similar to: A unique 'click to call' project - Could use some advice

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, I'm not sure what you mean, could you elaborate? Thanks, -- -- -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: caloi@usadatanet.com <mailto:caloi@usadatanet.com> -- -- -- _____ From: Wojciech Tryc [mailto:Wojciech.Tryc@pikatech.com] Sent: Friday,

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

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'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

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, 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

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:

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

Hi ups-devel, As I looked throw www.networkupstools.org site, haven't found any pointers where to send patches, so I hope this is the right place. (Please let me know weather this is the right place.) The patch contains a minor change in the logic of the driver, our Effekta MHD 3000 UPS continuously writes status data to the serial port, so one has to send a CR character and empty the

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

Sorry for being a little off topic, but I'mconsidering a few new phones for my Asterisk installation. I have a mix of Polycom 500/600s and an Aastra 480i CT. I'm considering adding a couple of Aastra 57i or 57i CT. Does anyone here have experience with the 480i CT and the newer 57i CT? I'm curious as to the real differences. Thanks, Michael Graves Sr Product Specialist Pixel Power

Hey All - I've got an interesting problem, here is what I'm trying to accomplish: Six agents, two queues, three skill levels Queue A (queue B is the same) - Level 1 -- Agent 1 -- Agent 2 - Level 2 -- Agent 3 -- Agent 4 - Level 3 -- Agent 5 -- Agent 6 I'd like a call to come in to Queue A, ring agent 1 and agent 2 for X seconds, if agent 1 and agent 2 ignore the call, try

Hello - I currently have 10 DID's coming into one Asterisk server, I seem to be having some difficulty routing based on the DID dialed and am hoping someone on the list can assist me. Here's the relevant info: Ingress SIP trunk: IP: 123.45.45.3456 DID's XXX-XXX-XX00-XX10 sip.conf: [general] useragent=Asterisk port=5060 context=default tos=lowdelay disallow=all allow=ulaw

Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say "press one to blah.." and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the