant system and separating the log files would be useful, instead of dropping all queue actions into one file. Is it possible this is a user configurable option I am missing? Cheers, -- ------ Christopher T Aloi ------

Hello List - I've been slowing growing my extensions.conf file and have been wondering how everyone manages their systems. I currently have my main extensions.conf where I reference my sub extensions (for tenants or customers) files using the include statements and define my global variables. Today while watching the asterisk console I noticed a call from a voicemail user bounced into

I run a SIP-only shop with a 23 channel PRI and single T100P. Here are my configs: /etc/zaptel.conf: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] context=default switchtype=dms100 signalling=pri_cpe pridialplan=unknown rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=no hidecallerid=no callwaiting=no

Hello List! I am hoping someone may be able to assist with the following feature I am looking to implement: I would like to use a visual park function if possible, here's how I see it working.... -> Call comes into caller A -> Caller A places the call in orbit (or park) by dialing 700 -> A message is then sent to illuminate a line on each of the phones in this office indicating a

Hi, I saw a new assert and core dump today in 1.0alpha1. Setup is Solaris 9, dovecot built with gcc 4.0.1, running as imap server for mbox format. The syslog said: IMAP(user): file message-body-search.c: line 393 (message_body_search_ctx): assertion failed: (input->v_offset <= part->physical_pos) A gdb analysis of the core dump is attached. BTW, I save core dumps in case you need

I don't think "very carefully" was the answer that the original poster was looking for. I would suggest one of the multi-tenant GUIs out there, or map out your call flow in a diagram, convert your diagram into the dialplan, then test, test, test. Make sure to test everything, including invalid extension, #, timeout, etc... Another strategy is to use a GUI on a dev box and then

---------- Forwarded message ---------- From: Christopher Aloi <chris.aloi@gmail.com> Date: Jun 18, 2006 9:52 PM Subject: Re: FW: [Asterisk-Users] Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts? To: Alexander Lopez <Alex.Lopez@opsys.com> Alexander, Thanks for your reply, may I ask a few questions? - Does the

Hi, We are currently trying to setup Asterisk with iBasis. One question/problem we have is that Ibasis has told us to send the INVITEs to one IP address and all media to a different IP address. How can we do that in Asterisk? Thanks

Ok, not exactly an Asterisk problem, but... I picked up some SNOM 200 phones because SNOM's have been recommended for use with Asterisk and they have line buttons that can subscribe to presence. However, they don't appear to power up when connected to my Negear FS108P, which is an 802.3af Power-over-ethernet capable hub. I am pretty sure these are the SNOM 200b, in that the ethernet

Hey All: Question; when using a virtual IP on an Asterisk server, I am having trouble getting sip user to register to the ViP. They are able to register with the true IP, just not the virtual. It seems Asterisk is rejecting the SIP invite, register, etc (like it's not destined for this server) I've added all the IP's to the domain listing in sip.conf and in the Asterisk console a

Hello, Long time subscriber/reader of this list - thank you for all the great ideas. Scenario: We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality: - Agents having the ability to

First of all sorry for the little offtopic post :-) For one of our customers i need to make a vpn between the Netherlands and Hungary. Over this vpn two * machines are gonna talk IAX and employees in Hungary are gonna use the Exchange server located in the Netherlands. So far no problem... The problem: I'm from in the Netherlands and don't understand the .hu websites :-( First question:

Hello List, So I have a few MiTel 5224 IP phones running in SIP mode. Per the phones documentation they honor SIP distinctive ring tones. I am able to send the correct ALERT_INFO message in an invite from Asterisk to the phone, but I don't know what ring tone to call. From the reading I've done the syntax is: [3155791234] exten => s,1,Set(_ALERT_INFO=Ring 8) exten =>

Hey List - I've been looking into the various options for small form factor customer premise gear, and am wondering what your using and what your reccomendations are. I'd like to drop a unit at the customer premise to handle their internal routing and trunk their outgoing calls back into my datacenter. The CPE unit would be useful if their WAN link drops as internal calls will stay up