similar to: Thoughts on CPE server...

different stand alone linux server which act as my routers. Here is a picture showing the output from the CISCO switch going to the two linux servers: http://www.grmtech.com/blog/wp-content/uploads/2009/02/cisco2950-24ports-farleft-two-output-300x89.jpg My questions are: 1. The black wire coming into the Mc Manstel box is that a fibre optic cable ? 2. What is the Mc Manstel box doing ? 3. What

---------- 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 I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but

Hello all, I'm looking for some help to try to understand why my CPE doesn't work good with Asterisk in MGCP. Here is what I want to do : - Register a TECOM AH4021 on Asterisk in MGCP with the following profile in mgcp.Conf : [general] port = 2727 bindaddr = 10.95.20.1 disallow=all allow=g729 allow=alaw 020202020202] context=mgcp host=dynamic canreinvite=no dtmfmode=rfc2833 nat=yes

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

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

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

Hey guys, Looking for some suggestions here on hardware to use. We have several business customers wanting to start using our VoIP service. We will be replacing their 10 year old Panisonic system with Cisco 7940 phones and a T1. Problem is, they have to keep 1 POTS line for directory listing and 911 emergency failover. I would like suggestions on how to accomplish this. Our asterisk cluster

Samba-List A reminder that we are having John H. Terpstra present some seminars on Monday, September 19th in Chicago. We're also offering a 4-day Samba/LDAP workshop the 20-23 of September instructed by Mr. Terpstra. Details and links below. RSVPs for the seminars and registration requests for the workshop should be guided to alexm@primastasys.com or via phone in the US to 650-279-1909.

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,

Hi! Im having some troubles trying to configure * as a bridge between a telco and a pbx with R2, the scenario is this: ------------ E1/R2 --------- E1/R2 ------------ | Telco |-------------| * |-------------| PBX | | (Telmex) | --------- | | ------------ ------------ I can receive calls from the telco

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

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''m looking at an example where a stub seems to work sometimes, and sometimes appears to become "unstubbed". I haven''t boiled it down to a minimal example, but it goes something like this: ---- the model: class Premise << ActiveRecord::Base def lookup_stuff_on_the_web $stderr.puts("entering lookup_stuff_on_the_web with #{self}") ... end end

A thought occurred to me to on how to further quantify the impact of "glare" on a properly dimensioned trunk group and debunk the ground start glare concern. A cursory traffic analysis clarifies: 1. Assume you have a two-way trunk group, dimensioned for average busy hour, average busy season for P01 grade of service, Erlang B or Poison distribution. 2. Further assume that the

Dear all I got one more error my asterisk E1 card connected with avaya E1 card [avaya]-------E1-----[asterisk] i got this 2 error what is start asteris on consol mode asterisk -vvvvc [Jul 27 09:51:29] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:30] WARNING[737] chan_zap.c: PRI Error on

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

Hi Gregory, GM> Flac decode is VERY fast: GM> http://web.inter.nl.net/users/hvdh/lossless/lossless.htm Thank you for the link. Although I switched to flac for other reasons, these data appear to support my switch, given the hardware limitations I have. As I mentioned, I use a 400 mhz laptop in my office, and at home a qnap ts-101 running slimserver software. While quite usable, neither