Re: [on-asterisk] Any feedback on Cogent Datacenters (151 + North York) and Cogent Internet Connections?
Chuck, Most any of the companies that are in 151 Front are very good (Priority Colo, Accelerated Connections, Cologix, Neutral Data, etc). I have no experience with Cogent data centres but I can tell you that their internet connections are pretty good (now). I added the now in brackets because many years ago (4+ years ago) they were awful. They had all sorts of problems and developed a bad name. They fixed those problems and have been rock solid for a number of years. As you have undoubtedly noticed, they are very price aggressive as well. Another price aggressive and reliable company to look at is Hurricane Electric. Regards, Bill Sent from my BlackBerry 10 smartphone on the TELUS network. From: Chuck Mariotti Sent: Tuesday, May 26, 2015 2:24 PM To: asterisk Mailing Subject: [on-asterisk] Any feedback on Cogent Datacenters (151 + North York) and Cogent Internet Connections? We haven’t seen any negative feedback on them and wanted to reach out to see what people have to say or alternatives. Regards, Chuck Mariotti - CTO / Co-Founder [cid:image001.gif@01D097BF.9EF21F80]http://t.signauxsept.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XZs1pNd50W65jFB45w6Gk-W2BgSpx56dBs9f1C97gs02?t=http%3a%2f%2fwww.xunity.com%2fsi=5460257179959296pi=39aa6f73-e4aa-4df6-985e-d238af1f18f4 Office (416) 469-5008 x222 Cell (416) 318-4224 Fax (416) 469-5009 [cid:image003.gif@01D097BF.9EF21F80]http://t.signauxsept.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XZs1pNd50W65jFB45w6Gk-W2BgSpx56dBs9f1C97gs02?t=http%3a%2f%2fca.linkedin.com%2fin%2fChuckMariottisi=5460257179959296pi=39aa6f73-e4aa-4df6-985e-d238af1f18f4 [cid:image004.gif@01D097BF.9EF21F80] http://t.signauxsept.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XZs1pNd50W65jFB45w6Gk-W2BgSpx56dBs9f1C97gs02?t=https%3a%2f%2fwww.facebook.com%2fchuck.mariottisi=5460257179959296pi=39aa6f73-e4aa-4df6-985e-d238af1f18f4 [cid:image005.gif@01D097BF.9EF21F80] mailto:cmario...@xunity.com [cid:image006.gif@01D097BF.9EF21F80]http://t.signauxsept.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XZs1pNd50W65jFB45w6Gk-W2BgSpx56dBs9f1C97gs02?t=https%3a%2f%2ftwitter.com%2fChuckXYZsi=5460257179959296pi=39aa6f73-e4aa-4df6-985e-d238af1f18f4 [cid:image007.gif@01D097BF.9EF21F80] skype:cmariotti?chat Xunity Inc. 13 Seymour Ave. Toronto, Ontario, M4J 3T3 CONFIDENTIAL COMMUNICATION - This message (which includes any attachments) is intended only for the designated recipient(s). It may contain confidential or proprietary information. If you are not a designated recipient, you may not review, use, copy or distribute this message. If you received this in error, please notify the sender by reply email and delete this message and all attachments, including any copies thereof. Thank you.
Re: [on-asterisk] ILEC Local Calling Area Code Matrix
we have found it to be very canadian friendly. each rate center has a link which will give you csv output to every NPA/NXX that is local to that code. with a little bit of scripting magic you can do just about anything from the data in local calling guide Sent from my BlackBerry 10 smartphone on the TELUS network. Original Message From: Reza - Voipernetics Sent: Wednesday, February 18, 2015 12:16 AM To: Bill Sandiford; Asterisk Users Group Subject: Re: [on-asterisk] ILEC Local Calling Area Code Matrix Thanks for the response Bill. I need the entire sheet / database. Basic google search pointed to that website as #1, but it does not seem to give what I am looking for. It also appears to be very Canadian unfriendly. Suggestions? Bill Sandiford wrote on 2/18/2015 12:12 AM: localcallingguide.com? Sent from my BlackBerry 10 smartphone on the TELUS network. Original Message From: Reza - Voipernetics Sent: Wednesday, February 18, 2015 12:10 AM To: Asterisk Users Group Subject: [on-asterisk] ILEC Local Calling Area Code Matrix Greetings: Does anyone know where I can get my hands on the local calling area code matrices? For example: - 905 Oakville area code is not local to 905 Pickering - 416 area code is local to 905 Mississauga but not necessarily 905 Oakville - 705 Peterborough is not local to 705 Barrie - 705 Peterborough local to 705 Bobcaygeon - 705 Bobcaygeon not local to 705 Barrie (not sure) etc. I am looking for both overlapping area codes and non-overlapping area code matrices / chart from originating rate centre to destination rate centres within a province that are local to each other. The purpose of this request is for the purpose of providing and identifying local access numbers of specific rate centres. Thanks! Reza. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Toll-free number assignements
Hi Bruce, We are a RESPOrg. There are no exchanges per se. We just request numbers from a central database when we get a request for them. As far as forwarding, we can forward to a PRI or POTS lines or SIP Trunk. Bill On 2013-02-14 5:04 PM, Bruce N brucev...@gmail.com wrote: Hi everyone, I am curious as to how Toll-Free numbers are assigned and allocated. As for as local exchange goes, CLECs or ILECs hold a whole exchange or part of it but is that also the case for Toll-Free numbers? For example if a toll-free number is not yet assigned to anyone does it reside with BOCs? And how does it work after it's assigned? Does it get forwarded to a PRI on CLEC or iLEC or directly to a local number? Any RespORG members out there? -Bruce - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Any suggestion for a solid small box flashable router?
Hi Jason, My experience has been the same although I would add the following 2 points. 1) We've never been able to get the wireless features of the routerboard products working well in a small business or residential environment. 2) stick with the 4.17 firmware. We've seen many issues with the 5.x and later firmwares, particularly with mlppp. Bill Sent from my iPad On 2013-01-29, at 6:44 PM, Jason Rose jjk...@rogers.com wrote: Hey All, When I needed MLPPP setup I tried tomato / DDWRT / one other. I found these to be not reliable enough, with so many variations for different hardware. I purchased a routerboard and I will never buy a consumer model router again, the hardware hasnt crashed in over a year (oldest unit I have is 1 year 3 months, full uptime). You have extensive authentication, firewall options, built in VPN options and everything is fully configurable (in an easier way then the opensource ones). I order these in from Montreal in batches... if you want to try a unit I can order one for you on my next order to save you shipping. If you want to play with one, I have an extra kicking around. Link to the cheap unit: http://routerboard.com/RB751G-2HnD Link to the new commercial unit: http://routerboard.com/RB2011UAS-2HnD-IN BTW these have 1w wireless radios vs the 200w ish that you get in consumer routers. J From: Bruce N brucev...@gmail.com To: asterisk@uc.org Sent: Sunday, January 20, 2013 12:09:30 PM Subject: [on-asterisk] Any suggestion for a solid small box flashable router? Hello, What is a solid ~$50 router board these days that allows flashing and has at least 16MB memory? Looking to install firmware like Tomato or any other that allow for various VPN technologies. Thanks, Bruce - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Introducing iaxproxy - a simple IAX2-SIP Proxy
HI Matt, That sounds very cool. Will certainly give it a try. Bill -Original Message- From: Matthew Gamble [mailto:mgam...@mgamble.ca] Sent: November 23, 2012 5:30 PM To: asterisk Mailing Subject: [on-asterisk] Introducing iaxproxy - a simple IAX2-SIP Proxy All, I'm very happy to be able to share with you a project I've been working on in my spare time - iaxproxy.org. While not directly related to Asterisk per se, I believe the project will be of interest to members of the Asterisk community. IAXProxy is an open source IAX2 to SIP Back-to-back Protocol Adapter (B2BPA) based originally off Asterisk. The goal of IAXProxy is to allow anyone the freedom to integrate IAX2 based end-points seamlessly into a SIP environment. Previously interconnecting IAX based devices to a SIP based network was challenging at best, requiring the network operator to run dedicated Asterisk PBX's to connect these devices. The result was that IAX2 based users were always second class citizens in a SIP environment - the SIP core was not aware of the device state of an IAX2 endpoint (registered/unregistered), etc. IAXProxy changes that by providing surrogate registration type functionality for IAX2 devices. When an IAX2 end point connects to IAXProxy the endpoint information is looked up in an internal in-memory database and assuming the IAX2 device passes authentication then a SIP Peer and SIP Registrar are created on the users behalf. When the IAX2 endpoint becomes unreachable the SIP Peer Registrar are deleted. This allows the SIP network to be fully aware of the state of IAX2 devices and features such as Call Forwarding Unreachable to be provisioned at the SIP Server level. The software is currently very alpha but it does work and allows you to make and receive calls using IAX2 devices to a SIP network. I'm currently looking for anyone interested in assisting with this project - I need help with testing, documentation, etc so if you are interested please let me know. You can download the initial release from http://www.iaxproxy.org or directly from GitHub (https://github.com/primuslabs/iaxproxy) I'm very excited about this initial release and look forward to the community feedback. Thanks! Matthew M. Gamble - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Next meeting location.
Stephan, If by downtown Toronto you mean the Primus location at 10 Bay or somewhere else close to Union Station then that is my preference. Regards, Bill -Original Message- From: Stephan Monette [mailto:monette.step...@gmail.com] Sent: Thursday, September 15, 2011 11:20 AM To: asterisk@uc.org Subject: [on-asterisk] Next meeting location. Hi everyone, I'm currently working on organizing our next TAUG meeting and looks like we will be able to have the meeting sometime during the week of October 17. My new employer (Primus Canada) will be sponsoring the conference room, refreshments and light food. I'm just waiting for some confirmation from our invited speaker and our speaker will also donate some prizes to be drawn at the end of the meeting. In order to accommodate as much people as we can, we need to know where would be the best place in Toronto to held the meeting. Can you guys send me an email and specify your preferred location?: 1- Downtown Toronto 2- Mississauga 3- Brampton 4- North York 5- North of Toronto 6- East of Toronto 7- West of Toronto Send me your answers and I will get the ball rolling. Thanks, Stephan Monette Primus Business Services. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Next meeting location.
Let me rephrase that. My preference would be for Toronto East as I live in Whitby, however I think that somewhere close to Union is probably the most practical. Bill -Original Message- From: Bill Sandiford [mailto:b...@telnetcommunications.com] Sent: Thursday, September 15, 2011 11:18 AM To: 'Stephan Monette'; asterisk@uc.org Subject: RE: [on-asterisk] Next meeting location. Stephan, If by downtown Toronto you mean the Primus location at 10 Bay or somewhere else close to Union Station then that is my preference. Regards, Bill -Original Message- From: Stephan Monette [mailto:monette.step...@gmail.com] Sent: Thursday, September 15, 2011 11:20 AM To: asterisk@uc.org Subject: [on-asterisk] Next meeting location. Hi everyone, I'm currently working on organizing our next TAUG meeting and looks like we will be able to have the meeting sometime during the week of October 17. My new employer (Primus Canada) will be sponsoring the conference room, refreshments and light food. I'm just waiting for some confirmation from our invited speaker and our speaker will also donate some prizes to be drawn at the end of the meeting. In order to accommodate as much people as we can, we need to know where would be the best place in Toronto to held the meeting. Can you guys send me an email and specify your preferred location?: 1- Downtown Toronto 2- Mississauga 3- Brampton 4- North York 5- North of Toronto 6- East of Toronto 7- West of Toronto Send me your answers and I will get the ball rolling. Thanks, Stephan Monette Primus Business Services. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] WIFI portable SIP phone
I use Bria for iPhone on my iPhone and iPad and it works great. -Original Message- From: Andre Courchesne [mailto:courc...@net-forces.com] Sent: Friday, September 02, 2011 8:15 AM To: Reza - Voipernetics Cc: asterisk Mailing Subject: Re: [on-asterisk] WIFI portable SIP phone Interesting Thanks for the feedback. Wondering how a latest generation iPod with a SIP or IAX soft phone would stickup to the Samsung S Android. Or something like a Nokia C3 with the embedded SIP phone. --- Andre Courchesne - Consultant http://www.net-forces.com MSN: courc...@net-forces.com Skype: VoipForces L'information contenue dans le présent document est la propriété de Andre Courchesne. Et est divulguée en toute confidentialité. Cette information ne doit pas être utilisée, divulguée à d'autres personnes ou reproduite sans le consentement écrit explicite de Andre Courchesne. The information contained in this document is confidential and property of Andre Courchesne. It shall not be used, disclosed to others or reproduced without the express written consent of Andre Courchesne. On 2011-09-02, at 2:37 AM, Reza - Voipernetics wrote: We have played with almost each and every wifi portable SIP phone in the market today. Including the SIP DECT phone's -- which is not really WiFi, but is wireless portable 2.4 Ghz and the 5.8 Ghz phones. NONE to our experience is worth the money spent. At a factory we merely connected industrial grade and consumer grade 5.8 Ghz DECT phones to either ATA's or a 8/12 port FXS (Audio Codes, Grandstream etc). The ONLY phone(s) WiFi which stood up to our standards in an office scenario is the Samsung S Android phone, which has a built in SIP client that runs both in WiFi and 3G mode - and the battery power on the Samsung S series with Wifi turned on ranges anywhere between 15-24 hours on standby and a good few hours talk time. Cheers! Reza. -- FOUNDER SR. TELECOM ANALYST VOIPERNETICS COMMUNICATIONS NATION WIDE DIDS, SIP TRUNKS VOIP 911. PARTIAL / FULL VIRTUAL PRI - NO CONTRACTS! HOSTED PBX TERMINATION SERVICES. TEL: 647-476-2067 Andre Courchesne wrote the following on 9/1/2011 9:55 PM: Any recommendations on WIFI portable SIP phones? Any good ones (range and battery life) ? --- Andre Courchesne - Consultant http://www.net-forces.com MSN: courc...@net-forces.com Skype: VoipForces L'information contenue dans le présent document est la propriété de Andre Courchesne. Et est divulguée en toute confidentialité. Cette information ne doit pas être utilisée, divulguée à d'autres personnes ou reproduite sans le consentement écrit explicite de Andre Courchesne. The information contained in this document is confidential and property of Andre Courchesne. It shall not be used, disclosed to others or reproduced without the express written consent of Andre Courchesne. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
[on-asterisk] Re: [biz] FreeSwitch Training Session - Are you interested?
Bruce, Have you asked Darren what his max class size is? We attended one of his courses in San Francisco last year and the max was 10. That may have been a classroom size limitation however my opinion was that any larger than 10 would have made the teacher to student ratio undesirable. Most technical training that I've ever been to has a limit if 10. You should check with Darren if you haven't already done so. FWIW, his training is excellent and I highly recommend it. Bill Sent from iPad On 2011-06-13, at 1:58 AM, Bruce N het...@hotmail.com wrote: Hi everyone, I am excited to see all the support. If we can just double the amount of people already committed Seems like there is a lot of interest for a summer VoIP party :) I am wondering if anyone has any connections to Montreal, Vancouver, Nova Scotia, Winnipeg asterisk user groups? or even any other general VoIP user groups. Committed so far: 2x Mike A. Luke D. Syed Z. Ryan M. Reza R. Jim M. Tim S. Bruce N. Have we got any more names that I might have missed? Please add your name to the list. -Bruce Date: Fri, 10 Jun 2011 21:50:08 -0400 From: mike.ash...@qualitytrack.com To: b...@taug.ca Subject: Re: [biz] FreeSwitch Training Session - Are you interested? Bruce, I'd also be very interested. I can provide some equipment, say a small 1U supermicro server if needed ( fits in backpack ). We're developing a custom freeswitch platform but definitely in attending and bringing one of my programmers. Mike On 06/10/2011 6:28 PM, Bruce N wrote: Hi Everyone, I know there has been a tutorial or two on FreeSwitch at TAUG meeting but there hasn't been any official trainings in Toronto or maybe all of Canada from the FreeSwitch team. I am very interested to explore FreeSwitch at a deeper level and I contacted Darren Schreiber (co-author of the FreeSwitch Book) and it seems they are interested to offer us their knowledge. I am writing here to see how many of the members or non-members would like to attend a whole day event or maybe even a 3 day event. I am also posting here to see if we can have any volunteers to organize this. The session will include hands-on installation and running of the system so we require space, computer(s), and organizers. I can dedicate some time and equipment to it and have already talked to a few other members who are interested in this as well. But before going forward we want to know if you will be interested and would it be a 1-day course or a full training of 3-days course? Below is an example of what is covered in a 3-day course: http://freeswitchtraining.eventbrite.com/ A 1-day course might cost peanuts given the trainer offered free training for the 1-day course. Plane tickets, hotel, etc...might bring costs to ~$250 for 20 people which is a sweet deal for getting under the hood with a help of a pro. 3-days course is also offered but I am not sure how much that will cost us. Please let me know your thoughts and if you would be interested to join such a session within the context of TAUG summer meeting. All proceeds are to be used for organizing the event and the plane ticket/hotel for the trainer. - Bruce -- Mike Ashton CTO Quality Track International Work:+1 647 724 3500 x251 Cell:+1 416 527 4995 QTI CONFIDENTIAL AND PROPRIETARY INFORMATION The contents of this material are confidential and proprietary to Quality Track International, Inc. and may not be reproduced, disclosed, distributed or used without the express permission of an authorized representative of QTI. Use for any purpose or in any manner other than that expressly authorized is prohibited. If you have received this communication in error, please immediately delete it and all copies, and promptly notify the sender. - To unsubscribe, e-mail: biz-unsubscr...@taug.ca For additional commands, e-mail: biz-h...@taug.ca - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Bell Fibre ?
In order to get FibeTV you MUST subscribe to Fibe Internet. They will not let you order FibeTV without Fibe Internet. You can order Fibe internet without FibeTV FibeTV viewing usage does NOT count against the Fibe Internet cap. There is a way to get more IP addresses, you could tunnel to a provider that would give you more and have all of your traffic delivered over the tunnel On 11-04-15 11:14 AM, Chuck Mariotti cmario...@xunity.com wrote: It is also important to point out that Fibe internet and Fibe TV are related... my understanding having looked into it early on is that the Fibe connection is the route for both Internet and Fibe IP TV. You MUST have Fibe Internet to be able to purchase Fibe TV (at least on the technology side, maybe different if they use Fibe for other marketing). If you are watching Fibe TV, your internet speed drops (I was told ~7Mbit)... If you watch TV on another TV, it's an additional drop in speed, etc... Whole home PVR is an option, of course, recording TV is the same thing as watching it... there is a speed drop. At first I thought it was a hosted by Bell PVR (which would be interesting!), but at least back then, it was a box in your house... I would bet that there is QOS on the TV packets... so maybe an issue with SIP packets if there are lots of TV Packets. I was also told at the time, that TV Viewing goes against the included bandwidth caps. Which blew me away. I would jump on this in a heartbeat if there was a way to get more IP Addresses! (any ideas? Let me know!) Of course, all the secondary ISPs want access to this stuff for the speed and the TV side (Teksavvy has had IPTV on their coming list for over a year). So maybe it's just a matter of time... Of course, I welcome someone to correct me since when I did look into it, no one knew what the hell I was asking or talking about, and I'm pretty sure some people were just guessing. Chuck -Original Message- From: Douglas Pickett [mailto:douglas.pick...@rogers.com] Sent: April-15-11 11:01 AM To: asterisk@uc.org Subject: Re: [on-asterisk] Bell Fibre ? I'm only speculating here, but I suspect that Fibe IS a DSLAM in the roadside brown boxes as has been mentioned in this discussion, but the high speed backhaul to the CO may or may not be an actual fibre-optic cable. For all practical purposes, does it matter if the backhaul is over fibre, or over copper, as long as Bell offers equivalent performance? I suspect that new installs are probably fibre to the roadside, whereas in older areas such as Reza describes the existing copper facilities will be used. Regards, Doug. On 15/04/2011 10:40 AM, Reza - Voipernetics wrote: I appreciate all the feedback and corrections folks have provided here... However I still don't understand why they tagged it as fiber (unless infact this is a marketing term), because I have **personally** verified and looked on site at the distribution box and have been confirmed by a bell technician on duty at the site, that in the distribution box which is only couple of hundred yards from my client that **indeed** there is no fiber underground (connected to the distribution box). It is all copper. The neighbourhood is also about 20+ years old. I thought, and I am still under the impression that its merely an upgrade of their DSLAM, that allows a higher thorough put of data to make 25 down and 7 up possible. Cheers! -- * *FOUNDER SR. TELECOM ANALYST* /VOIPERNETICS COMMUNICATIONS http://www.voipernetics.com//* NATION WIDE DIDS, SIP TRUNKS VOIP 911. PARTIAL / FULL VIRTUAL PRI - NO CONTRACTS! HOSTED PBX TERMINATION SERVICES. TEL: 647-476-2067 Patrick Song wrote the following on 4/15/2011 10:00 AM: it is a marketing term. it is based on VDSL technology facing to customers but Ethernet Fiber backhual to the network on DSLAM. On Fri, Apr 15, 2011 at 9:32 AM, Reza - Voipernetics r...@voipernetics.com mailto:r...@voipernetics.com wrote: Henry, I have this at my clients. I don't understand why they claim it to be Bell Fiber. The modem they use for this service is actually a DSL modem. The white brick type modem they supply has a led marked DSL,and all the other leds that are similar to a standard dsl modem etc. And yes, upload speed is close to the advertised speed and download close to their advertised speed. -- * *FOUNDER SR. TELECOM ANALYST* /VOIPERNETICS COMMUNICATIONS http://www.voipernetics.com//* NATION WIDE DIDS, SIP TRUNKS VOIP 911. PARTIAL / FULL VIRTUAL PRI - NO CONTRACTS! HOSTED PBX TERMINATION SERVICES. TEL: 647-476-2067 tel:647-476-2067 Henry Coleman wrote the following on 4/14/2011 11:31 PM: Any one have any experience of Bell Fibre Internet. Their F25 product claims to be 7Mbs Upload 25Mbs Download at about $85 per month. -- Thank you Patrick Song CCIE #28023, CCVP M.Eng in Telecommunications Cell:1-647-868-2950 - To unsubscribe,
RE: [on-asterisk] Bell Fibre ?
In the case of IPv6 you could tunnel to one of the various providers out there offering IPv6 tunneling For IPv4, all you need is a router that plugs into your DSL modem that is capable of establishing IPSEC, L2TP, PPTP, or GRE tunnels. Then you find a provider that could give you your IPs and traffic. There are many out there that can do this. We, Telnet Communications, are one of them. (It is a custom setup so don't expect to see pricing on our webpage, email me for details). -Original Message- From: Chuck Mariotti [mailto:cmario...@xunity.com] Sent: Friday, April 15, 2011 12:21 PM To: James Knott; asterisk@uc.org Subject: RE: [on-asterisk] Bell Fibre ? More info _please_! What's your setup? Reliability? -Original Message- From: James Knott [mailto:james.kn...@rogers.com] Sent: April-15-11 11:45 AM To: asterisk@uc.org Subject: Re: [on-asterisk] Bell Fibre ? Bill Sandiford wrote: There is a way to get more IP addresses, you could tunnel to a provider that would give you more and have all of your traffic delivered over the tunnel I do that to get my own IPv6 /56 subnet (2^72 addresses*). I get it from gogoNET http://gogonet.gogo6.com. * I haven't used them all yet. ;-) - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Upgrade test 001
Recv'd -Original Message- From: Simon Ditner [mailto:spdit...@gmail.com] Sent: Friday, March 11, 2011 9:38 AM To: Asterisk Users Group Subject: [on-asterisk] Upgrade test 001 test - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Asterisk 1 Gigabit Internet.
We have done it in selected areas. It's not cheap to deploy and/or operate. It is about $125k per Bell CO by the time you deploy an initial DSLAM with limited configuration (ie not fully loaded). We have 4 of them done Oshawa Toronto - Adelaide Toronto - Simcoe Toronto - Asquith We will be doing 3 or 4 more next year. One of the bigger costs that people don't realize is the backhaul from the CO back to our network at 151 Front (or wherever). Backhaul is expensive. The other problem is that because of the ever expanding use of remotes, the number of customers that can be served from remotes is going down. In a suburban central office like Oshawa we can get to about 25% of the population from the CO. In an a dense urban CO like the 3 downtown Toronto COs we are in we can get to about 65%. The other advantage is that when we can reach the customer from the CO we can use ADSL2+ including Annex M or if customers really need it we can do SDSL at up to 5.7 Mbps per pair. Bill -Original Message- From: Stephan Monette [mailto:monet...@unlimitel.ca] Sent: Wednesday, December 22, 2010 10:09 AM To: Henry Coleman Cc: TAUG Technical Subject: Re: [on-asterisk] Asterisk 1 Gigabit Internet. Has anyone in the group installed their own DSLAMs to provide DSL services and could comment on pricing to do so? I suspect the cost of internet bandwidth is pretty low. The cost is probably on deploying last mile access that is expansive. And then there's maintenance of copper loops, My point is I don't think the cost of bandwidth is high, but the cost to deliver it is very high in Canada considering all the distance we need to cover to connect everyone. Our cities are not as dense as Asian cities or European cities. Don't take my words on it, but I would like to see someone who has the experience in this field to comment on the cost to deploy high speed access. Merry Christmas and Happy New Year! Stephan Monette Unlimitel Inc. Tel.: 1-877-464-6638 Fax: (613) 482-1077 On 2010-12-22, at 9:56 AM, Henry Coleman wrote: Okay, well perhaps I was a little too hard on the CRTC but the limited amount of competition in this industry is the main reason we don't have low cost bandwidth and services that give the consumer what the want. Henry n Mon, Dec 20, 2010 at 11:50 PM, Bruce N het...@hotmail.com wrote: I think Cogeco is the only provider who comes close to what you might want but they operate more to the west of Toronto (I mean cities west of Toronto). 50Mbps download - Upload unknown - I haven't tested it though but $99 sounds fine. http://www.cogeco.ca/cable/on/en/residential/internet/hsi/explore_hsi.h tml -Bruce Date: Mon, 20 Dec 2010 10:51:10 -0600 From: j...@johnlange.ca To: asterisk@uc.org Subject: Re: [on-asterisk] Asterisk 1 Gigabit Internet. The root problem is lack of competition. We have no competition because under our foreign ownership regime, Canada does not allow any. Unfortunately the CRTC are not helping matters, their board consists of ex Bell and Cable people who are very conservative; to the point of choking off any competition before is viable. I'm not a big fan of the CRTC for a lot of reasons but on this particular issue I always feel I have to defend them. The fact is, the CRTC was taking steps to increase competition and one of the first things the Conservatives did when they came to power in 2007 was order them to stop. Here is a nice little article in an archive that explains just how ticked off the bureaucrats at the CRTC were at the Conservatives (conform or quit): http://www.pugetsoundradio.com/cgi-bin/forum/Blah.pl?v-print/m- 1170784107/ And an original story from the CBC that explains what the government did and how happy the incumbents were: http://www.cbc.ca/money/story/2006/06/13/crtc.html The Conservatives have also not lifted the foreign ownership restrictions in Telecom despite several bi-partisan reports that recommended they do so. As recently as this November, Tony Clement announced they would not be allowing competition (foreign owned companies) in Canadian Telecom any time soon. So on this one, it's squarely in the hands of the politicians, not the CRTC. Not to promote my own blog on this list but I've written extensively on competition, deregulation and the CRTC including a whole post on the favorite Canadian sport of bashing the CRTC for everything: http://www.johnlange.ca/2008/07/16/apparently-the-crtc-is-to-blame- for-everything/ -- John Lange www.johnlange.ca --- -- To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org -- *Henry L. Coleman * ***Per: VoIP-PBX.ca * * *
RE: [on-asterisk] Asterisk 1 Gigabit Internet.
John: The actual monthly co-location charges in the Bell CO for space are very affordable. Monthly charges for power is reasonable (could be better), setup costs of the power are ridiculous ($15k plus for 30amps of DC power). The big cost is the exorbitant fees they charge just to setup your colo area. For example $20,000 of project management fees, $2,000 for a fluorescent light fixture (x3), $1,500 for an AC plug to be used for test gear only, etc, etc, etc. For backhaul, yes that is what I'm talking about. The cost to get from the Bell CO back to our network. As for your last point, yes you are correct. That is what I was talking about when I referred to only being able to reach approximately 25% of our target market from the CO. Regards, Bill -Original Message- From: John Lange [mailto:j...@johnlange.ca] Sent: Wednesday, December 22, 2010 11:56 AM To: Bill Sandiford Cc: TAUG Technical Subject: Re: [on-asterisk] Asterisk 1 Gigabit Internet. Bill, couple of follow up questions. Is it true that the co-location charges for locating the DSLAM at the CO are very high? When you say backhaul, are you talking about the connection from your DSLAM back to your network? I had heard that ILECs are increasingly building fibre to the pedestal and locating there DSLAMs there to reduce the length of the copper and increase speeds. Doesn't this effectively eliminate your ability to compete since you aren't allowed to do the same? -- John Lange www.johnlange.ca - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Asterisk 1 Gigabit Internet.
John: We are only moving ahead with additional CO collocates under very specific circumstances. 1) We already have a sufficient volume of customers from that CO to justify the expense. 2) Backhaul can be obtained affordably 3) Any other tangible benefits from that CO that could be of value to us as a CLEC With regards to your point 2 at the bottom, we don't pay the ILEC to get out of the CO in any of our CO collocates and in one case we build our own dark fibre. In that case the cable management fees that you refer to were very affordable. Bill -Original Message- From: John Lange [mailto:j...@johnlange.ca] Sent: Wednesday, December 22, 2010 1:23 PM To: Bill Sandiford Cc: TAUG Technical Subject: Re: [on-asterisk] Asterisk 1 Gigabit Internet. On Wed, Dec 22, 2010 at 10:57 AM, Bill Sandiford b...@telnetcommunications.com wrote: The big cost is the exorbitant fees they charge just to setup your colo area. You could file a Part VII with the CRTC requesting that they review and vary the charges. Have any competitive DSL providers tried that? Problem is all the costing is submitted to the CRTC in secret so it's pretty hard to challenge anything. As for your last point, yes you are correct. That is what I was talking about when I referred to only being able to reach approximately 25% of our target market from the CO. I'm surprised to hear that you are moving ahead with co-lo DSLAM given that percentage of customers will continue to decline. Is there something in place that will allow you access to the pedestal in the future? Bell could turn around tomorrow and convert all their peds to DSLAMs with fibre and you'd be screwed. Co-Lo DSLAM (along with Third Party Access for cable DSL) was the CRTC's original vision for competition in broadband. Largely it has failed for the following reasons: 1) CRTC allowed the ILECs Cable Co's to set the above mentioned insane setup and co-lo charges. This might not have been a show stopper if companies were able to get foreign investment but alas, Canada does not allow it. 2) Even if you do invest in a co-lo DSLAM, the only way to get the traffic out is to buy access from the ILEC and this is unregulated. Even if you ran your own fibre, you still have to pay insane monthly cable management fees. On top of that, in most places in the country there are only 2 ways to the internet, the ILEC or the Cable co. so you pay them again. It's an impossible business model which is why virtually nobody is doing it. -- John Lange www.johnlange.ca - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Asterisk 1 Gigabit Internet.
It depends. If they already have a Bell phone line and we are simply adding OUR service on top of their Bell POTS line, it is very minimal (mainly because the cost of the loop is borne by their POTS service with Bell. If it is a dry-loop, or we are providing the POTS service, it is the Type A unbundled loop rate from the LNI tariff. So for Band A, about $8.50, for Band B about $12.50, and it goes up from there. Anything higher than Band E is cost prohibitive. Bill -Original Message- From: Stephan Monette [mailto:monet...@unlimitel.ca] Sent: Wednesday, December 22, 2010 1:39 PM To: Bill Sandiford Cc: 'John Lange'; TAUG Technical Subject: Re: [on-asterisk] Asterisk 1 Gigabit Internet. Bill, What percentage of your total cost does the copper loop represent per subscriber? Thanks, Stephan Monette Unlimitel Inc. Tel.: 1-877-464-6638 Fax: (613) 482-1077 On 2010-12-22, at 1:32 PM, Bill Sandiford wrote: John: We are only moving ahead with additional CO collocates under very specific circumstances. 1) We already have a sufficient volume of customers from that CO to justify the expense. 2) Backhaul can be obtained affordably 3) Any other tangible benefits from that CO that could be of value to us as a CLEC With regards to your point 2 at the bottom, we don't pay the ILEC to get out of the CO in any of our CO collocates and in one case we build our own dark fibre. In that case the cable management fees that you refer to were very affordable. Bill -Original Message- From: John Lange [mailto:j...@johnlange.ca] Sent: Wednesday, December 22, 2010 1:23 PM To: Bill Sandiford Cc: TAUG Technical Subject: Re: [on-asterisk] Asterisk 1 Gigabit Internet. On Wed, Dec 22, 2010 at 10:57 AM, Bill Sandiford b...@telnetcommunications.com wrote: The big cost is the exorbitant fees they charge just to setup your colo area. You could file a Part VII with the CRTC requesting that they review and vary the charges. Have any competitive DSL providers tried that? Problem is all the costing is submitted to the CRTC in secret so it's pretty hard to challenge anything. As for your last point, yes you are correct. That is what I was talking about when I referred to only being able to reach approximately 25% of our target market from the CO. I'm surprised to hear that you are moving ahead with co-lo DSLAM given that percentage of customers will continue to decline. Is there something in place that will allow you access to the pedestal in the future? Bell could turn around tomorrow and convert all their peds to DSLAMs with fibre and you'd be screwed. Co-Lo DSLAM (along with Third Party Access for cable DSL) was the CRTC's original vision for competition in broadband. Largely it has failed for the following reasons: 1) CRTC allowed the ILECs Cable Co's to set the above mentioned insane setup and co-lo charges. This might not have been a show stopper if companies were able to get foreign investment but alas, Canada does not allow it. 2) Even if you do invest in a co-lo DSLAM, the only way to get the traffic out is to buy access from the ILEC and this is unregulated. Even if you ran your own fibre, you still have to pay insane monthly cable management fees. On top of that, in most places in the country there are only 2 ways to the internet, the ILEC or the Cable co. so you pay them again. It's an impossible business model which is why virtually nobody is doing it. -- John Lange www.johnlange.ca - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Asterisk 1 Gigabit Internet.
I think Reza was referring to the fact that the price per megabit of internet access in Canada is among the worst in the developed world. The reasons for this are mostly what James described earlier. The Berkman Center at Harvard has done extensive research on this. http://cyber.law.harvard.edu/ Bill -Original Message- From: Stephan Monette [mailto:monet...@unlimitel.ca] Sent: Monday, December 20, 2010 9:45 AM To: Reza - Asterisk Consultant Cc: Asterisk Users Group Subject: Re: [on-asterisk] Asterisk 1 Gigabit Internet. Reza, I found a few providers that offers what you are looking for (not at 1Gbps), but you may not be in their serving area: Videotron (province of Quebec only) 120Mbps download, 20Mbps upload: http://www.videotron.com/service/internet-services/internet- access/ultimate-120 Bell FIBE 25: 25Mbps download and 7Mbps upload. http://www.bell.ca/shopping/jsp/pageblock_styles/includes/quickview.jsp ?quickView=truewlcs_catalog_item_sku=DSLTIMONNewMassNCOMX25lang=enre gion=ON But both providers offer cap on internet transit. I feel the caps are too low using such high speed connections. I don't see anything from Rogers that offers similar upload speeds. It may not help you, but it shows that we (as Canadians) will have something similar available in the near future. Stephan Monette Unlimitel Inc. Tel.: 1-877-464-6638 Fax: (613) 482-1077 On 2010-12-20, at 5:35 AM, Reza - Asterisk Consultant wrote: Ok... I'm VERY disappointed in Canada as a country that claims to have one of the highest rate of internet users on the globe per population. The average cost of Cable/DSL for somewhat of the so called High Speed (10mbps down and 1mpbs up) is in the range of ~$60 CDN per month. Just got news from two friends: 1. Friend in Kyoto is having a full blast of amazing 1gbps ** synchronous** internet for less than $150 CDN. 2. Friend in developing nation Moldova former USSR, is paying $15 USD for his 20 Mpbs down and 10 Mbps up. Basic testing with Video via H264 in Asterisk is demonstrating flawless crisp video calls. With speeds such as above in Kyoto and Moldova every household could be an ITSP. To be quite frank, I wouldn't mind paying $250 / month for the capability of 20 Mbps upstream. Can anyone shed some light as to whether we in Canada (in Toronto to be more specific) could reach this speed at this moment at the technical level with the current cable dsl infrastructure? Or is it just the greed and self imposed limitation set by the incumbent internet carriers that are unnecessarily preventing us from greater potential speed? Would appreciate some insight. Best regards, Reza. -- Toronto based VoIP / Asterisk Trainer, I.T. Consultant and Hosted PBX Solutions Provider. +1-647-476-2067. http://www.linkedin.com/in/seminar - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Bell Canada has paid a $1.3 million penalty for violating the National Do Not Call List Rules
Yes, we got this from our legal counsel too. The reality is that $1.3 million may be a drop in the bucket compared to the revenue that they brought in from the activities for which they have been fined. Hard to say. Bill -Original Message- From: John Lange [mailto:j...@johnlange.ca] Sent: Monday, December 20, 2010 5:32 PM To: asterisk Mailing Subject: [on-asterisk] Bell Canada has paid a $1.3 million penalty for violating the National Do Not Call List Rules I guess this isn't really on topic but still; Wow. http://crtc.gc.ca/eng/com100/2010/r101220.htm -- John Lange www.johnlange.ca - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Backup and Restore
Reformat the stick as FAT and your problem should go away.
Can be done from Windows...right click on the drive the in the My
Computer...choose Format...and then select FAT as the file system type from the
dropdown.
Bill
From: Henry Coleman [mailto:henry.cole...@voip-pbx.ca]
Sent: Tuesday, December 14, 2010 10:25 AM
To: TAUG Technical
Subject: Re: [on-asterisk] Backup and Restore
Thanks John, this is a really elegant solution.
Plugging in a USB stick however, CLI reports:
FAT: Unrecognised mount option relatime or missing value
I checked the link you included but there's nothing obvious that would give
this error.
(My stick is formatted NTFS)
Thanks Henry
On Mon, Dec 13, 2010 at 5:43 PM, John Lange
j...@johnlange.camailto:j...@johnlange.ca wrote:
In short; create this file ( /etc/udev/rules.d/11-backup-auto-mount.rules ) by
copying and pasting the text between the start and end tags:
--- start ---
KERNEL!=sd[a-z][0-9], GOTO=backup_auto_mount_end
# Import FS infos
IMPORT{program}=/sbin/blkid -o udev -p %N
# Global mount options
ACTION==add, ENV{mount_options}=relatime
# Filesystem-specific mount options
ACTION==add, ENV{ID_FS_TYPE}==vfat|ntfs,
ENV{mount_options}=$env{mount_options},utf8,gid=100,umask=002
# Mount the device
ACTION==add, RUN+=/bin/mount -o $env{mount_options} /dev/%k
/var/lib/asterisk/backups
# Clean up after removal
ACTION==remove, ENV{dir_name}!=, RUN+=/bin/umount -l
/var/lib/asterisk/backups
# Exit
LABEL=backup_auto_mount_end
--- end ---
You can then insert and remove your key without touching the system. (I'm
making a lot of assumptions about FreePBX that should be correct, for example
that it uses udev).
Lots more suggestions on udev rules here:
https://wiki.archlinux.org/index.php/Udev#Auto_mounting_USB_devices
--
John Lange
www.johnlange.cahttp://www.johnlange.ca
On Mon, Dec 13, 2010 at 4:05 PM, Henry Coleman
henry.cole...@voip-pbx.camailto:henry.cole...@voip-pbx.ca wrote:
Thanks for all the suggestions. So far I think that Doug' solution looks the
most promising, however
it needs automating so that inserting the USB stick will automatically mount
the stick and disconnection will umount the stick.
(A sort of plug and play if you will) Restore might work also based on this
principle.
So I'm gonna pick your Linux brains (again) and ask if this can be done?
Henry [cid:voip-pbx_ca.330@goomoji.gmail]
--
Henry L. Coleman
Per: VoIP-PBX.ca
Re: [on-asterisk] Asterisk 1.8 is out.
It sure is great to finally see IPv6 support in Asterisk. With only 12 IPv4 /8s remaining in the IANA free pool and complete exhaustion of the free pool likely to occur later this year or early next at the latest, Asterisk users would certainly be behind the 8-ball without IPv6 support. Bill Sent from iPad On 2010-10-22, at 1:06 AM, Reza - Asterisk Consultant aster...@neoenova.com wrote: For all those of you who are curious and who might have missed an update - Asterisk 1.8 is out! This has been released and submitted less than 24 hrs ago!The reason for my excitement here is 1.8 supports Secure RTP and IPv6 support right inside the SIP channel driver. -- Toronto based VoIP / Asterisk Trainer, I.T. Consultant and Hosted PBX Solutions Provider. +1-647-476-2067. http://www.linkedin.com/in/seminar - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Can a 416 number be ported over as a DID onto a 905 PRI link from Bell Canada? Reps tell me NO and I don't understand why.
As a CLEC, we offer forwarding back to the PSTN on a per concurrent channel basis for this type of stuff. Before doing so, we need to know the expected channel volume and type of business (ie, our rates are SIGNIFICANTLY higher for call centres). Bill -Original Message- From: Nabeel Jafferali [mailto:nab...@x2n.ca] Sent: Friday, October 08, 2010 3:53 PM To: 'Bruce N'; 'asterisk Mailing' Subject: RE: [on-asterisk] Can a 416 number be ported over as a DID onto a 905 PRI link from Bell Canada? Reps tell me NO and I don't understand why. You can only port a number to a PRI if it is in the same rate centre as the PRI. That's standard practice for all telcos, except for some that may offer virtual multi-rate-centre PRIs (usually for an extra fee). Use localcallingguide.com to look up the rate centres of your current DIDs and any DIDs you are interested in porting. -- Nabeel Jafferali X2 Networks Inc. -Original Message- From: Bruce N [mailto:het...@hotmail.com] Sent: October-08-10 3:48 PM To: asterisk Mailing Subject: [on-asterisk] Can a 416 number be ported over as a DID onto a 905 PRI link from Bell Canada? Reps tell me NO and I don't understand why. Hi Everyone, Just got off the phone with a Bell Canada PRI rep and was disappointed when she told me that a 905-686- and 416-686- number can not be ported over to a PRI link which is sourced from the Melton exchange 905-405- x. Both the 416-686 and 905-686 are currently SNR from Bell Canada. The reason was that none of these numbers are from the same exchange and that NEVER a 416 can come onto a 905 PRI. I understand the rationality behind different exchanges but I thought this wouldn't be a problem if it was brought over as a DID. Previously we asked Bell Canada to port over a MALTON010MD switch number (belongs to Allstream) to Bell Canada switch MALTON22CG1 and they said it's not possible as a standalone business number but then they were able to bring it over as a DID onto the MALTON22CG1 and take it away from Allstream. So, in practice, it's possible to change switches and even snatch it from another provider. I don't see why they won't bring over a 416 or 905- 686 that belongs to Ajax-Pickering over to a Malton switch. Do Rate Centers determine the portability rather than the switches whereabouts? Are they only doing because their billing system is not capable of tracking calls if numbers are moved to different rate centers? Can someone please shed light on it or let me know if I was talking to a rep who was misinformed? Also, we can try to port this number out to a third party and then port it back into Bell if that makes a difference or if it makes them accept it. Thanks,Bruce - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] VoIP Pay Phone?
Googled voip payphone and got this back as number 1 result http://www.voip-info.org/wiki/view/VOIP+Payphones -Original Message- From: Henry Coleman [mailto:henry.cole...@voip-pbx.ca] Sent: Monday, September 20, 2010 1:24 PM To: asterisk@uc.org Subject: Re: [on-asterisk] VoIP Pay Phone? You will probably have to use an ATA with an analog Payphone from a company like Payphone.com H On Mon, Sep 20, 2010 at 12:09 PM, Chuck Mariotti cmario...@xunity.comwrote: I have a client that is paying bell to have a payphone on their property (not enough volume to justify the income from the phone for bell to put it there). Does anyone know if there is a VoIP phone that takes coins? Or if someone makes a LD call, the system would ask for more money to be inserted? Or are there other recommendations? Calling cards? Where can I get cards printed up?, etc... Regards, Chuck -- *Henry L. Coleman * ***Per: VoIP-PBX.ca * * * - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Crowd sourcing rules for blocking hacking attempts?
Matthew: I think something like this is good, although be very very careful as doing so could get you sued. If you don't believe me, ask my friends Paul Vixie and Dave Rand. They are the inventors of one of the first (IF NOT THE FIRST) e-mail RBLs (MAPS) and doing so got them sued. http://en.wikipedia.org/wiki/Mail_Abuse_Prevention_System Vixie also happens to be the guy that wrote BIND - the DNS server that runs over 75% of the Internet. Once again, I think it's a great idea...just be sure to protect yourself. Bill -Original Message- From: Matthew Gamble [mailto:mgam...@mgamble.ca] Sent: Wednesday, September 01, 2010 8:21 PM To: Bruce N; asterisk Mailing Subject: Re: [on-asterisk] Crowd sourcing rules for blocking hacking attempts? Bruce, What I'm proposing (and actually just started writing the code for) is a system where we allow anyone to sign up (the power of the crowds) but require a few things: 1) Authenticated email address. Not hard to get, but it does stop random signups 2) Reports from new accounts are not added to the global list for X days to monitor the quality of the data they are submitting. Further to the above, I'm adding a score feature to the output, so when you request a list of bad hosts you would get a file with IP, last reported date, and score. The score would be a function of a few things: 1) How well do you trust the reporter(s)? Age of accounts, never flagged for reporting bad data, etc 2) How many people reported this IP? 1? It wouldn't be in the database until a few different sites reported it, etc 3) Other criteria I'm still writing. The third piece of security would be a system for people to flag data as being bad, creating a feedback loop to ensure that if a person submitted false data that it was quickly removed from the DB. Remember that crowd sourced rule systems already work for email (Cloudmark for example) and with a trust system and good scoring rules the issue of false positives becomes much less of a risk. On Wed, Sep 1, 2010 at 8:13 PM, Bruce N het...@hotmail.com wrote: Hello Mathew, Are you suggesting an open system for everyone and anyone to input an IP address? Two scenarios: 1- Allow only people who you trust - CONS: a- Still can't negate the fact that some authorized user may mistakenly put a client's IP in the BAD IP table. b- Limiting the number of reported BAD IPs to the number of trusted people which I would like to believe would be very small or else it won't be a trusted circle. PROS: a- Can be MORE or LESS a trusted database - As long as no bulk IPs are allowed to be entered and there are restrictions to add more than 1 IP per hour let's say. 2- Allow anyone to sign-up and add BAD IPs. CONS: a- Anyone can sign-up. Even the cracker!!! He can put our legit IPs in the database and BOOM, shutdown service for clients for no good reason. I mean an IP that is BAD today can be a potential customer tomorrow. What would be the rules to remove them when you have a whole bunch of people submitting these - specially if this grows really big. b- The list will grow so big that it won't be possible to handle or it might again block legit users as the attacks are usually co-ordinated not from the cracker IP address but rather compromised servers and it might literally block a good portion of the USA continental as lots of attacks do originate from compromised servers in USA while the cracker is enjoying his tea break in Russia. PROS: a- Would be a more complete list of BAD IP addresses. These work around will be somehow useful but isnt' it about time that SIP becomes more transparent to the common folks (simpler, less ambiguous output, and more manageable SIP debug) - as it's becoming more commonplace now-a-days? Or maybe pay more attention to it's security feature innately like other popular protocols rather than keeping them as an option for the user to turn on? As an example, just few years ago, all wireless routers were possible to setup without a wireless security (one could literally jump from neighbour to neighbour in the whole block) and now any router you take out of the box either has a randomly generated wireless password or asks for one before setting up the wireless leaving you with no access to neighbours hot spot. -Bruce Date: Wed, 1 Sep 2010 19:48:54 -0400 From: mgam...@mgamble.ca To: asterisk@uc.org Subject: [on-asterisk] Crowd sourcing rules for blocking hacking attempts? I've been following the threads over the past weeks about Asterisk hacks being on the rise, and I have to say I've been seeing the same thing in my logs. I'm wondering if there is any
[on-asterisk] Fwd: CRTC Decision - My summary
options. Cables carriers are ordered to make GigE interconnections available to competitors. The Commission has also determined that as higher speed options become available (presumably 10G, 40G and 100G), the incumbents should make them available to competitors. 3. Restrictions on use. Commission has instructed the cable carriers to remove the working regarding LAN connection services and VPN services from their tariffs. (it is interesting to note that there is no mention of VoIP or IPTV in this section of the decision). The Commission has also found that ILECs and cable carriers do NOT have to make multi-casting functionality available at a wholesale level. With regards to static IP’s on cable, the Commission has noted that the cable carriers have claimed that they cannot provide static IPs to TPIA customers, however static IPs are available to their own cable customers and also static IPs are available to ILEC ADSL customers. In light of this, the Commission has ordered the cable carriers to show cause within 30 days from today as to why they cannot provide IP addresses to TPIA customers. 4. ITMP for aggregated ADSL and TPIA. The commission finds that is premature to decide on this matter at this time and instead will do so as part of Bell Canada’s applications to review and vary the UBB decision. ACCESS TO NEW INFRASTRUCTURE The Commission has decided that it is not going to get caught up in word games about what is Next Generation Networks (NGN) or not. They have determined that the facilities that are subject to wholesale obligations include FTTN and DOCSIS 3.0 facilities. When newer technologies are deployed it will assess them on a case-by-case basis, consistent with the requirements of the Telecommunications Act, the Policy Direction, and the principles set out in the Order in Council. CO-based ADSL access service from the ILECs and local head-end based cable access service from the cable carriers. In a nutshell, the Commission has denied the implementation of CO-based ADSL services, and refused to force the cable carriers to unbundle their networks. They have done so, in part, on the believe that only 2 of 5 ISPs would take advantage of such a service and as such there would not be a sufficient lessening or prevention of competition without these services. It is interesting to note that Commissioner Tim Denton dissented with regards to this portion of the decision and his excellent 5 page dissent is attached and may provide us with adequate grounds for appeal should we decide to go that way. So, that’s it for me from Gatineau. I hope you have found this summary helpful. Bill Sandiford Telnet Communications w: 905-674-2000 x100 f: 905-728-7918 b...@telnetcommunications.commailto:b...@telnetcommunications.com IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
Re: [on-asterisk] Meeting ?
WHEN: AUGUST 19TH, AT 7:00 PM. WHERE: NORTH YORK CIVIC CENTRE / COUNCIL CHAMBERS.* 5100 Yonge Street, Toronto, ONx-apple-data-detectors://2 M2N5V Bill Sent from iPad On 2010-08-18, at 10:06 AM, Henry Coleman henry.cole...@voip-pbx.camailto:henry.cole...@voip-pbx.ca wrote: Can't remember if the meeting is tonight or on thursday Please advise H -- Henry Coleman
RE: [on-asterisk] 3.5G Router to connect your USB 3.5G Modem, aka.Internet Stick.
Also, there are external antennae available for almost all models of 3G sticks. There are omni-directional and directional antennae available depending on whether you want a solution for fixed location or mobility. Email me offline if you want some pricing on them, we are a dealer. -Original Message- From: Liviu Toma [mailto:liviu.t...@gmail.com] Sent: Monday, August 02, 2010 8:23 AM To: asterisk@uc.org Subject: Re: [on-asterisk] 3.5G Router to connect your USB 3.5G Modem, aka.Internet Stick. One thing nobody has mentioned was which network was the 3G stick from. I doubt all carriers have the exact same coverage. Liviu On Mon, Aug 2, 2010 at 8:16 AM, Frank Bax f...@sympatico.ca wrote: Andrew Kohlsmith (mailing lists account) wrote: On Sunday, August 01, 2010 02:03:07 pm Bruce N wrote: Impressive. Reza, few months ago, I was testing a Rogers wireless modem from Toronto to London and there were lots of dead spots. I don't think the driving speed mattered. Drivign at 120 km/hr it was still functioning well. But there were some serious gaps and loss of signal in large parts of the trip. There are dead spots in 3G coverage on the 401 corridor between Toronto and London? You've *got* to be kidding me I've done some serious driving around that area in the last few years and listen to CBC radio (streaming over 3G, better than futzing with stations) and I've never had a dropout. We used bluetooth from laptop to 3G cellphone on trip from London to Toronto for the first time last weekend; lost connection several times. Is it possible radio streaming has some software buffers? - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
[on-asterisk] SIP for Blackberry
I thought that the group may find this an interesting read. http://connectedplanetonline.com/mobile-apps/news/rim-wi-fi-sip-fmc-0426/ Bill Sandiford Telnet Communications w: 905-674-2000 x100 f: 905-728-7918 b...@telnetcommunications.commailto:b...@telnetcommunications.com IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
[on-asterisk] DPI Website
Hi All: Excuse the cross-post please, but there is a new website out there that some on both lists might find very interesting. http://www.deeppacketinspection.ca/ Bill
RE: [on-asterisk] Re: TAUG T-Shirts Available
Does it come with free batteries? :) -Original Message- From: Dave Donovan [mailto:donovan.da...@gmail.com] Sent: Tuesday, April 06, 2010 12:13 PM To: asterisk@uc.org Subject: Re: [on-asterisk] Re: TAUG T-Shirts Available On Tue, Apr 6, 2010 at 11:30 AM, Mike Ashton mike.ash...@qualitytrack.comwrote: Wow, won't loose you at night with that on! You've inspired my shirt slogan: TAUG T-Shirts for Sale: Now in 'Pylon Orange'! or Statistic: TAUG members 30% less likely to be hit by vehicles on their way home from a meeting, when compared to other OSS user groups. It's a bold colour to be sure. It's not as fluorescent as it might seem. I used a flash and that wall behind me is midnight blue. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Asterisk as a reduntant/fail over solution
Hi Andrew: You are correct, current calls are still dropped. Bill -Original Message- From: Andrew Heagle [mailto:and...@logaan.com] Sent: Monday, February 15, 2010 7:41 PM To: asterisk@uc.org Subject: Re: [on-asterisk] Asterisk as a reduntant/fail over solution On Wednesday 10 February 2010 15:33:24 Dave Donovan wrote: On Wed, Feb 10, 2010 at 1:40 PM, Robert Brock robert.br...@mks.com wrote: Coming from the Nortel world (well Avaya now) I have integrated an Asterisk server to all our Meridian PBX systems to support VOIP and interoffice calls using G729 and it actually works better than the Nortel solution. Over the next few years we are planning to retire our Nortel PBX systems, however these system are rock solid from reliability point of view; I have not had any problems with the Asterisk servers that I setup and they are working much better than expected. With that said I would like to hear/know about what people have done to make the asterisk server a high availability solution. Hi Robert, You might get some good ideas from a presentation that Bill Sandiford, one of our members, made to the group a couple of years ago. http://taug.ca/node/68 Hi, If one were to implement this, would current calls be dropped or would they get re-established after the fail-over from the master to the slave was completed? It didn't mention this in the presentation. I'm guessing current calls would still be dropped. Thanks, Andrew - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
[on-asterisk] RE: Long distance fraud... $24,000+
Chuck: Unfortunately your customer may be stuck. I can tell you that we had one case of this in the past here at Telnet where one of our customers was hit by fraud (they had an insecure box, wasn't our fault) and ended up with a $1,500 bill after about 24 hours...luckily we noticed it and let them know after 24 hours. So how did it pan out for our customer. Although we (Telnet) felt no responsibility whatsoever to do anything (as it wasn't our fault), in the spirit of being co-operative and compassionate we agreed to re-rate all of the customers calls at our cost. We did have a hard cost for those calls and we didn't feel that we should be out our costs. I don't know if Allstream will agree to the same (I doubt it), but you can try. One thing that you may try is asking them why their fraud management dept didn't pick up on this earlier. Regards, Bill -Original Message- From: Chuck Mariotti [mailto:cmario...@xunity.com] Sent: Friday, January 29, 2010 11:14 AM To: asterisk@uc.org Subject: [on-asterisk] Long distance fraud... $24,000+ Anyone have any experience with large long distance phone bills ($20k) that are fraudulent? The phone system was compromised via dial in / call transfers. Overseas calls made. Specifically how to not have to pay All Stream because of it? What's the common practice and outcome? I mean, I would imagine that All Stream would get their costs back out of it eventually, how can they pass that onto their client? How can I go about getting them to zero it out? Regards, Chuck Mariotti - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Router Recommendations
I currently have a WRT54GL in my home running the MLPPP version of Tomato, and it is pretty solid but does lock up from time to time. The lockups aren't to troublesome in my home situation, but would be annoying in a business environment. We found the same thing in the field for most of the readily available routers, whether they be Linksys, D-Link, Buffalo or otherwise. Most of the time they were pretty good, but in certain circumstances they just locked up, or wouldn't reconnect PPPoE after an outage, or other weird stuff. For that reason, we are now solely deploying Cisco 1721 routers for all of our business customer deployments (whether they use VoIP or not). You can pick them up on eBay from a variety of sources for $100. I think we bought 100 of them for $50 each. Then we put the WIC-1ADSL card into the router (they are also around $50 on eBay). In some cases we put in 2 DSL cards and bond the links with MLPPP. The great part of this solution is that for around $100 (for the single DSL, or $150 for dual) we get a router that runs Cisco IOS and all the great things that come along with that. The reliability is outright awesome...they just never need to be rebooted. The downside is no web interface, so you have to know Cisco IOS or be fairly comfortable with a command-line interface. Also, there is no wireless in this series of routers, so you will need some sort of stand-alone AP if the customer wants wireless (most of them do). Regards, Bill -Original Message- From: Wai Vii [mailto:wai...@gmail.com] Sent: Wednesday, January 20, 2010 6:06 PM To: TAUG Technical Subject: Re: [on-asterisk] Router Recommendations Another vote for Tomato, the traffic shaping works great whereas it just seemed to cause problems with DD-WRT. Used to have DD-WRT loaded on up to ten WRT54GS but found it slower than Tomato and the interface more cumbersome. Another vote for the ASUS routers mentioned. Heard that the Buffalo routers are OK too but I've never used one before. If you want to spend a bit more, consider Soekris or Routerboard. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Router Recommendations
To my knowledge the only Ethernet WICs available for the 1721 are the WIC-1ENET which is single 10BaseT only. Do not confuse WIC-4ESW to be a 4 port Ethernet card either. It is a 4 port Ethernet switch. It does however support 802.1q vlan trunking, so it may be possible to separate the ports that way using subinterfaces and vlans. Keep in mind however that PPPoE is not supported on subinterfaces, but I believe DHCP is. (translation for cisco laymen...you won't be able to use the WIC-4ESW ports for PPPoE connections like DSL, but you may be able to use it for DHCP connections like Cable and/or satellite) I know someone who inadvertently bought a WIC-4ESW thinking it would work for them. I'll see if they still have it and if they do I'll try and do some testing with it (as time permits). Bill From: Bruce N [mailto:het...@hotmail.com] Sent: Thursday, January 21, 2010 2:08 AM To: Bill Sandiford; wai...@gmail.com; asterisk Mailing Subject: RE: [on-asterisk] Router Recommendations Sounds like a really solid/resonably priced option. Cisco 1721 has a one 10/100 Fast Ethernet Port. I am looking to use this as a load balancer for three ISPs if it's possible with this router. Providers are: Bell (ADSL) - RJ-11 interface = WIC-1ADSL Rogers - RJ-45 interface = ? Sattalite- RJ-45 interface = ? POE Switch - RJ-45 interface = ? So, in total 3 RJ-45 and 1 ADSL port is needed. I can live with 3 RJ-45 and no ADSL ports as well. Supporting 100mbps on all RJ-45 ports would definitely be a bonus. I know that the router has two WIC slots. WIC-1ADSL exists as Bill suggested. Is there another WIC which can support two 10/100Base RJ-45 base in the other WIC slot? Or maybe even a one port 10/100Base? The reason why I am posing this question is because I only found a one port 10Base WIC module on the list of compatible modules for this router and no 100Base WICs. Thanks, Bruce From: b...@telnetcommunications.com To: wai...@gmail.com; asterisk@uc.org Date: Wed, 20 Jan 2010 23:02:59 -0500 Subject: RE: [on-asterisk] Router Recommendations I currently have a WRT54GL in my home running the MLPPP version of Tomato, and it is pretty solid but does lock up from time to time. The lockups aren't to troublesome in my home situation, but would be annoying in a business environment. We found the same thing in the field for most of the readily available routers, whether they be Linksys, D-Link, Buffalo or otherwise. Most of the time they were pretty good, but in certain circumstances they just locked up, or wouldn't reconnect PPPoE after an outage, or other weird stuff. For that reason, we are now solely deploying Cisco 1721 routers for all of our business customer deployments (whether they use VoIP or not). You can pick them up on eBay from a variety of sources for $100. I think we bought 100 of them for $50 each. Then we put the WIC-1ADSL card into the router (they are also around $50 on eBay). In some cases we put in 2 DSL cards and bond the links with MLPPP. The great part of this solution is that for around $100 (for the single DSL, or $150 for dual) we get a router that runs Cisco IOS and all the great things that come along with that. The reliability is outright awesome...they just never need to be rebooted. The downside is no web interface, so you have to know Cisco IOS or be fairly comfortable with a command-line interface. Also, there is no wireless in this series of routers, so you will need some sort of stand-alone AP if the customer wants wireless (most of them do). Regards, Bill -Original Message- From: Wai Vii [mailto:wai...@gmail.com] Sent: Wednesday, January 20, 2010 6:06 PM To: TAUG Technical Subject: Re: [on-asterisk] Router Recommendations Another vote for Tomato, the traffic shaping works great whereas it just seemed to cause problems with DD-WRT. Used to have DD-WRT loaded on up to ten WRT54GS but found it slower than Tomato and the interface more cumbersome. Another vote for the ASUS routers mentioned. Heard that the Buffalo routers are OK too but I've never used one before. If you want to spend a bit more, consider Soekris or Routerboard. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org Tell the whole story with photos, right from your Messenger window. Learn how!http://go.microsoft.com/?linkid=9706112
RE: [on-asterisk] Scratch Card Printing
http://www.directcheckmarketing.com/ http://www.tele-pak.com/ http://www.scratchoff.com/ http://www.chinasuppliers.globalsources.com/china-suppliers/Scratch-Card-Printing.htm -Original Message- From: Bruce N [mailto:het...@hotmail.com] Sent: Friday, January 01, 2010 3:39 PM To: asterisk Mailing Subject: [on-asterisk] Scratch Card Printing Hi Guys, Do you guys know of any local or maybe even international scratch card (calling card type) printing service provider? Happy New Year. Thanks,-Bruce _ Windows Live: Make it easier for your friends to see what you're up to on Facebook. http://go.microsoft.com/?linkid=9691816 - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Need help tracing a number
Bruce: Are you trying to figure out the actual current provider of an actual number? For example, the number has been ported away from the original carrier so you can't rely on the info about the NPA-NXX block? If so, I can look this up for you if you send me the info off-list. Regards, Bill -Original Message- From: Bruce N [mailto:het...@hotmail.com] Sent: Tuesday, November 24, 2009 9:38 PM To: asterisk@uc.org Subject: [on-asterisk] Need help tracing a number Hi Guys, I have been checking all sources I could think of (411 reverse lookup, local calling guide . com, etc...) but it seems that the two phone numbers I am looking for have been part of Call-Net Communications and Fido which got transfered over to different providers. Is there a real list or some way to find out who owns these? Maybe an industry insider can find out? Many thanks,Bruce _ Windows Live: Keep your friends up to date with what you do online. http://go.microsoft.com/?linkid=9691815 - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Router/vpn devices that are VoIP friendly
Don't forget QoS !!! -Original Message- From: Aloysius Thevarajah Lloyd [mailto:lloyd.aloys...@gmail.com] Sent: Thursday, September 03, 2009 10:35 AM To: Dave Donovan Cc: asterisk@uc.org Subject: Re: [on-asterisk] Router/vpn devices that are VoIP friendly Currently I am using PC Engines Alix + m0n0wall in production environments. So far no problems. It is working really good. In the Past I have bad experience with VOIP+pfsense. I never try the most recent version 1.2.3. But I would say a VOIP friendly router should have the Following features 1. WAN Port - PPOE - DHCP - STATIC - PPOE + MLPP 2. LAN Port's - DHCP - VLAN - DHCP Option 66 3. VPN Support 4. Should pass the TFTP traffic from WAN to LAN 5. WAN Failover 6. Monitoring tools But I could not find any open source firmware support all of the above. Thank you. A.T.Lloyd On Wed, Sep 2, 2009 at 4:00 PM, Dave Donovan donovan.da...@gmail.comwrote: On Wed, Sep 2, 2009 at 1:51 PM, Dave Donovandonovan.da...@gmail.com wrote: I like the suggestion of using the Snoms for remotes. I never would have guessed that a phone had OpenVPN built in. Maybe the Alix is a good solution for your aggregation point. I'm going to reply to my own post here and add some disclosure, lest anyone get false hope or be misled. In my environment I'm running pfSense at my remote sites on refurb Dell P4s. I've bough the Alix systems and I'm testing them pending deployment. So far things look good. At my head office, I'm not running pfSense. I'm running Untangle because it has a web filtering, antivirus, antispam, etc and a cool interface for generating and distributing the OpenVPN install packages with the keys and everything all rolled up. It's gone wonky on me a few times and I've sworn to rip it out but then reconsidered the calmness of the following day. The Untangle system is not meant for the hacker set. It's the Trixbox of routers. Dave - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Re: [biz] Greetings from Reza - a TAUG enthusiast from the past.
Telnet Communications will donate the $100 if need be to get a room for a meeting, assuming that is the route you want to go. Bill -Original Message- From: spdit...@gmail.com [mailto:spdit...@gmail.com] On Behalf Of Simon P. Ditner Sent: Tuesday, July 21, 2009 11:34 AM To: Keith Major | Aquarius Telecom Cc: Reza Asterisk Consultant; b...@taug.ca; asterisk@uc.org Subject: [on-asterisk] Re: [biz] Greetings from Reza - a TAUG enthusiast from the past. I was really hoping they'd just be done with it by now... I'll look into some pubs with party rooms. U of T has rooms for ~ $100 a session. On Tue, Jul 21, 2009 at 11:24 AM, Keith Major | Aquarius Telecomk.ma...@aquariustel.com wrote: Welcome Home Reza! I was thinking the city workers strike has put a dent into our monthly meetings since we use their facilities for most of our meetings. Any ideas on a place we could use as a one-off for a meeting? Best Regards, Keith Major Network Engineer Business Development Aquarius Telecom Inc. www.aquariustel.com Hosted PBX Services Business VoIP Solutions ___ This e-mail may be privileged and/or confidential, and the sender does not waive any related rights and obligations. Any distribution, use or copying of this e-mail or the information it contains by other than an intended recipient is unauthorized. If you received this e-mail in error, please advise me (by return e-mail or otherwise) immediately. Ce courrier électronique est confidentiel et protégé. L'expéditeur ne renonce pas aux droits et obligations qui s'y rapportent. Toute diffusion, utilisation ou copie de ce message ou des renseignements qu'il contient par une personne autre que le (les) destinataire(s) désigné(s) est interdite. Si vous recevez ce courrier électronique par erreur, veuillez m'en aviser immédiatement, par retour de courrier électronique ou par un autre moyen. -Original Message- From: Reza Asterisk Consultant [mailto:aster...@neoenova.com] Sent: July-21-09 2:47 AM To: b...@taug.ca; asterisk@uc.org Subject: [biz] Greetings from Reza - a TAUG enthusiast from the past. *Hello Everyone:* I don't mean to cross post - but I do have a lot of friends, associates and people I have done business with from both the tech list and the biz list (almost 100 of you I know personally). As most of you know I had left for overseas opportunities in January 2008 and have traveled as far as Malaysia. I am back home in Toronto, re-settled and would like to initiate a get together, spike up interest and boost the TAUG energy once again. I understand the city strikes has put a LOT of things in chaos mode and in the midst of the chaos - most Torontonians are finding innovative solutions to problems that have generated as a result. I would like to organize a get together and will try to contact some of my sources to see if they would be generous to extend to us some of their facilities. I'm anxious to meet all of you whom I've met over the course of the last few years. Please don't reply to this thread, instead contact me directly if you are interested to meet up - and I will try to organize a meeting place for us. For those of you who do not know me -- I'm your TAUG promoter at major conference and trade shows - almost everywhere when there is an opportunity and a free booth was provided to us and have actively participated as a trainer and public speaker on Asterisk at international and local events. I have also represented promoted TAUG locally and at overseas events, non-profits, user groups - and at Career Fairs where TAUG booths were generously provided to us. Looking forward to connect with all of you with whom I've connected with in the past. *Best wishes, Reza, Toronto based Asterisk Trainer, ** Consultant ** Enthusiast. * - To unsubscribe, e-mail: biz-unsubscr...@taug.ca For additional commands, e-mail: biz-h...@taug.ca -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
[on-asterisk] Fire at 151 Front Street
If anyone is affected by the fire at 151 Front and needs emergency gear, collocation, internet transit, etc. give me a shout. Telnet will give assistance free of charge to any TAUG member that needs help. My cell phone is 905-409-5228. Don't hesitate to call me if you need help. Bill Sandiford Telnet Communications w: 905-674-2000 x100 f: 905-728-7918 b...@telnetcommunications.commailto:b...@telnetcommunications.com IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
RE: [on-asterisk] Backup and restore to a USB stick
Exactly what Johannes said, but based on the output of your /var/log/messages I think the second step will be mount /dev/sda /mnt/usb -Original Message- From: Johannes Vanderknyff [mailto:johannes.vanderkn...@gmail.com] Sent: Tuesday, June 16, 2009 3:13 PM To: aster...@voip-pbx.ca Cc: asterisk@uc.org Subject: Re: [on-asterisk] Backup and restore to a USB stick Hmmm. Doesn't look like it is mounted yet. I just googled linux how to mount usb drive... Create the directory mkdir /mnt/usb Next, mount the drive mount /dev/sda1 /mnt/usb Then, see if it works: cd /mnt/usb ls You should see a list of the files on your USB drive. Johannes On Tue, Jun 16, 2009 at 3:07 PM, Henry L.Colemanaster...@voip-pbx.ca wrote: This is what I get r...@pbx:~ $ mount /dev/hda2 on / type ext3 (rw) proc on /proc type proc (rw) sysfs on /sys type sysfs (rw) devpts on /dev/pts type devpts (rw,gid=5,mode=620) /dev/hda1 on /boot type ext3 (rw) tmpfs on /dev/shm type tmpfs (rw) none on /proc/sys/fs/binfmt_misc type binfmt_misc (rw) - Henry L. Coleman [VoIP-PBX.ca] = Johannes Vanderknyff (NOTE: I don't know much about FreePBX) If you can get to the console, type mount and that should give you a listing of mounted filesystems. If FreePBX mounted the USB drive, you'll be good to go and you can just copy (cp) the files to that location. Johannes On Tue, Jun 16, 2009 at 1:48 PM, Henry L.Coleman aster...@voip-pbx.cawrote: Hi all, as many are aware FreePBX has a backup and restore function that can schedule a backup of conf, cdr, and vmail. This works very well but I need to back the files up to a USB stick instead of a default backup directory. My question (not being a good Linux man) is how do I the redirect the backup path to this device. When I plug the stick in, the consule indicates that the stick is recognised and is working but does't tell me where in the directory structure to find it. Help please? Henry L.Coleman [VoIP-PBX.ca] - - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] service in sacromento
Try Voice Network...Norm has USA DIDs. -Original Message- From: Dean Yorke [mailto:dean.yo...@xyc.ca] Sent: Wednesday, April 29, 2009 4:50 PM To: asterisk Mailing Subject: [on-asterisk] service in sacromento Hi All, Anyone know of a good service provider that can give me a number in sacramento for a week while I am on vacation. Thanks - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Changing the outbound CID via the dial plan
Replace ) with CALLERID(num)
-Original Message-
From: Apache [mailto:apa...@tsx3.computeradvocacy.com] On Behalf Of Henry
L.Coleman
Sent: Friday, April 24, 2009 4:15 PM
To: asterisk@uc.org
Subject: [on-asterisk] Changing the outbound CID via the dial plan
Hi all, I've been attempting to change the outbound CID from an extension using
the dial plan.
exten = *37,1,Read(SPOOFID,agent-newlocation,10,3)
exten = *37,2,Set()=${SPOOFID})
exten = *37,3,SayDigits(${SPOOFID})
Help anyone
Henry L.Coleman [VoIP-PBX.ca]
-
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[on-asterisk] FW: **Maintenance Notification Update**
Tim This is probably why you had an outage today. Your provider should have notified you of this in advance. There were troubles during this maintenance today that caused generator power to be lost to some parts of the building. Luckily for us our equipment is not located in the sections that were affected. We had our own staff on site all day just in case. Not sure about your provider, but I believe they have representatives on this list and they may address this issue. ( I will leave them nameless as well ) Bill -Original Message- Sent: April 18, 2009 5:59 AM Subject: FW: **Maintenance Notification Update** *Scheduled Maintenance Update* Date: April 18, 2009 Time: 06:00hrs EST · Update: Maintenance window is now OPEN to perform the scheduled maintenance listed below During the hours: 06:00hrs until 20:00hrs Local time at the site, on Saturday, April 18 we will be performing the following activities: * 151 Front Street will be performing their annual building shut down. * During the above maintenance window, all commercial power to the building will be disconnected for maintenance and upgrades. All suites will transfer to Generator Power for the duration of the window. **NO SERVICE INTERUPTION IS EXPECTED** Unless otherwise indicated, the scope of work to be performed by the building will include: 600 volt cell for switch board 175 An additional 13.8kv cell, located in the main hydro vault Thermal Scan on all electrical room services Ultra sound of HV Cells Complete inspection of all Hydro Vault electrical services Repair of any deficiencies uncovered during the required inspection ***Please note, there will only be limited access to the building/equipment during the above operational window. Staff will be on site to handle any technical/emergency needs*** If you have any questions or comments about the above maintenance, please contact us. Thank you for your cooperation.
RE: [on-asterisk] Various regulatory issues that could affect you
increases and cost of provisioning decreases accordingly, how often will the commission revise the tiers and associated charges? Every day that goes by at a fixed tier, Bell's profit margin increases and so does it's competitive advantage ISPs; creating a regulated preferential environment. As disappointed as I am with this single application, it should be noted that it is more disconcerting when taken as a whole with Bell's other efforts to avoid speed matching and force traffic shaping upon their ISPs. Of further concern is Bell's view that it can use the CRTC as a forum of convenience and resort to political interference (section 4 of the tariff application) when due process yields an undesired outcome. I urge the commission to act in the interest of the public and ensure that the regulatory framework promotes diversity of services, pricing and competition. The inherent conflict of interest presented by the dual nature of 'Bell the upstream wholesale provider', and 'Bell the marketplace competitor' must be checked by the regulatory authority of the commission. I ask that the commission deny Bell's application for a tiered pricing as set out in tariff application 7181, and act quickly to seek compliance with it's CRTC Telecom order 2009-111 which would bring speed matching to the wholesale DSL marketplace and freedom of choice to consumers. Sincerely, David Donovan Toronto Ontario, Canada On Tue, Apr 14, 2009 at 2:40 PM, D. Hugh Redelmeier h...@mimosa.com wrote: I'm repeating an old message since I think that it is important and urgent. I've seen no response to it on this list (the biz list had some). Note: this is the last day to comment on URB. It's easy: there is an awkward web form. Here's how Teksavvy explained using it: http://support.crtc.gc.ca/crtcsubmissionmu/forms/Telecom.aspx?lang=e Select the word Tariff from the drop down list. Add the following in Subject Line File Number # 8740-B2-200904989 - Bell Canada - TN 7181 and make your thoughts known! I think that there are 30 days from March 27 to respond to the petition to cabinet. | Date: Mon, 30 Mar 2009 13:56:30 -0400 | From: Bill Sandiford b...@telnetcommunications.com | To: 'asterisk Mailing' asterisk@uc.org, 'b...@taug.ca' b...@taug.ca | Subject: [on-asterisk] Various regulatory issues that could affect you | | All: | | I apologize in advance for the cross-posting between lists, however I | felt this was of significant enough importance to justify it (hopefully | Simon and Dave agree with my assessment). I suggest that any discussion | of this topic take place on the general list | (asterisk@uc.orgmailto:asterisk@uc.org). | | There is a lot of regulatory proceedings going on right now. These | proceedings are actually what is taking up the most of my time these | days. The most important of which are the following | | http://www.ic.gc.ca/eic/site/smt-gst.nsf/eng/sf09316.html | | We are opposing the Bell and Telus petitions and supporting the | Allstream petition. | | Bell is basically petitioning Governor in Council to overturn CRTC | decisions that they don't like. There are huge implications to this, | especially in these few cases. | | But more importantly Bell has recently filed a tariff notice for what is | known as UBB (Usage based billing) | | http://www.crtc.gc.ca/8740/eng/2009/b2_7181.htm | | UBB will force wholesalers (like Telnet, Teksavvy, and others) to | enforce 60 GB/month caps on our subscribers and bill for usage above | 60GB. Bell is asking the CRTC for this to be effective May 31, 2009. | As you will see from the link above, Telnet (in association with some | others) have already filed a letter with the CRTC asking that interim | approval of this tariff not be granted as we plan to intervene and fight | this tariff. | | Have a read of the files in the links. | | Regards, | | Bill Sandiford | Telnet Communications | 905-674-2000 x100 | b...@telnetcommunications.commailto:b...@telnetcommunications.com | | IMPORTANT NOTICE: This message is intended only for the use of the | individual or entity to which it is addressed, and may contain | information that is privileged, confidential and exempt from disclosure | under applicable law. If the reader of this message is not the intended | recipient, you are hereby notified that any dissemination, distribution | or copying of this communication is strictly prohibited. If you have | received this communication in error, please notify the sender | immediately by email and delete the message. Thank you. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] alarm systems
I think you mean Local Link line which is a POTS service offered by Bell. It is an analog line, it is not digital. -Original Message- From: Dean Yorke [mailto:dean.yo...@xyc.ca] Sent: Thursday, March 05, 2009 5:47 PM To: Andre Courchesne Cc: asterisk Mailing Subject: Re: [on-asterisk] alarm systems Sorry, local loop line is a service that has call features like call forwarding and call display and caller id. but it is different that the service for a standard bell line like home. I think it has a little digital service. Thanks On 5-Mar-09, at 4:33 PM, Andre Courchesne wrote: Hi Dean, I think local loop refers to a dry-loop so you do not have a dialtone on those lines thus they can only be used for DSL services... Dean Yorke wrote: Hi All, I know that this might be a little off topic but... Wondering if someone can help me understand the difference between an analogue line and a local loop line from bell. we have a couple pieces of equipment, (pitney bowes mail machine and personal install alarm system) that are having issues communicating on these lines. Thanks - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] alarm systems
Dean: The only difference is the features that are included. It is still a regular analogue line served off the same equipment at the Bell CO as every other line. Regards, Bill -Original Message- From: Dean Yorke [mailto:dean.yo...@xyc.ca] Sent: Thursday, March 05, 2009 10:20 PM To: Bill Sandiford Cc: 'Andre Courchesne'; 'asterisk Mailing' Subject: Re: [on-asterisk] alarm systems yes this is what i am talking about? is there any difference with a standard analogue line? also, anyone have any luck with a postal machine working over ata and voip lines? thanks for the responses! On 5-Mar-09, at 8:26 PM, Bill Sandiford wrote: I think you mean Local Link line which is a POTS service offered by Bell. It is an analog line, it is not digital. -Original Message- From: Dean Yorke [mailto:dean.yo...@xyc.ca] Sent: Thursday, March 05, 2009 5:47 PM To: Andre Courchesne Cc: asterisk Mailing Subject: Re: [on-asterisk] alarm systems Sorry, local loop line is a service that has call features like call forwarding and call display and caller id. but it is different that the service for a standard bell line like home. I think it has a little digital service. Thanks On 5-Mar-09, at 4:33 PM, Andre Courchesne wrote: Hi Dean, I think local loop refers to a dry-loop so you do not have a dialtone on those lines thus they can only be used for DSL services... Dean Yorke wrote: Hi All, I know that this might be a little off topic but... Wondering if someone can help me understand the difference between an analogue line and a local loop line from bell. we have a couple pieces of equipment, (pitney bowes mail machine and personal install alarm system) that are having issues communicating on these lines. Thanks - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Comwave Callback and Similar
Bruce: I certainly can't speak to what Comwave is using specifically, but there are many platforms out there that can offer this type of service (including Asterisk). Bill -Original Message- From: Bruce N [mailto:het...@hotmail.com] Sent: Sunday, March 01, 2009 6:40 PM To: asterisk Mailing Subject: [on-asterisk] Comwave Callback and Similar Hello Everyone, In the past few months Comwave has been advertising about a sort of CALLBACK/DISA combination for $20/month. Can anyone speculate as to what sort of equipment/solution they use to achieve their goal? Even if they sign up 100k clients that is still a lot of calls knowing every call is actually two calls going through their system and then bridged. Do they use an Asterisk SER farm? Do they use some propriety equipment like VPS, MERA, etc...? Listed: http://www.comwave.net/mobile/ Thanks, Bruce _ Share photos with friends on Windows Live Messenger http://go.microsoft.com/?linkid=9650734 - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Conference bridge
I have a spare dual-quad core blade that we aren't using (yet) in our datacenter. I could make it available for this test...but I will need it back in 3-4 weeks -Original Message- From: Simon P. Ditner [mailto:si...@uc.org] Sent: Friday, February 27, 2009 12:17 PM To: asterisk@uc.org Subject: RE: [on-asterisk] Conference bridge Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield
RE: [on-asterisk] Conference bridge
Jim and all: We are in the process of deploying one of these in our datacenter. http://www.supermicro.com/products/SuperBlade/datacenterblade/ we are using this blade with dual quad core xeons, 12GB ram and a 640GB RAID5 array. http://www.supermicro.com/products/SuperBlade/module/SBI-7425C-S3.cfm We have the hardware but its going to be another month or so before we have the staff resources to deploy. So if you want 1 blade for each of FreeSwitch, Asterisk and YATE, I can make available but it has to get done in the next month. Bill -Original Message- From: Jim Van Meggelen [mailto:j...@vanmeggelen.ca] Sent: Friday, February 27, 2009 4:17 PM To: Simon P. Ditner Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge You know what? That's a fantastic idea! We should set up a basic idea of what we want to test (keep it simple), and the get together and have a 'mix-off' between YATE, FS and Asterisk and see which one can handle the most channels before it starts having issues. We probably wouldn't be able to be too formal in our methodology, so some purists might scoff, but if we did the same sort of test on the same sort of system, it'd at the least give some sort of benchmark as to where the performance differences lie. I would love to test this on the following: - Atom (I have a lab box for that) - Core 2 Duo - Core 2 Quad - Xeon - Dual Xeon Sorting out performance differences between the various projects would be interesting enough, but what would also be neat to discover is whether, say a core2 quad is that much better than a core2 duo, and whether a xeon is far better, or only a little better. I have an intel Atom box that I can bring to the party. Can't currently help with the others, though. Jim Simon P. Ditner wrote: Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk]
RE: [on-asterisk] CISCO Asterisk 729 Issue
Syed: I have quite a bit of experience with Cisco AS5300. I'm should be in the office today until around 4pm although I am trying to leave early for thank you Hallmark for finding another reason to spend money on my wife day (aka Valentine's). Give me a call if you want to compare notes. Oshawa: 905-674-2000 ext 100 Toronto: 416-613-2000 ext 100 Regards, Bill -Original Message- From: Syed Zia [mailto:syed...@rogers.com] Sent: Saturday, February 14, 2009 11:17 AM To: asterisk@uc.org Subject: [on-asterisk] CISCO Asterisk 729 Issue FOlks, I need some help from you while I am trying to get my CISCO gear working with Asterisk. I have CISCO AS5300 and I am trying to send calls to my Asterisk Server by using CISCO's version of g729r8. Some how when I do that My Asterisk Box says No compatible codec's, not accepting this offer!. When I use 711 on both sides (ulaw), It works like a charm, No Issues. I did some googling and saw some recommendations like try to enforce g729r8 bytes 40 at the CISCO end but NO Luck. I am using Digium licensed version of G729. Any help or advise would be much appreciated Syed Zia - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] clogging the tubes
I've been too busy watching old episodes of Polka-Dot Door to respond. :)
-Original Message-
From: Simon P. Ditner [mailto:si...@uc.org]
Sent: Wednesday, February 11, 2009 8:26 PM
To: asterisk@uc.org
Subject: Re: [on-asterisk] clogging the tubes
No responses? I'm so disenchanted.
On Tue, 10 Feb 2009, Simon P. Ditner wrote:
For your entertainment...
Usage:
./inject.sh 4165551212 'http://www.youtube.com/watch?v=ghWpAgFfgV0'
extensions.conf:
[inject-wav]
exten = s,1,Answer()
exten = s,n,Wait(1)
exten = s,n,Playback(${WAV})
exten = s,n,System(rm ${WAV}.wav)
[outbound]
exten = X.,1,Dial(SIP/tr...@${exten})
inject.sh:
#!/bin/sh
FILE=`tempfile`
EXTEN=$1
URL=$2
RES=`lynx -source $URL | grep fullscreenUrl | cut -d'' -f4,8`
mplayer http://www.youtube.com/get_video?$RES; -dumpaudio -dumpfile
$FILE-t.mp3
mpg123 -w $FILE-t.wav $FILE-t.mp3
sox $FILE-t.wav -c 1 -r 8000 -w $FILE.wav
# Remove temporary files
rm $FILE $FILE-t.wav $FILE-t.mp3
echo Channel: Local/$ex...@outbound
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: inject-wav
Extension: s
Priority: 1
Set: WAV=$FILE
/var/spool/asterisk/outgoing/inject-$EXTEN.call
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RE: [on-asterisk] Confused about Sendmail and FreePBX
Henry: Seeing as the 22,000 messages are waiting to be sent (assuming they haven't bounced yet) they will be in /var/spool/mqueue or /var/spool/client-mqueue If they have already bounced you will likely find them in /var/spool/mail like Mike suggested or perhaps in /var/mail The other option is to get an ISP that doesn't block your port 25 (cough cough) :) Bill From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: Monday, February 09, 2009 10:53 AM To: asterisk@uc.org Subject: Re: [on-asterisk] Confused about Sendmail and FreePBX Henry, Sendmail is a bit of a bear when it comes to configuring. If your up to the task you need to configure it to use either SASL or TLS but it's not easy. This will allow you to us either port 465 (SSL) or 567 (TLS) which are ports that are not blocked for sending email, but require that you use one of the two authentication methods. The easier rote like was previously suggested would be to use your ISP's smtp to handle the relay. Now in regards to the 22000 emails, if you really want to delete them all, login to your server, then cd /var/spool/mail, then ls -l will show you the email files for the users on the system. Your probably going to have just two accounts, root asterisk, if you sudo rm root you will delete them. Now if you get your smtp outbound corrected, you can also make the emails for these accounts get automatically forwarded to a real account so they don't build up. Alter the file /etc/aliases by adding asterisk: root root: aster...@voip-pbx.camailto:aster...@voip-pbx.ca Then save the file and recompile the database issuing: newaliases now email sent to those accounts should relay to the destination email account Mike Henry L.Coleman wrote: I suspect that many of the TAUG members have had trouble getting sendmail to work from their * boxes and now I am one of them. Here are the facts: I am running PBX in a Flash Freepbx and Asterisk 1.4 Part of this distro includes webmin and Sendmail I want to send vmail attachments and error messages via email I only need to send messages out. My ISP blocks port 25 I have no fixed IP address but use a dynamic DNS (dyndns)service for this * server I have tried to get this up and running but with no luck. I don't have the time or the inclination to learn about email servers like sendmail Can anybody help me with the configuration ? Also I need to be able to delete 22,000 emails that have been placed on the server by warnings and errors etc. (before my HD fills up :) Henry = Henry L.Coleman [www.VoIP-PBX.cahttp://www.VoIP-PBX.ca] Tel: 647-723-5160 Ext.203 = - To unsubscribe, e-mail: asterisk-unsubscr...@uc.orgmailto:asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.orgmailto:asterisk-h...@uc.org -- Mike Ashton Quality Track Intl CTO Ph: 647-724-3500 x 301 Cell: 416-527-4995 Fax:416-352-6043 QTI CONFIDENTIAL AND PROPRIETARY INFORMATION The contents of this material are confidential and proprietary to Quality Track International, Inc. and may not be reproduced, disclosed, distributed or used without the express permission of an authorized representative of QTI. Use for any purpose or in any manner other than that expressly authorized is prohibited. If you have received this communication in error, please immediately delete it and all copies, and promptly notify the sender.
[on-asterisk] RE: Nortel Files for Bankruptcy, Victim of Falling Sales
It's about time -Original Message- From: Chuck Mariotti [mailto:cmario...@xunity.com] Sent: Wednesday, January 14, 2009 11:47 AM Cc: asterisk Mailing Subject: [on-asterisk] Nortel Files for Bankruptcy, Victim of Falling Sales I'm sure most of you knew this was coming. But it's implications are significant. http://www.bloomberg.com/apps/news?pid=20601082sid=a8E1VQ1yRq8krefer=canada - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Anyone has used PIKA
You should talk to Stephan at Unlimitel. I believe he is the authorized distributor for these units in Canada. -Original Message- From: Yajie Si [mailto:siya...@gmail.com] Sent: Wednesday, January 14, 2009 9:32 PM To: asterisk@uc.org Subject: [on-asterisk] Anyone has used PIKA Anyone uses this PIKA WARPAsterisk appliance, sounds interesting, better than Asterisk Digium Applicance http://www.pikatechnologies.com/ anyone can share his experience? Thanks! -- Yajie(Roger) Si - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] 23 Analogue lines
http://www.williamsglobal.com/ - Original Message - From: Alex Kink alexk...@gmail.com To: Philip Mullis philip.mul...@syx.ca; asterisk@uc.org Sent: Thursday, December 11, 2008 7:24 PM Subject: Re: [on-asterisk] 23 Analogue lines Philip, who is this Williams you speak of? :) Regards, Alex Kink On Wed, Dec 10, 2008 at 5:51 PM, Philip Mullis philip.mul...@syx.ca wrote: You can always check with Williams, I know when I was there a few years back they had that sorta stuff in the warehouse from time to time as they bought alot of it on trade, you might luck out there with something. Phil Jim Van Meggelen wrote: Andrew Kohlsmith (lists) wrote: On December 10, 2008 01:59:01 pm Jim Van Meggelen wrote: Most channel banks you'll be able to pick up used will be FXS, but what is needed is FXO. You can order the FXO cards for the channel bank, but by the time you're done that you're going to be into some coin. I've ebayed every single piece of my telecom equipment, including fat-ass MaxTNTs for terminating DS3s. Ebay rules for getting used telecom equipment, no doubt about it, and Adtran AT750s are really common. What's not so common is the FXO cards for them (although I suppose with patience they can be found too). - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] 23 Analogue lines
If you've been to Vegas lately you'll see their slot machines and video poker all over the place too. :) http://www.wms.com http://en.wikipedia.org/wiki/Williams_Electronics - Original Message - From: Jim Van Meggelen j...@vanmeggelen.ca To: Alex Kink alexk...@gmail.com Cc: Philip Mullis philip.mul...@syx.ca; asterisk@uc.org Sent: Thursday, December 11, 2008 7:36 PM Subject: Re: [on-asterisk] 23 Analogue lines Don't they make pinball machines? Jim Alex Kink wrote: Philip, who is this Williams you speak of? :) Regards, Alex Kink On Wed, Dec 10, 2008 at 5:51 PM, Philip Mullis philip.mul...@syx.ca wrote: You can always check with Williams, I know when I was there a few years back they had that sorta stuff in the warehouse from time to time as they bought alot of it on trade, you might luck out there with something. Phil Jim Van Meggelen wrote: Andrew Kohlsmith (lists) wrote: On December 10, 2008 01:59:01 pm Jim Van Meggelen wrote: Most channel banks you'll be able to pick up used will be FXS, but what is needed is FXO. You can order the FXO cards for the channel bank, but by the time you're done that you're going to be into some coin. I've ebayed every single piece of my telecom equipment, including fat-ass MaxTNTs for terminating DS3s. Ebay rules for getting used telecom equipment, no doubt about it, and Adtran AT750s are really common. What's not so common is the FXO cards for them (although I suppose with patience they can be found too). - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org -- -- Jim Van Meggelen j...@vanmeggelen.ca http://www.oreillynet.com/pub/au/2177 A child is the ultimate startup, and I have three. This makes me rich. Guy Kawasaki -- - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
[on-asterisk] FBI Warning
Anyone else see this? http://www.fiercevoip.com/story/fbi-issues-voip-security-warning-asterisk-which-version/2008-12-07?utm_medium=nlutm_source=internalcmp-id=EMC-NL-FVdest=FV Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
Re: [on-asterisk] Hardware specs for 40 seats
I can tell you that we have had good experiences with Linksys overall, but absolutely HORRIBLE experiences with the SRW248G4P. The switch has QoS capabilities and we found that if we tried to turn any of it on the switch became very unstable. We found the only way the switch was usable was at factory default settings without taking advantage of any of its features. At this point, it was a dumb PoE switch only. - Original Message - From: Bruce Nik [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Cc: asterisk Mailing asterisk@uc.org Sent: Monday, December 08, 2008 9:43 PM Subject: RE: [on-asterisk] Hardware specs for 40 seats Thanks for the input guys. Could this setup work without a hiccup: - High end server with lots of RAM and CPU power + 5 Sata 750GB HDDs. Trying RAID 5 with this. Client needs to record every single call. 8 hours shift * 20 sets. - Sangoma 2XT1 card with echo cancel = 2 PRI - Linksys SRW248G4P (48 port poe; 15.5W for 24 port POE or 7.7W if all 48 ports used) - 48 Sets of Aastra 51i(apparently they are POE ready and don't need an AC Adapter) - 24 gauge unshielded of 25 pair cable 1- Not sure how many of the 25 pair should run. Would POE be fine over this type of cable? (interference of unshielded cable with POE). How 48 Aastra 51i power up using the forementioned Linksys switch? 2- Computers connect to the phone. Internet usage is not really high. 3- No cables bought so far. I am flexible with dropping as many cables as I want. What type of cable should be used? Thanks, Date: Sun, 7 Dec 2008 23:17:46 -0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] CC: [EMAIL PROTECTED]; asterisk@uc.org Subject: RE: [on-asterisk] Hardware specs for 40 seats let us not forget the fxs gateways that can take so much of this networking skill set pain away :) and they are getting cheap now. -Original Message- From: TianLun Song [mailto:[EMAIL PROTECTED] Sent: Sun 12/7/2008 11:14 PM To: Jim Van Meggelen Cc: Bruce Nik; asterisk Mailing Subject: Re: [on-asterisk] Hardware specs for 40 seats if you are not good understanding of QoS, Cisco develops a tool called AutoQoS on its mainframe switch which is to facilitate QoS setup. it is a very simple command under interface level. however, the switch from CISCO is very pricey. another opition for you is linksys and it works well if you know QoS and VLAN fully. On Sun, Dec 7, 2008 at 9:15 PM, Jim Van Meggelen [EMAIL PROTECTED] wrote: In my experience it is still very common to run voice and data on separate networks. In theory, this is not necessary since a managed switch should be able to deliver PoE, VLan, and QoS over a single wire, but when all the costs are added up, it can still be cheaper to pull two (or even three) separate runs to each desk, and then run it all on cheaper network switches (since you don't need the Vlan and QoS for the VoIP anymore). The cost of managing a network is often not taken into consideration, and it can be costly, especially if you need specific expertise (such as Cisco certs). The advantage of an unmanaged network is that the skill level needed to understand it and work with it is lower. A manged network is much more powerful, but not as many have the skill to work with it, so for smaller companies this can often end up being a burden. I think if cost was no object and I had a team of solid network specialists on call, I'd probably buy into the Cisco vision. There is something very compelling about all that control of each and every network port. But realistically, I would recommend pulling two cables to every desk (it's not much more expensive, especially if you shop around and get competitive quotes from the cablers). That'll give you lots of flexibility back in the network room to decide how complex you want your LAN to be, and it's amazing how fast too many cables turns into too few. Pull the cable when you have the budget for it. Good chance you'll need it sooner than you think. JimBruce Nik wrote: Hi guys, I am just drawing diagrams here and trying to find out what equipment I need for a 30-50 seats. I have seen few people post their full solutions (hardware) as to what they used. I can't seem to find it anymore on the net. Anyone has a URL for few solutions that are already deployed and work just fine? Everything can be done from scratch, including cabling and that is what throws me off because I have so many options in terms of cabeling. What do you suggest I should do? Thanks,Bruce _ -- -- Jim Van Meggelen [EMAIL PROTECTED] http://www.oreillynet.com/pub/au/2177 A child is the ultimate startup, and I have three. This makes me rich. Guy Kawasaki -- - To unsubscribe, e-mail: [EMAIL
Re: [on-asterisk] Hardware specs for 40 seats
I know someone who is running 48 Linksys SPA942 phones off of that switch. - Original Message - From: Bruce Nik [EMAIL PROTECTED] To: [EMAIL PROTECTED]; asterisk Mailing asterisk@uc.org Sent: Monday, December 08, 2008 10:07 PM Subject: RE: [on-asterisk] Hardware specs for 40 seats Hello, That would the cabling from the server room to the wall jacks and to the cubicles. The place doesn't have walls. I can do the cabling too. So, very flexible. Do you know if all the 48 phones will fire up? Linksys Switch has 15.5W per port if only 24 port is used. It has 7.7W if all ports are used for POE. Thanks Date: Mon, 8 Dec 2008 21:47:57 -0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [on-asterisk] Hardware specs for 40 seats Hi Bruce, Yes the 51i is PoE Why 24 gauge unshielded of 25 pair cable what are you planning to use this for? Bruce Nik wrote: Thanks for the input guys. Could this setup work without a hiccup: - High end server with lots of RAM and CPU power + 5 Sata 750GB HDDs. Trying RAID 5 with this. Client needs to record every single call. 8 hours shift * 20 sets. - Sangoma 2XT1 card with echo cancel = 2 PRI - Linksys SRW248G4P (48 port poe; 15.5W for 24 port POE or 7.7W if all 48 ports used) - 48 Sets of Aastra 51i (apparently they are POE ready and don't need an AC Adapter) - 24 gauge unshielded of 25 pair cable 1- Not sure how many of the 25 pair should run. Would POE be fine over this type of cable? (interference of unshielded cable with POE). How 48 Aastra 51i power up using the forementioned Linksys switch? 2- Computers connect to the phone. Internet usage is not really high. 3- No cables bought so far. I am flexible with dropping as many cables as I want. What type of cable should be used?Thanks, Date: Sun, 7 Dec 2008 23:17:46 -0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] CC: [EMAIL PROTECTED]; asterisk@uc.org Subject: RE: [on-asterisk] Hardware specs for 40 seats let us not forget the fxs gateways that can take so much of this networking skill set pain away :) and they are getting cheap now. -Original Message- From: TianLun Song [mailto:[EMAIL PROTECTED] Sent: Sun 12/7/2008 11:14 PM To: Jim Van Meggelen Cc: Bruce Nik; asterisk Mailing Subject: Re: [on-asterisk] Hardware specs for 40 seats if you are not good understanding of QoS, Cisco develops a tool called AutoQoS on its mainframe switch which is to facilitate QoS setup. it is a very simple command under interface level. however, the switch from CISCO is very pricey. another opition for you is linksys and it works well if you know QoS and VLAN fully. On Sun, Dec 7, 2008 at 9:15 PM, Jim Van Meggelen [EMAIL PROTECTED] wrote: In my e xperience it is still very common to run voice and data on separate networks. In theory, this is not necessary since a managed switch should be able to deliver PoE, VLan, and QoS over a single wire, but when all the costs are added up, it can still be cheaper to pull two (or even three) separate runs to each desk, and then run it all on cheaper network switches (since you don't need the Vlan and QoS for the VoIP anymore). The cost of managing a network is often not taken into consideration, and it can be costly, especially if you need specific expertise (such as Cisco certs). The advantage of an unmanaged network is that the skill level needed to understand it and work with it is lower. A manged network is much more powerful, but not as many have the skill to work with it, so for smaller companies this can often end up being a burden. I think if cost was no object and I had a team of solid network specialists on call, I'd probably buy into the Cisco vision. There is something very compelling about all that control of each and every network port. But realistically, I would recommend pulling two cables to every desk (it's not much more expensive, especially if you shop around and get competitive quotes from the cablers). That'll give you lots of flexibility back in the network room to decide how complex you want your LAN to be, and it's amazing how fast too many cables turns into too few. Pull the cable when you have the budget for it. Good chance you'll need it sooner than you think. JimBruce Nik wrote: Hi guys, I am just drawing diagrams here and trying to find out what equipment I need for a 30-50 seats. I have seen few people post their full solutions (hardware) as to what they used. I can't seem to find it anymore on the net. Anyone has a URL for few solutions that are already deployed and work just fine? Everything can be done from scr atch, including cabling and that is what throws me off because I have so many options in terms of cabeling. What do you suggest I should do? Thanks,Bruce _ -- -- Jim Van Meggelen [EMAIL PROTECTED]
Re: [on-asterisk] Underground conduit design (for fiber optics) help
Steven: We have some underground fiber projects going on right now. If you send me the sketches I would be happy to look at them for you. For your conduit, are you using 10ft sections that you are gluing together or are you using the HDPE style that comes in rolls? How many conduits are you running? I'd run at least 2 even if you don't think that you will use it in the future. The cost of conduit is minimal compared to the cost of trenching. Conduits (especially when built with 10 ft sections) can easily become blocked. 800ft is not a long run at all. Unless there is a really good reason for pull boxes or flush-to-grade enclosures I would just do it in one long shot. Reasons for putting a pull box in would be multiple bends greater than 90 degrees, or possibility of splicing (teeing off) to another building in the future. Are the buildings on the same property? If not, there may be permitting issues as well. Regards, Bill - Original Message - From: Steven McCann [EMAIL PROTECTED] To: asterisk@uc.org Sent: Sunday, November 23, 2008 3:06 PM Subject: [on-asterisk] Underground conduit design (for fiber optics) help Hello, I'm running some conduit between buildings (to later install fiber) and there's a few items I'm not sure about. Does anyone have any knowledge or experience with running conduit underground? Here's a few specifics of the plan: -we're using 2 PVC DUCT conduit -the buildings we are connection are around 600-800ft apart. The plan is to put a pull station every 300ft or so. The runs are fairly straight. -we're burying the conduit about 36, so its below the frost line -considering using underground pulling boxes in the middle of runs and wall/post mounted enclosures at buildings (which will also store slack fiber) There's a few details I'm not sure about: -what type of boxes are available to mount outside buildings where the conduit comes up to - the conduit is going to be used to run indoor/outdoor fiber. I would like to have maybe 50ft or so excess fiber coiled up outside the building in the box where the conduit comes up out of the ground -any specifics on mounting the box on buildings to allow easy pulling -suggestions for methods to setup pulling stations in the ground to allow easy pulling and excess fiber storage. I can sketch out the site plan and other plans if it would be helpful. Thanks, Steven --- cell: 416-618-1354 - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Linksys SPA 942
Does it say Checking DNS. It probably can't get an answer from the DNS servers that it is configured to use Instead of host names try IP addresses in the phone. For example on the Line1 tab...for Telnet customers that normally have sip1.telnetcommunications.com I would tell them to use 74.51.34.37 . If he is behind NAT he should enable the STUN features of that phone on the SIP tab as well. If you are not sure of the IP address...just ping the hostname and you'll get it. - Original Message - From: Henry L.Coleman [EMAIL PROTECTED] To: asterisk@uc.org Sent: Friday, October 31, 2008 8:52 PM Subject: [on-asterisk] Linksys SPA 942 Hi guys, can anyone help ? A friend of mine purchased an SPA 942 for use in California using a Unlimitel supplied voip line. It worked fine in Toronto but when he tried to use it in California it can't find a DNS server. I have walked him through the set up but it still doesn't work His ITSP is Verizon. What little hair I have is going fast! Would be grateful for any help Henry L.Coleman CEO [VoIP-PBX] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Linksys SPA 942
Nah...its probably getting DHCP...when SPA-942s don't get DHCP the screen statys stuck on Initializing Network. - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: asterisk@uc.org Sent: Friday, October 31, 2008 8:53 PM Subject: RE: [on-asterisk] Linksys SPA 942 Do you mean it doesn't get DHCP? -- Nabeel Jafferali X2 Networks -Original Message- From: Apache [mailto:[EMAIL PROTECTED] On Behalf Of Henry L.Coleman Sent: October-31-08 8:52 PM To: asterisk@uc.org Subject: [on-asterisk] Linksys SPA 942 Hi guys, can anyone help ? A friend of mine purchased an SPA 942 for use in California using a Unlimitel supplied voip line. It worked fine in Toronto but when he tried to use it in California it can't find a DNS server. I have walked him through the set up but it still doesn't work His ITSP is Verizon. What little hair I have is going fast! Would be grateful for any help Henry L.Coleman CEO [VoIP-PBX] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] speaker for oct 29th.
Very interested in YATEVERY ! - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: Simon P. Ditner [EMAIL PROTECTED] Cc: asterisk@uc.org; [EMAIL PROTECTED] Sent: Tuesday, October 07, 2008 12:00 PM Subject: Re: [on-asterisk] speaker for oct 29th. Simon P. Ditner wrote: Any volunteers for the meeting at the end of this month, or perhaps the month after? I could prolly hack something together on YATE, if folks are interested. -- -- Jim Van Meggelen [EMAIL PROTECTED] http://www.oreillynet.com/pub/au/2177 A child is the ultimate startup, and I have three. This makes me rich. Guy Kawasaki -- - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] The Open Source HomeAutomation Project
Are you looking for this one? - Original Message - From: Bruce Nik [EMAIL PROTECTED] To: asterisk@uc.org Sent: Monday, September 08, 2008 11:11 PM Subject: [on-asterisk] The Open Source HomeAutomation Project Hello Everyone, Sorry, if I am a bit off topic but about a year ago someone posted a URL about an open source HomeAutomation system that was pretty elaborate. I believe it took advantage of Asterisk too but I could be wrong. I just can't seem to find with a google search or in the posts at taug.ca. It wasn't linuxmce project. I would really appreciate it if you remember and post the site here. Hint: I believe they used a wiki to explain the system at the time. Thanks, Bruce _ ---BeginMessage--- My main complaint about MythTV (and satellite/digital cable systems) is that the interface that I've used (for myth, 2006) becomes useless once you have any substantial amount of content. When all I have is a remote control with a directional pad and a few numbers, I need some more creative navigation solutions than paging through long lists like on DirecTV/Dish/ExpressVu. We run into the exact same problem when you have any substantial content behind an IVR. There's a massive pile of content, and my access to what (I think) I want is bottlenecked by these crummy interfaces. What could we do differently? On Nov 26, 2007 9:57 AM, John Van Ostrand [EMAIL PROTECTED] wrote: Duane wrote: Came across this link tonight, and I have no use for it at the moment, but it looks interesting none the less. http://linuxmce.com/ I'm installing one now for home. It's an open source fork of Pluto Home. Basically the intent of LinuxMCE is to be a residential server interfacing with home automation, security and A/V hardware. Asterisk and MythTV are also added. And it can act as a network server as well. It seems very ambitious in what it wants to do and promises effortless installation and configuration. I'm giving a presentation on it at the KWLUG (http://kwlug.org) in January. It's going to be focused more on the MythTV side of things. -- *John Van Ostrand* *Net Direct Inc.* CTO, co-CEO 564 Weber St. N. Unit 12map http://maps.google.ca/maps?q=564+Weber+Street+North+Unit+12,+Waterloo,+ON+N2L+5C6,+Canadall=43.494599,-80.548222spn=0.038450,0.073956iwloc=Ahl=en Waterloo, ON N2L 5C6 [EMAIL PROTECTED] Ph: 866-883-1172ext.5102 *Linux Solutions / IBM Hardware*Fx: 519-883-8533 -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | The Toronto Asterisk Users Group | Join the discussion group by visiting http://taug.ca - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] ---End Message--- - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] dryloop dsl 911 service
Umm I can tell you that all Bell Dryloops that we have ever sold have always had dialtone on them. The dialtone isn't usable for much other than dialling the ANAC number for identification, but dialtone is absolutely present (at least at the beginning). I'm not sure about 911. - Original Message - From: Liviu Toma [EMAIL PROTECTED] To: TAUG asterisk@uc.org Sent: Thursday, June 26, 2008 8:58 AM Subject: Re: [on-asterisk] dryloop dsl 911 service It doesn't. There's no dialtone at all. On Thu, Jun 26, 2008 at 8:47 AM, Simon P. Ditner [EMAIL PROTECTED] wrote: Does dryloop DSL have 911 service on it, or is the line truly phone service free? - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Hoping for some discussion on Local Number Portability (and toll free too, for that matter)
Comments inline below - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: asterisk@uc.org Sent: Sunday, April 06, 2008 11:50 AM Subject: [on-asterisk] Hoping for some discussion on Local Number Portability (and toll free too, for that matter) Folks, I'd be grateful if we could have a discussion about number portability. I can probably answer most any question about LNP. We are part of the CLNPC (Canadian Local Number Portability Consortium) I have a couple of specific things I'd like to know, but I'd also like to have an idea of what knowledge we have in our group, and get it into a discussion so that it will be in the list archives. My questions are: 1) If I have an 800 number, do I own that? If so, can I simply move it to another carrier? Currently my 800 number comes in on my analog circuits, and Bell is quite happy to gouge me for things like CallerID (and a recently added $8/mo charge that does not seem to relate to anything). I'd much prefer to assign it to a VoIP service. No idea what my rights are with respect to this, nor what I might expect to pay. I don't want to find out one day that because I moved my number to wherever, I no longer have any rights to it (assuming I have any rights to it now). You do have rights to it. You can move it to any toll free carrier you wish by simply filling out a form called a RESPORG. The new TF Carrier should be able to provide you with one. As far as the pricing goes, its pretty common knowledge that Bell charges $8 per month just for the privilege of having a TF number and then charge you for every incoming call. Some TF carriers don't charge a monthly fee at all and just charge for the minutes (like us for exampleshamless plug..sorry) 2) I have a customer who has locations all over the 905 area code North of Toronto. They would like to have all their numbers come in on their PRI in Aurora, and then distrubute to each site via VoIP. I believe that local numbers cannot move out of whatever exchange they are assigned to, but I have seen some pretty creative things done, and there is such a thing as a foreign exchange circuit, so I am curious what is actually possible. Creativity is the key. Lots can be done. They can ask for a wide area PRI. The trick is going to be finding a provider that has LNP coverage in all of the areas that they have local numbers on. Most providers can only provide LNP in the main area (like Newmarket for example) but can't provide LNP for the outlying areas. The following URL will give you a list of the rate centres in which we can do LNP either now or in the very near future (June 1) http://www.localcallingguide.com/lca_prefix.php?npa=nxx=x=ocn=190eregion=lata=switch=pastdays=0nextdays=0 There are a few that we can do that aren't on this list as well. We will be putting a more complete list on our website soon. Any other comments regarding number portability in Canada (whether related to my questions or not) or valuable resources would be most welcome. Jim -- Jim Van Meggelen Core Telecom Innovations [EMAIL PROTECTED] www.coretel.ca 416-425-6111 x6001 877-CORETEL x6001 (Canada) www.oreillynet.com/pub/au/2177 http://downloads.oreilly.com/books/9780596510480.pdf - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] DTMF RFC283
Is the carrier capable of doing a packet capture on their end to make sure they are getting the DTMF from you correctly?. It would be interesting to find out where the DTMF is breaking? Are you able to do one from your box (or a mirrored switchport if you don't like running packet traces on a production box) to make sure you are transmitting it correctly to that carrier? - Original Message - From: Reza M. Reza [EMAIL PROTECTED] To: asterisk@uc.org Sent: Tuesday, March 25, 2008 12:36 PM Subject: [on-asterisk] DTMF RFC283 I have an issue with a carrier and hoping for some feedback. My carrier has been amazing and responsive, but as in any business, there are minor things that arise from time to time The issue I am facing is that RFC2833 does not work on certain numbers, specially government IVRs. This is on Asterisk 1.2.9.1 I then tested the exact same thing RFC2833 on the government numbers in question from my Asterisk 1.4.18It works perfectly. HOWEVER!!! 1. When I use other CDN trunks from 1.2.9.1 Asterisk with RFC2833 it works FINE. 2. When I use Unlimitel it is fine. 3. When I use a dozen other carriers (American), its ALL fine. 4. Regardless of 1.2.9.1 or 1.4.18 RFC2833 works from ALL other carriers. SO. IF ALL other carriers are passing DTMF perfectly from my 1.2.9.1 except for the one in question, then would it be fair to say the problem is the specific carrier which is consistent in failure to transmit DTMF to those government numbers? Or are there other possible scenarios that I may be missing out? The quick fix here is to divert outgoing calls from 1.2.9.1 to the trunks that are 100% on DTMF via RFC2833. Any suggestions is welcome! Best, Reza. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
[on-asterisk] Bell strike likely imminent
If you, like myself, do a lot of business with Bell Canada on a regular basis you may want to have a read of the following. Bell's new release: http://www.newswire.ca/en/releases/archive/March2008/17/c4822.html The union's new release: http://www.newswire.ca/en/releases/archive/March2008/17/c4803.html The Union's memo to members: http://www.cep25.com/documents/Vote%20results.pdf The issues from a union perspective: http://www.cep25.com/documents/bell_summary_offer_080131.pdf So it looks like we are headed for some tough weeks ahead. Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
Re: [on-asterisk] Bell blocking SIP traffic again?
Is this Bell Sympatico DSL or the wirelees unplugged version? - Original Message - From: Dave Bour [EMAIL PROTECTED] To: 'TAUG Asterisk Mailing List' asterisk@uc.org Sent: Friday, March 07, 2008 5:16 PM Subject: [on-asterisk] Bell blocking SIP traffic again? I've one client that started having problems 2:36pm today. Disconnect/reconnect the modem, through 7 cycles found 5 of the IP addresses provided were blocking, 2 were successful. That said, I also found an excessive number of dropped connections on their system today. Most under 2 minutes. Anyone else seeing anything like that? Dave Bour Desktop Solution Center 905.381.0077 X501 [EMAIL PROTECTED] For people who just want IT to work Business http://www.desktopsolutioncenter.ca Personal http://www.davebour.com - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
[on-asterisk] Limits on scheduled outgoing calls
Hi All: We are working on a project for a client that requires a large number of scheduled outgoing calls for message playback. There is about 1000 calls scheduled at a time. We want to limit the simultaneous number of outgoing calls to 10. The following URL recommends a method to limit outgoing calls by using an external script to limit the number of files in the /var/spool/asterisk/outgoing directory and moving more files in as necessary http://www.voip-info.org/wiki-Asterisk+auto-dial+out#HowtoscheduleaCallintheFuture Has anyone been successful in writing a script for this purpose that they would be willing to share? Does anyone know of any other methods for doing this? (putting a call-limit on the sip peer doesn't really cut it). Regards, Bill
Re: [on-asterisk] Survey: what are people's experience with various routers?
Hands down the best device that we have found so far for small offices (especially if the customer wants support) is the Juniper SSG5 http://www.juniper.net/products_and_services/firewall_slash_ipsec_vpn/ssg_5_slash_ssg_20/ There are many different models of the SSG5 including models that have serial backup WAN, integrated V.92 modem backup WAN, or ISDN backup WAN. There is a non-wireless and a wireless a/b/g model. For example: The SSG-SB which is the entry level model with serial backup and no wireless is around $599 The SSG-SB-W-US which is the entry level model with serial backup and 802.11a/b/g wireless is around $799 These aren't the cheapest routers on the block by any stretch of the imagination but they work well and the support is incredible. We have several large deployments of customers on hosted PBX that use them and they are rock solid. Regards, Bill - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: asterisk@uc.org Sent: Tuesday, January 29, 2008 8:31 AM Subject: [on-asterisk] Survey: what are people's experience with various routers? Folks, Lately it seems that the GNU/Linux firewall, iptables, is emerging as one of the best. Even many hardware products are based on it. If cost were no object, and you needed to buy a firewall (that of course had to do a good job with VoIP), what would be on your wish list? What would you avoid like a plague? (ask Leif about SonicWall) Any thoughts and opinions are most welcome. Regards, Jim -- Jim Van Meggelen Core Telecom Innovations [EMAIL PROTECTED] www.coretel.ca 416-425-6111 x6001 877-CORETEL x6001 (Canada) www.oreillynet.com/pub/au/2177 http://downloads.oreilly.com/books/9780596510480.pdf - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
[on-asterisk] Re: [biz] Asterisk PA/Paging
Keith: We have several deployments that make use of loud paging from Asterisk. We always use the Viking products driven by the FXO port of a Linksys 3102. It works great. In addition, the same device can be used to provide loud ringing over the PA speakers by using the FXS port from the same Linksys 3102. They also have a music input so a radio or other music device can play music into the warehouse that gets automatically muted when the paging or loud ringing cuts in. Viking makes at least 2 products that will work in this environment. CPA-7B http://www.vikingelectronics.com/products/view_product.php?pid=29 PA-30 http://www.vikingelectronics.com/products/view_product.php?pid=330 The CPA-7B also comes with a small horn style speaker. We have found that for most of our installs the CPA-7B just didn't have enough power and we ended up using the PA-30. If you end up using these I have some dialplan configs that I can send you (its pretty straight forward though) Regards, Bill - Original Message - From: Keith Major | Aquarius Telecom Inc. [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 18, 2008 3:37 PM Subject: [biz] Asterisk PA/Paging Hi All, We have a prospect in a warehouse that would require paging to be done on a PA type of system through the phone, i.e. someone answers an inbound call, puts the person on hold, enters a feature code and says, Keith line 1 please, Keith line 1. Similar to what you would find at the grocery store. We would like to deploy an Asterisk solution for them but, this is one requirement that is needed. If anyone would like to share their thoughts on this it would be appreciated. Thanks, Keith Major Business Development Manager Aquarius Telecom Inc. HYPERLINK http://www.aquariustel.com/www.aquariustel.com 416.800.0833 ext. 2005 ___ This e-mail may be privileged and/or confidential, and the sender does not waive any related rights and obligations. Any distribution, use or copying of this e-mail or the information it contains by other than an intended recipient is unauthorized. If you received this e-mail in error, please advise me (by return e-mail or otherwise) immediately. Ce courrier électronique est confidentiel et protégé. L'expéditeur ne renonce pas aux droits et obligations qui s'y rapportent. Toute diffusion, utilisation ou copie de ce message ou des renseignements qu'il contient par une personne autre que le (les) destinataire(s) désigné(s) est interdite. Si vous recevez ce courrier électronique par erreur, veuillez m'en aviser immédiatement, par retour de courrier électronique ou par un autre moyen. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.6 - Release Date: 17/01/2008 12:00 AM - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Suck it up - No Caller ID - comments?
Syd: CallerID Name is a funny animal. I'm not 100% sure that your provider's explanation makes a lot of sense to me because most of the signals sent are digital but allow me to give this little backgrounder on CallerID Name First of all it is important to note that we are very lucky in Canada. The way the CallerID name system works here is much different than it is in the US. In Canada CallerID Name information is set with every call. In the US it is stored in central databases. Unless you have access to those databases (or your provider will update it for you) you are out of luck when it comes to changing the CallerID Name. Ok, back to Canada. Chances are that your provider is using ISDN-PRI trunks to their provider. There are 2 ways that CallerID Name can be passed with ISDN-PRI. The first method, commonly used by Nortel DMS switches (think Bell, Allstream), is by passing the name as a parameter in the call-setup message. The other way, commonly used by Lucent 5ESS switches (think Telus), is by passing the name as a Facility IE usually in the progress message. So the problem is *likely* that your provider's equipment is either configured for the wrong switchtype, doesn't have the CallerID name configured correctly, and/or doesn't support sending CallerID name in the same fashion as their provider. Asterisk supports sending CallerID Name using both methods. The method can be set via the facilityenable parameter in zapata.conf. But this is only if your connections to your provider are PRI. So the reality is that there is nothing that you are doing wrong with your CallerID Name via SIP. The problem is just that your provider and their provider are likely either configured wrong or incompatible with each other when it comes to CallerID Name Regards, Bill PS - HAPPY NEW YEAR TO ALL !!! - Original Message - From: Syd Carter [EMAIL PROTECTED] To: TAUG asterisk@uc.org Sent: Sunday, December 30, 2007 1:39 PM Subject: [on-asterisk] Suck it up - No Caller ID - comments? Hey all. My caller id _name_ info is not coming across. According to my provider, we send callerID correctly however the ISP's have old equipment that don't understand the new digital signal therefore they can't understand the call display. I've check my sip dialog and know that I'm sending name and number. I never had this problem with my previous voip service provider. Can anyone provide me with additional insight into the new digital signal versus the old one? Thanks.. Syd - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
[on-asterisk] Exit Codes
Does anyone know where I can find a list of Asterisk's exit codes? In my lab I have a box that I'm putting through some testing and it Asterisk keeps exiting with code 135. I've also (although rare) seen it exit with code 138. I don't see anything in the logs at all to indicate why. Regards, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
Re: [on-asterisk] Scary Call from Bell Muscle Men...
Reza: I think when most people refer to de-regulation in the industry they are referring to the CRTC telling the incumbents what they can and cannot sell you and at what price. The only reason most ITSPs are in business (apart from intelligence and business savvy) is because the CRTC decided (with a few small exceptions ie 911, etc) not to regulate VoIP (unless it was being sold by an ILEC) Bill - Original Message - From: Reza - Asterisk Enthusiast [EMAIL PROTECTED] To: John Lange [EMAIL PROTECTED]; asterisk@uc.org Sent: Friday, December 07, 2007 12:08 PM Subject: Re: [on-asterisk] Scary Call from Bell Muscle Men... John: Just a small bit of clarification; the industry has not been deregulated. Far from it. Can you please clarify? Are you 100% sure the industry has not be de-regulated? There was a time when one could not be an ITSP or small companies could not provide phone services with their own equipment. No one could provide phone services other than the ILEC or the CLEC. Speaking with the guys at Industry Canada / CRTC -- anyone can apply for licenses these days and when you pay the right fee with the correct paper work submission, obtaining a license is relatively straight forward. Becoming a CLEC is a different story. If the industry was not de-regulated... then as per your claim, we are doing illegal business. Are you talking about De-regulation as in regulating prices. If that's the case I completely agree with you. When I am using the term De-regulation, I mean anyone can be a telephone company or an ITSP these days regardless of them being an ILEC and CLEC. Please provide your thoughts and feedback on this when you have a chance. Best, Reza. - Original Message - From: John Lange [EMAIL PROTECTED] To: asterisk@uc.org Sent: Friday, December 07, 2007 9:52 AM Subject: Re: [on-asterisk] Scary Call from Bell Muscle Men... On Thu, 2007-12-06 at 21:56 -0500, Reza - Asterisk Enthusiast wrote: Thanks to the de-regulation of the industry... its provided the smaller guys for greater opportunities... and if the big guns like Bell, Allstream and others are not quick to change -- they will loose millions in revenue. They already are. Just a small bit of clarification; the industry has not been deregulated. Far from it. In almost every area of Canada the rates the ILEC charges is set artificially high to encourage competition. Up until recently the ILECs were not allowed to compete on price for any wireline service. However, with the introduction and subsequent success of cable VoIP (Rogers, Shaw, etc), most residential service in Canada where cable VoIP exists is now free from economic (price) regulation. Note that this applies to residential, not business service. In the above described scenario; Bell is the ILEC, Allstream is the CLEC. Bell is price regulated; Allstream is not. Regards, -- John Lange President Canadian Association of Voice Over IP Service Providers. 1-866-940-CAVP (2287) - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Scary Call from Bell Muscle Men...
Oktime to get technical It's not exactly accurate to say they use Packet Cable. They use a variety of technologies from Cable Labs, the proprietors of the PacketCable project. Basically the Rogers Home Phone adapter is a normal DOCSIS cable modem with a NCS VoIP ATA built in. For those of you that don't know, NCS is a bunch of extensions to MGCP designed for and by the cable industry (kind of like H.248). So what you basically have is a modem/ATA combo not much different than the DSL modem/ATA combos that are available (think Zoom x5v, Zhone 6238 or Thomson 780wl) As far as the sharing comment goes, well this is true however it is a very unfair comment to make. Rogers puts all of their DOCSIS based Home Phone devices on a specific cable frequency (channel) dedicated for home phone. So the packets are on a shared channel, but the only other thing on that channel are other Home Phone units. The potential impact of this on voice quality is minimal to none. As far as the anyone can listen to your call comment, once again possible but highly unlikely. In order to do this someone would have to meet all of the following requirements. 1) connect to the coax cable IN YOUR neighborhood, either at their home or at a tap (not very hard to do) 2) convert a normal cable modem into a bridge and configure it to listen to the channel (frequency) that Rogers has dedicated for home phone in your neighborhood (very hard) or buy some *VERY EXPENSIVE* RF test gear 3) run a protocol analyzer like wireshark on the data coming off of the 'cable modem bridge' or RF test gear (easy) So, it is possible to listen to a neighbor's Rogers Home phone calls, but very unlikely. And like Phil said, all you need to listen to someones Bell line is a $3 phone, and a pair of alligator clips. If you live in the same neighborhood and have a BIX tool and a 5/8 socket you could even divert their line to you house...permanently. - Original Message - From: [EMAIL PROTECTED] To: asterisk@uc.org Sent: Thursday, December 06, 2007 8:38 PM Subject: Re: [on-asterisk] Scary Call from Bell Muscle Men... On 6 Dec 2007 at 19:29, Michael Richardson wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Are you sure? I know that Rogers' has said this, but it is my understanding that they haven't actually implemented this. And, even if it is on the same frequencies as the Internet, you can't see other people's traffic on cable without non-standard equipment. (And I don't mean a Linux CD). Positive, they use Packet Cable technology Unless you've got some serious test/monitoring equipment, you ain't seeing anything on the drop into your house I think the biggest concern I have is the battery life on the phone line is fairly short, in case of an emergency (ala major blackout). jp The battery life of the phone is supposedly 8 hours, not sure jp about battery life of equipment shelters between you and the jp cable head-end And... Bell lost power to many of their LEC's after about 20 hours during the ice-storm, and had to get generators out to there. Anything that is connected directly to a CO shouldn't ever loose power unless the loop is cut Although nowadays they have optical quite close to the kerb JP - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Scary Call from Bell Muscle Men...
Hi Terry: You have a bunch of good questions. I would say that any type of VoIP, either DSL or Cable (don't let Rogers make you believe that Home Phone isn't VoIP...it is), is far more secure than a traditional Bell land line. For telephone banking, it is much easier to intercept someones DTMF (touch tone) digits on an analog phone line than it is on a digital one. They don't need to be outside your house or on a pole. They just need access to any point on your loop (house, pole, crossbox, splice can). These can be located anywhere between your house and the nearest Bell CO or Remote. And the person wouldn't necessarily have to stay their either. The bottom line is that the last mile of any connection is the most vulnerable to intercept. Oh and don't let you think that Rogers is very safe either. Just last week I was in the Oshawa Bell CO working on a customer trouble with a Bell technician. We were looking for a spare pair and were surprised to find dial tone (and then someone talking) on a pair. It turned out that the Rogers installer didn't disconnect the copper line outside the house and when the customer plugged their Home Phone adapter into a normal jack (so they could get dialtone through the house) they were driving the copper pair all the way back to the CO...in analog. As far as security goes, your calls are going to be pretty secure. Not as secure as they could be if you were able to use IPSEC or some other encryption method (Kevin will probably chime in with zRTP), but I don't know any VoIP providers (including ourselves) that offer this. It requires decryption on the far end...so your iTSP or ISP has to be in on it It is pretty hard for someone on the internet to intercept your packets unless they have access to the core routers of the ISPs that are transiting the data. Not impossible, but highly unlikely. Yes, there is an initiative for a quasi-ssh like VoIP. Its called zRTP and comes from Phil Zimmerman, the same guy that invented PGP encryption that is used in email. It is very new however and not very mature yet. The person on the far end would have to have the same setup. Bill - Original Message - From: terry D. Cudney [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: asterisk@uc.org Sent: Thursday, December 06, 2007 9:37 PM Subject: Re: [on-asterisk] Scary Call from Bell Muscle Men... Hello Philip and everyone, This thread interests me, since, just over a month ago, I terminated my relationship with Bell when we relocated. Question is: How secure is, or how can one make voip secure? i.e. Is Telephone Banking vulnerable over voip? For our residential phone we are now using acanac and lesnet over aDSL with dry-loop, asterisk 1.4.11 on my linux box here at home with a couple of Aastra SIP phones and a Linksys 3102 to the analog phones in the house. Cost is much lower than Bell's minimal service and we now have all the bells and whistles that Bell charges an arm and a leg for, at no extra. I could go on about Bell's bumbling monopolistic methods, like repeated phone calls to try to convince me to come back to Bell Sympatico for adsl, billing me for a month after the service was terminated, when I call them to try to straighten it out I get someone in India who can hardly speak English who tells me that I have Bell Expressview on my account and that the account was never terminated/settled... I tell them there is no Expressview on the account and the account was terminated when I left that address... 20 minutes of elevator music later I get dead air... (Boy am I glad I no longer have any affiliation with Bell!!!) Sorry about that rant... Question, if you've read this far, is related to the comments below about security on a voip call: Philip Mullis wrote: Anyone with enough skills can listen to your calls on the rogers network, but that would imply they also have access to the switching fabric in which your calls go through., also if you want to be super secure, get a voip provider that does ipsec connections from you to them ,this will ensure very high security. Not using Rogers, how secure are calls using adsl/asterisk to a itsp like acanac or lesnet? Everytime I think I'm getting a handle on networking/routing/dns/traffic-shaping/etc something new turns up. Like ipsec. How do I determine if, or if not, ipsec is being used? Can I set it up on my end unilaterally? or must it be a provision from the itsp? Bell copper... m what can i say here... anyone with a 3$ phone from wallmart, plyers and aligator clips can listen in on your call :/ True, but he'd have to be outside my house or on a pole somewhere, right? With IP isn't it possible for anyone on the internet, savvy enough to do it, to intercept packets and monitor calls/data transmissions from the comfort of his living room? Unless we are using some kind of security or tunneling protocol, or maybe IPSEC? What would be the
Re: [on-asterisk] Scary Call from Bell Muscle Men...
Wellthey terminate all their Home Phone to a bunch of BTX4KI believe the BTX4K is capable of encryption, but I've been told by a former Rogers employee that he is pretty sure they aren't using it (it doesn't scale well) I just found this on the BTX4K. It says it signals TGCP (which is the PacketCable extensions to MGCP) http://www.nuera.com/products/ORCA_BTX-4K.cfm - Original Message - From: Michael Richardson [EMAIL PROTECTED] To: Bill Sandiford [EMAIL PROTECTED] Cc: asterisk@uc.org Sent: Thursday, December 06, 2007 11:17 PM Subject: Re: [on-asterisk] Scary Call from Bell Muscle Men... -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bill == Bill Sandiford [EMAIL PROTECTED] writes: Bill Basically the Rogers Home Phone adapter is a normal DOCSIS Bill cable modem with a NCS VoIP ATA built in. For those of you Bill that don't know, NCS is a bunch of extensions to MGCP designed Bill for and by the cable industry (kind of like H.248). So what Bill you basically have is a modem/ATA combo not much different Bill than the DSL modem/ATA combos that are available (think Zoom Bill x5v, Zhone 6238 or Thomson 780wl) So Rogers actually deployed NCS then? Yes, it's MGCP, which is *TOTALLY* insecure unless you encrypt. Fortunately NCS does encrypt using IPsec, keyed using a kerberos based method. (NCS infrastructure is *TESTED* using a variation of Openswan) Bill As far as the anyone can listen to your call comment, once Bill again possible but highly unlikely. In order to do this Bill someone would have to meet all of the following requirements. ... or... crack the system which is used by the COPS to do eavesdropping, which actually is not very secure - -- ]Bear: Me, I'm just the shape of a bear. | firewalls [ ] Michael Richardson,Xelerance Corporation, Ottawa, ON|net architect[ ] [EMAIL PROTECTED] http://www.sandelman.ottawa.on.ca/mcr/ |device driver[ ] panic(Just another Debian GNU/Linux using, kernel hacking, security guy); [ -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Finger me for keys iQEVAwUBR1jJSYCLcPvd0N1lAQKr4AgAsKZ8QTtQxxLFV2v40Jrk/wBBLwi6yAMG vAb6zcOVUVdGUb/hSR/QXzEezFklE2IOCL/xq4OZ+BQGR2K5AdLJ7JWfslij+VUx Fy3G8bES61nJ8FnI5SCh52sp2wyVaqkMC9RLGqajcvQ1ELfyUNTh8GjeNQDPpdxF nUZYsZNsqpPaZsCY2MBiVqebsZEiS8CsrpX/tCOPWX04jFtzHfRzpUEc3llt/BCt ewaTLCQ8iiADLiYLLnwCit/aQrxw5Mw04wiSkIK0CDe4GNKLY26cKwYR1DePmAMg 68K+IgoZHNgdrTxo8SLyBAGjUCXqCXAyINcxx9+BgFZdxtY8GkVeEw== =TeR4 -END PGP SIGNATURE- - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] asterisk web GUI's et-al
Stephan: I think what Henry meant to say is we can port numbers from some rate centres that others can't. For example we can port from: Ajax-Pickering, ON Ancaster, ON Aurora, ON Barrie, ON Beaverton, ON Belleville, ON Bobcaygeon, ON Bowmanville, ON Brampton, ON Brantford Brockville, ON Brooklin, ON Burlington, ON Calgary, AB Cannington, ON Chatham, ON Clarkson, ON Coboconk, ON Cooksville, ON Dundas, ON Edmonton, AB Fenelon Falls, ON Georgetown, ON Guelph, ON Halifax, NS Hamilton, ON Hull, QC Keswick, ON Kingston, ON Kitchener, ON Lachine, QC Lindsay, ON London, ON Longueuil, QC Malton, ON Maple, ON Markham, ON Milton, ON Mississauga, ON Montreal, QC New Westminster, BC Newmarket, ON North Vancouver, BC Oak Ridges, ON Oakville, ON Oshawa, ON Ottawa, ON Peterborough, ON Pointe-Claire, QC Pont-Viau, QC Port Credit, ON Port Perry, ON Prescott, ON Richmond Hill, ON Richmond, BC Roxboro, QC Sarnia, ON South Pickering, ON St. Catharines-Thorold, ON Ste-Geneviève, QC St-Lambert, QC Stoney Creek, ON Streetsville, ON Sunderland, ON Surrey, BC Tecumseh, ON Thornhill, ON Toronto, ON Trenton, ON Unionville, ON Vancouver, BC Victoria, BC Waterloo, ON Welland, ON Whalley, BC Whitby, ON Windsor, ON Winnipeg, MB Woodbridge, ON and coming in late January 2008 to: Apsley, ON Bethany, ON Bolton, ON Campbellford, ON Cobourg, ON King City, ON Kleinburg, ON Lakefield, ON Omemee, ON Palgrave, ON Port Hope, ON Queensville, ON Schomberg, ON Stouffville, ON Sutton, ON Uxbridge, ON Regards, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] - Original Message - From: Stephan Monette [EMAIL PROTECTED] To: asterisk@uc.org Sent: Thursday, November 15, 2007 11:16 AM Subject: Re: [on-asterisk] asterisk web GUI's et-al Henry, You are saying we can't port numbers? You're buying me a beer at the next meeting! We can port numbers from any of the following rate centers: Toronto, Edmonton, Calgary, Vancouver, Winnipeg, Oshawa, Brampton, Mississauga (Cooksville) Brantford, Hamilton, Kitchener, Windsor, Ottawa, Kingston, Montreal, Quebec City. You can apply for LNP online: https://secure.unlimitel.ca/forms/validatelnp.php Thanks. Stephan Monette Unlimitel Inc. Tel.: 1 (877) 464-6638, x221 Henry L.Coleman wrote: Unlimitel, most of Canada is on-net But Telnet if you need to port a number (905,416 etc.) Telnet are a CLEC so they can do some things that Unlimitel can't - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] asterisk web GUI's et-al
Henry: How long is that challenge open for? We have notified Bell just yesterday that we intend to interconnect in the following rate centres effective February 15, 2008: Alliston, ON Barrie, ON Beeton, ON Bradford, ON Collingwood, ON Orillia, ON Penetanguishene, ON Wasaga Beach, ON But knowing how Bell works, it will likely be sometime in the first few weeks of March before we are ready there. Regards, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] - Original Message - From: Henry L.Coleman [EMAIL PROTECTED] To: asterisk@uc.org Sent: Thursday, November 15, 2007 11:41 AM Subject: Re: [on-asterisk] asterisk web GUI's et-al Opps.. Sorry, I was referencing 905 430/428 (Whitby) where you don't have a POP. Anyway, there's a beer for anyone who can port an Orillia number (705-326-), now there a challenge !. My humble apologies (roughly equal to a thousand pardons) Henry Stephan Monette Henry, You are saying we can't port numbers? You're buying me a beer at the next meeting! We can port numbers from any of the following rate centers: Toronto, Edmonton, Calgary, Vancouver, Winnipeg, Oshawa, Brampton, Mississauga (Cooksville) Brantford, Hamilton, Kitchener, Windsor, Ottawa, Kingston, Montreal, Quebec City. You can apply for LNP online: https://secure.unlimitel.ca/forms/validatelnp.php Thanks. Stephan Monette Unlimitel Inc. Tel.: 1 (877) 464-6638, x221 Henry L.Coleman wrote: Unlimitel, most of Canada is on-net But Telnet if you need to port a number (905,416 etc.) Telnet are a CLEC so they can do some things that Unlimitel can't - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] asterisk web GUI's et-al
Henry: Look into Elastix. www.elastix.org Regards, Bill - Original Message - From: Henry L.Coleman [EMAIL PROTECTED] To: asterisk@uc.org Sent: Tuesday, November 13, 2007 10:05 PM Subject: [on-asterisk] asterisk web GUI's et-al For those people who (like me) are getting fed up with Trixbox I am doing some evaluations of various other management tools (GPL) I plan to just give you a quick evaluation just to wet your appetite If you have an interest in these evaluations give me some feedback on what you think. The first one tried is CENTPBX This uses the latest Freepbx GUI framework and can be generated for * 1.2 or 1.4 Has everything Trixbox has in terms of features, plus a very useful Night Answer feature (*28) So far I like it because it runs 1.4 and has the complete feature set Rating (4 asterisks out of 5) -- Henry - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Speech Recogniton Reliability
I assume you just got the targeted email (aka SPAM) from Digium as well. - Original Message - From: Reza - Asterisk Enthusiast [EMAIL PROTECTED] To: TAUG asterisk@uc.org Sent: Wednesday, October 17, 2007 3:52 PM Subject: [on-asterisk] Speech Recogniton Reliability Has any one implemented SPEECH RECOGNITION in Asterisk in a production environment? If not... in a test environment, what are your findings, when the same English sentence is spoken by a dozen different people with a dozen different accents/dialects in English. How accurate is the speech engine with wrong grammar spoken. How many simultaneous recognitions it can do on a P4, GHz dual core on 2 Gigs of Ram. etc. Any bench mark is appreciated. Best, Reza. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] CNAM - Caller ID Name
1. Why dynamic CNAM issue is a big deal? From a technical point of view it isn't usually a big deal at all. From a liability point of view it could be. Some larger carriers are worried about the implications if scam artists started setting their CNAM to things like TORONTO POLICE and started calling people fraudlently. My argument to the carriers has always been Well you let us set the number and nothing is stopping the scam artists from setting the number to match the number of the police, so whats the big deal? The usual answer I get is a confused look and a shrug of the shoulders. 2. Why does the Name info have to be dug out each time from a shared database (if its shared) based on the number (some carriers do this) by the carrier's lookup servers and then spit out to its subscribers? Well, in Canada it isn't done that way at all. The CNAM database method is how the CNAM info is looed up in the good old U S A. In the US carriers access a shard CNAM database and query the database for the CNAM for every call. In Canada we use a totally different method (described below) 3. Why can't the NAME be sent with the NUMBER info from one carrier to the other? In Canada it is The CNAM in Canada is passed from carrier to carrier as part of the messaging that take place when a call is initiated. 4. or Is transmitting NAME with the NUMBER simply a limitation on the switch the telcos carry? Most of the time if its not working its as a result of interoperability between varying switch types. If the trunks between carriers are SS7 then the CNAM will most certainly work properly. However must people using Asterisk don't have SS7, they have PRIs. The ISDN signalling between different switches expect CNAM to be passed in different ways. For example, a Nortel DMS (Bell Canada), expects the CNAM to be sent as part of the initial Q.931 call setup message. Lucent 5ESS switches (Telus, Rogers) expect it to be part of a subsequent progress message as a Facility IE. So, in order to get CNAM to work on your PRI, its important to know what method your carrier uses. I can tell you that Asterisk definately supports the Facility IE message because I was the one that initiated the feature request and did the testing for it back in April 2005. http://bugs.digium.com/view.php?id=4046 To enable CNAM by Facility IE set facilityenable=yes in zapata.conf. If you have SIP trunking, you are at the mercy of your carriers switch capabilites and their PSTN connectivity arrangments. Hope this helps clear things up for you !!! Regards, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
Re: [on-asterisk] Using Cable as a failover from DSL
Henry: I 100% agree with the rational for what you are doing, however I wouldn't do this on the Asterisk box. I would look into a hardware appliance for this. Something like a HotBrick LB-2 would work great for this and they are fairly cheap as well. http://www.redundantinternet.com/en/LB-2.html Regards, Bill - Original Message - From: Henry L.Coleman [EMAIL PROTECTED] To: asterisk@uc.org Sent: Tuesday, August 28, 2007 1:16 PM Subject: [on-asterisk] Using Cable as a failover from DSL We all are aware that the uptime of a single DSL or Cable connection to an ITSP is less than that of analog lines or a T1 so I'm putting out the idea of using DSL and Cable where the chances of both failing at the same time are very low. In normal operation the DSL would handle Voice (Asterisk) while the cable would handle any data traffic. So here's the plan (remember I'm not a network guy) A cron job on the Asterisk server continually pings an external server should the Ping take more than a two seconds then * IP address is change to share the Data network until service is resumed. Is this a plan or am I just whistling Dixie TTFN Henry -- Henry L. Coleman. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Using Cable as a failover from DSL
John: Good question, I have the answer for you. You can tell the box to prefer one connection over the other for certain routes (based on source or destination IP address) during normal operation (both connections up). If the preferred connection fails, then the traffic falls over to the other connection. Bill - Original Message - From: John Van Ostrand [EMAIL PROTECTED] To: Bill Sandiford [EMAIL PROTECTED] Sent: Tuesday, August 28, 2007 2:17 PM Subject: Re: [on-asterisk] Using Cable as a failover from DSL Bill Sandiford wrote: Henry: I 100% agree with the rational for what you are doing, however I wouldn't do this on the Asterisk box. I would look into a hardware appliance for this. Something like a HotBrick LB-2 would work great for this and they are fairly cheap as well. http://www.redundantinternet.com/en/LB-2.html This might not perform as you expect. It's probably used to load balance things like HTTP sessions where the from IP address is less of an issue. How would one control which WAN would be used to register SIP? Then when a WAN connection fails Asterisk would still need to re-register. Could you register via both? When I've done this in the past, I've established a VPN over the Internet (I controlled both ends) and when the IP changed the VPN would re-establish and all the existing connections would lurch forward and continue one. Still in your case calls would go dead until the LAN re-established. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
[on-asterisk] FALL VON 2007 - Time to hit road again
Hello All: Back in June I organized a small little road trip to the NXTcomm Telecom tradeshow in Chicago. Reza and I told the group about the show at June meeting. It was amazing. Well the time has come to start planning again. This time the show is the Fall VON (Voice on the Net) show which is being held in Boston from October 29th-November 1st. The exhibit hall is open on the 30th and 31st. The VON show is mostly focused on VoIP (NXTcomm was all aspects of Telecom). Here is the link to the info site on the show: http://www.von.com/2007/boston/web/ I have attended this show for the past 3 years. Although the show seemed to have shrunk a little bit last year, I have just looked at the floorplan online and it looks like this years show is back up to its previous size. Here is the link to the floorplan http://fp1.a2zinc.net/clients/fpvon/fall2007/public/fp.aspx Of specific interest to our group is that this years Fall VON is hosting Digium/Asterisk World. http://www.digiumasteriskworld.com/2007/boston/web/ From the floorplan it looks like there is at least 4 booths setup for Digium/Asterisk World, 20ft x 20ft, 20ft x 30ft, 75ft x 30ft, and a monstrous 95ft x 40ft. Thats a total of over 7000 sq ft of space dedicated Digium/Asterisk World. Not to mention the fact that once again most of my major vendors are there so I should be able to get us in to some good parties in the evenings. So, calling all TAUG members. Who is up for a road trip to Boston? Let me know asap as I would like to get this one planned well in advance. We could have a couple of different options for travel. We could carpool in a van again (hopefully multiple vans if we have enough people). It is about a 9 hour drive. Also, as much as I enjoyed the 8.5 hour drive to/from Chicago with Norm, Todd and Reza (cough cough), JetBlue has really cheap non-stop flights from Buffalo to Boston for about 129 USD taxes included (about $140 CDN these days). So, I'm somewhat leaning towards flying down, probably early on the 30th (there is a 7:00am flight out of Buffalo that gets to Boston at 8:25am) and flying home early on the 1st. These cheap flights will fill up fast, so we will need to book soon !!! Any takers? Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
Re: [on-asterisk] Symetrical DSL or Cable
I don't benefit from them either, but I know both of the owners personally. They have a very good fixed wireless network that does symmetrical service. From what I understand their network covers most of the west end of the city very well and some areas of downtown (if you have line of sight) - Original Message - From: Dave Donovan [EMAIL PROTECTED] To: asterisk@uc.org Sent: Wednesday, August 01, 2007 3:03 PM Subject: Re: [on-asterisk] Symetrical DSL or Cable On 8/1/07, Henry L.Coleman [EMAIL PROTECTED] wrote: So my question is..does anyone know of a service that can give me symetrical upload/download speed at a reasonable cost $200 per month. ? This is a borderline biz question, but I'm willing to bend the rules and play along :-) It's tough to find a symmetrical service in that price range. The last price I heard for an SDSL line was in the $500 range. I was talking with a company yesterday that does fixed wireless service. I haven't had it implemented so I can't speak to their services but I think they might be able to meet your requirement depending on site location relative to their POP. I spoke with a guy named Mark at Internet Access Solutions. www.iasl.com Bill Sandiford put me onto them at the last TAUG meeting. It might be worth a call. Disclosure: I don't work for these guys or benefit from their sales. I'm just a potential customer. Dave - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] VOIP Bandwidth - Calculation
As Phil described, its much more than 8 kbps, and the lower the frame rate, the more bandwidth it takes due to the packet overhead. While turning up our softswitch, we did a bunch of testing with different codecs at different sample/frame rates on ethernet. Here are my *real world* results: g711 (ulaw) 10ms frame ~= 126.5 kbps 20ms frame ~= 95.2 kbps 30ms frame ~= 84.7 kbps 40ms frame ~= 79.6 kbps g729a (8 kbps) 10ms frame ~= 70.5 kbps 20ms frame ~= 39.2 kbps 30ms frame ~= 28.8 kbps 40ms frame ~= 23.5 kbps We didn't test GSM. So, most ADSL providers (including us) offer sync rates at speeds up to 800 kbps, however that speed is if the traffic is ATM (which it isn't), so by the time you add IP/Ethernet overhead, the 800 kbps speed usually gives you a payload somewhere around 670-690 kbps. So divide that by one of the numbers above and round down and you will have your *theoretical* max simultaneous calls assuming the connection is only used for VoIP. Now throw in a few users and their email and web traffic and the number starts to go down. Hope this helps you out. Regards, Bill - Original Message - From: Aloysius Thevarajah Lloyd [EMAIL PROTECTED] To: TAUG asterisk@uc.org Sent: Monday, July 23, 2007 9:52 PM Subject: [on-asterisk] VOIP Bandwidth - Calculation Hi, For a 10 SIP /g729 simultaneous use. What is the Best internet connection recommended. DSL or ADSL What is the upload and download speed required? How to calculate the the band width? ( g729a - 8 kbit/s 10 ms frames) if u can guide me greatly appreciated. Thank you -- Lloyd - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] VOIP Bandwidth - Calculation
After my last post, I did a bit of googling and found this calculator. It seems to be giving numbers that are very close to our real world tests. http://blog.asteriskguide.com/bandcalc/bandcalc.php Regards, Bill - Original Message - From: Bill Sandiford [EMAIL PROTECTED] To: Aloysius Thevarajah Lloyd [EMAIL PROTECTED]; TAUG asterisk@uc.org Sent: Monday, July 23, 2007 11:15 PM Subject: Re: [on-asterisk] VOIP Bandwidth - Calculation As Phil described, its much more than 8 kbps, and the lower the frame rate, the more bandwidth it takes due to the packet overhead. While turning up our softswitch, we did a bunch of testing with different codecs at different sample/frame rates on ethernet. Here are my *real world* results: g711 (ulaw) 10ms frame ~= 126.5 kbps 20ms frame ~= 95.2 kbps 30ms frame ~= 84.7 kbps 40ms frame ~= 79.6 kbps g729a (8 kbps) 10ms frame ~= 70.5 kbps 20ms frame ~= 39.2 kbps 30ms frame ~= 28.8 kbps 40ms frame ~= 23.5 kbps We didn't test GSM. So, most ADSL providers (including us) offer sync rates at speeds up to 800 kbps, however that speed is if the traffic is ATM (which it isn't), so by the time you add IP/Ethernet overhead, the 800 kbps speed usually gives you a payload somewhere around 670-690 kbps. So divide that by one of the numbers above and round down and you will have your *theoretical* max simultaneous calls assuming the connection is only used for VoIP. Now throw in a few users and their email and web traffic and the number starts to go down. Hope this helps you out. Regards, Bill - Original Message - From: Aloysius Thevarajah Lloyd [EMAIL PROTECTED] To: TAUG asterisk@uc.org Sent: Monday, July 23, 2007 9:52 PM Subject: [on-asterisk] VOIP Bandwidth - Calculation Hi, For a 10 SIP /g729 simultaneous use. What is the Best internet connection recommended. DSL or ADSL What is the upload and download speed required? How to calculate the the band width? ( g729a - 8 kbit/s 10 ms frames) if u can guide me greatly appreciated. Thank you -- Lloyd - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] what's local calling area from 905 430 xxxx ?
Hi Henry: You're the second guy that asked me today. You'll find all these at www.localcallingguide.com . Here they are: Outbound calling from Whitby, ON NPA;NXX;Rate Centre;Region;Plan Type;Call Type;Monthly Limit;Note;Effective 289;220;Whitby;ON; 289;893;Whitby;ON; 905;217;Whitby;ON; 905;430;Whitby;ON; 905;444;Whitby;ON; 905;493;Whitby;ON; 905;556;Whitby;ON; 905;665;Whitby;ON; 905;666;Whitby;ON; 905;668;Whitby;ON; 289;222;Oshawa;ON; 289;224;Ajax-Pickering;ON; 289;225;Port Perry;ON; 289;227;Brooklin;ON; 289;240;Oshawa;ON; 289;314;Ajax-Pickering;ON; 289;315;Ajax-Pickering;ON; 289;316;Oshawa;ON; 289;372;Ajax-Pickering;ON; 289;385;Oshawa;ON; 289;404;Oshawa;ON; 289;688;Oshawa;ON; 289;886;Oshawa;ON; 289;892;Ajax-Pickering;ON; 905;213;Oshawa;ON; 905;215;Oshawa;ON; 905;231;Ajax-Pickering;ON; 905;233;Oshawa;ON; 905;239;Ajax-Pickering;ON; 905;240;Oshawa;ON; 905;242;Oshawa;ON; 905;243;Oshawa;ON; 905;244;Oshawa;ON; 905;245;Oshawa;ON; 905;259;Oshawa;ON; 905;260;Oshawa;ON; 905;261;Oshawa;ON; 905;391;Ajax-Pickering;ON; 905;404;Oshawa;ON; 905;409;Ajax-Pickering;ON; 905;410;Oshawa;ON; 905;423;Ajax-Pickering;ON; 905;424;Ajax-Pickering;ON; 905;425;Brooklin;ON; 905;426;Ajax-Pickering;ON; 905;427;Ajax-Pickering;ON; 905;428;Ajax-Pickering;ON; 905;429;Oshawa;ON; 905;431;Oshawa;ON; 905;432;Oshawa;ON; 905;433;Oshawa;ON; 905;434;Oshawa;ON; 905;435;Oshawa;ON; 905;436;Oshawa;ON; 905;438;Oshawa;ON; 905;439;Oshawa;ON; 905;440;Oshawa;ON; 905;441;Oshawa;ON; 905;442;Oshawa;ON; 905;443;Oshawa;ON; 905;447;Oshawa;ON; 905;448;Oshawa;ON; 905;449;Oshawa;ON; 905;498;Oshawa;ON; 905;550;Ajax-Pickering;ON; 905;571;Oshawa;ON; 905;576;Oshawa;ON; 905;579;Oshawa;ON; 905;619;Ajax-Pickering;ON; 905;620;Brooklin;ON; 905;621;Ajax-Pickering;ON; 905;622;Ajax-Pickering;ON; 905;626;Ajax-Pickering;ON; 905;644;Oshawa;ON; 905;655;Brooklin;ON; 905;674;Oshawa;ON; 905;675;Oshawa;ON; 905;683;Ajax-Pickering;ON; 905;686;Ajax-Pickering;ON; 905;706;Ajax-Pickering;ON; 905;718;Oshawa;ON; 905;720;Oshawa;ON; 905;721;Oshawa;ON; 905;723;Oshawa;ON; 905;725;Oshawa;ON; 905;728;Oshawa;ON; 905;743;Oshawa;ON; 905;744;Ajax-Pickering;ON; 905;767;Ajax-Pickering;ON; 905;809;Oshawa;ON; 905;903;Ajax-Pickering;ON; 905;914;Oshawa;ON; 905;922;Oshawa;ON; 905;924;Oshawa;ON; 905;925;Oshawa;ON; 905;926;Oshawa;ON; 905;982;Port Perry;ON; 905;985;Port Perry;ON; 905;986;Blackstock;ON; 905;995;Ajax-Pickering;ON; 905;999;Ajax-Pickering;ON; Regards,Bill - Original Message - From: Henry L.Coleman [EMAIL PROTECTED] To: TAUG - Tech asterisk@uc.org Sent: Monday, July 16, 2007 5:15 PM Subject: [on-asterisk] what's local calling area from 905 430 ? A quick question... Does any one know what exchanges (rate centers) are local to 905 430 (Whitby) or can suggest a link to find out ? Thanks (going mad trying to find out ) Henry - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
[on-asterisk] Polycom Echo Problems
Hi All: I'm having a problem with a customer that has a bunch of Polcom 501 and 601 sets. They are complaining about echo. Does anyone have some suggestions for some good settings for AEC and AES in sip.cfg for the Polys? Any other suggested settings or changes to the stock sip.cfg? Thanks, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
Re: [on-asterisk] Polycom Echo Problems
Intermittently, both have the echo. They have other other sets in the office not experiencing the echo problem. Their PSTN connection has been properly tuned for echo (via Milliwatt, etc). I'm just looking for a good config for AEC and AES on the Polys (and perhaps gains). By default they are turned off in the stock sip.cfg - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: 'Bill Sandiford' [EMAIL PROTECTED]; asterisk@uc.org Sent: Friday, July 13, 2007 12:30 PM Subject: RE: [on-asterisk] Polycom Echo Problems Couple of things that need to be known: Who has the echo? Your users? or the people who are calling them? How does the system connect to the outside world? (PSTN) Jim -Original Message- From: Bill Sandiford [mailto:[EMAIL PROTECTED] Sent: July 13, 2007 12:23 PM To: asterisk@uc.org Subject: [on-asterisk] Polycom Echo Problems Hi All: I'm having a problem with a customer that has a bunch of Polcom 501 and 601 sets. They are complaining about echo. Does anyone have some suggestions for some good settings for AEC and AES in sip.cfg for the Polys? Any other suggested settings or changes to the stock sip.cfg? Thanks, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you. No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Polycom Echo Problems
John: Thanks for the description, it is very good. Here is the scenario in which this customer is experiencing echo. Polycom --- Asterisk --- Internet --- Polycom also Polycom --- Asterisk --- IX Private Line (delay 10ms) --- Polycom also Polycom --- Asterisk SIP Trunk to carrier --- Carrier's CLASS 5 --- PSTN This particular customer is seeing echo in all three scenarios. Hence the reason I'm looking into the AEC and AES features of the Polycom. Bill - Original Message - From: John Lange [EMAIL PROTECTED] To: asterisk@uc.org Cc: [EMAIL PROTECTED] Sent: Friday, July 13, 2007 4:38 PM Subject: Re: [on-asterisk] Polycom Echo Problems Echo *always* comes from the far end point. The amount a given person perceives the echo is determined by how loud and how delayed the echo is. Volume and delay are influenced by a number of factors along the call path. Echo is a very complex issue but I'll try and give a brief explanation. In the situation where you have a Polycom phone connected to an Asterisk server which is in turn connected to the PSTN talking to a residential wireline customer, e.g.: Polycom -- Asterisk -- PRI -- Wireline Handset If the Polycom customer hears echo it's coming from the wireline handset (and/or the hybrid but I'm trying to keep this example simple). Most consumer handsets just don't care about generating echo because its never been a problem. So echo is normal on all local wireline calls but you don't perceive (hear) echo because the echo is not delayed. Now when you throw Asterisk in the mix the act of encoding and decoding the voice adds delay. This added delay causes you to perceive echo even though the volume of the echo is roughly the same. Technically, to solve echo you fix the endpoint that's causing the echo. But since you can't replace every wireline phone ever made and the telco certainly isn't going to help you that isn't a practical solution. The best you can do is put an echo canceler as close as you can to the endpoint and in this case it's on the Asterisk box. Unfortunately Asterisk's standard built in echo cancelers are crap. They don't even come close to reaching the level of the ITU G.164 standard for echo cancel. That is why you buy cards with add-on hardware echo cancelers that meet the G.164 standard (Sangoma, Digium). Recently you can also buy add-on software echo cancellation from both Sangoma Digium which meet the G.164 standard but beware it exacts a heavy toll on your CPU. But depending on call volume and hardware it might work just fine for you. All of this is a long winded way of saying; you can tune your phone settings until your blue in the face but you won't get rid of the echo. Sorry. So to prove my theory conduct the following tests: Polycom -- Polycom (no echo) Polycom -- Cell phone (no echo) (cell phones do extensive echo cancel) Polycom -- Longdistance (no echo) (telcos do echo cancel on LD) Polycom -- wireline residential (echo!!) That is why your customer reports intermittent echo problems. Hope the above helps you out. John On Fri, 2007-07-13 at 12:51 -0400, Bill Sandiford wrote: Intermittently, both have the echo. They have other other sets in the office not experiencing the echo problem. Their PSTN connection has been properly tuned for echo (via Milliwatt, etc). I'm just looking for a good config for AEC and AES on the Polys (and perhaps gains). By default they are turned off in the stock sip.cfg - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: 'Bill Sandiford' [EMAIL PROTECTED]; asterisk@uc.org Sent: Friday, July 13, 2007 12:30 PM Subject: RE: [on-asterisk] Polycom Echo Problems Couple of things that need to be known: Who has the echo? Your users? or the people who are calling them? How does the system connect to the outside world? (PSTN) Jim -Original Message- From: Bill Sandiford [mailto:[EMAIL PROTECTED] Sent: July 13, 2007 12:23 PM To: asterisk@uc.org Subject: [on-asterisk] Polycom Echo Problems Hi All: I'm having a problem with a customer that has a bunch of Polcom 501 and 601 sets. They are complaining about echo. Does anyone have some suggestions for some good settings for AEC and AES in sip.cfg for the Polys? Any other suggested settings or changes to the stock sip.cfg? Thanks, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you. No virus found in this incoming
Re: [on-asterisk] Polycom Echo Problems
John: Once again, thank you. Your response was again very detailed and helpful. Since my last post (~30 minutes ago), I was able to solve the problems with the third scenario (PSTN). Turned out the carrier had the echo cans turned off for this client on their SIP trunking. Carrier turned on the echo cans and the echo was gone. However, there is still echo in the first 2 scenarios which were both all SIP/VoIP (no PSTN). Its essentially this: Polycom --- Asterisk --- Polycom I tried your suggestion of having the far end mute and you are correct. I am on site and when I place a call I have near-end echo (I hear myself). When the far side mutes, the echo is gone. When the far side unmutes the echo is back. I have turned on the AEC and AES in the Polycom sip.cfg but it hasn't had much of an impact. There are some other settings to do with AEC and AES in the file and hence I am still looking for some recommended settings from anyone that has used them. Regards, Bill - Original Message - From: John Lange [EMAIL PROTECTED] To: Bill Sandiford [EMAIL PROTECTED] Cc: asterisk@uc.org Sent: Friday, July 13, 2007 5:55 PM Subject: Re: [on-asterisk] Polycom Echo Problems In the first two scenarios you describe, you are essentially doing pure SIP to SIP using the Polycoms and that should not cause echo unless echo cancel is disabled on the far end handset. I just now re-read your original posting and indeed you have AEC turned off so that is definitely your problem. One thing to try is to ask the far end to put the phone on mute and see if the echo goes away. If it does, then your echo is being caused by acoustic echo, not an impedance miss-match or other network problem. If you narrow it down to acoustic echo (which is actually the only possibility) then the responsibility of eliminating that echo is squarely with the handset (Polycom) and you'll have to try tuning the related settings. We don't use Polycoms but their conference phones have a reputation for very good echo cancel so I'd be surprised if their handsets weren't equally as good. Mind you I just had a look at their spec sheets and they don't claim G.164 so maybe they don't? If you mute the far end and you still get echo then something else very strange is going on. Like your Asterisk is actually looping the call through the PRI or its traversing an analog circuit or some other thing that shouldn't be happening. In the final scenario (SIP Trunk to carrier), you can't do anything since you don't control the Carrier - PSTN where the echo cancel needs to happen. Your carrier should have their own echo cancel so If they are running asterisk with PRI interface cards that don't have echo cancel they you should consider changing carriers. Also have a look at this which gives a pretty good explanation of what causes echo. http://en.wikipedia.org/wiki/Echo_cancellation But bottom line; turn on AEC on the handsets and the problem will go away. Regards, John On Fri, 2007-07-13 at 17:24 -0400, Bill Sandiford wrote: John: Thanks for the description, it is very good. Here is the scenario in which this customer is experiencing echo. Polycom --- Asterisk --- Internet --- Polycom also Polycom --- Asterisk --- IX Private Line (delay 10ms) --- Polycom also Polycom --- Asterisk SIP Trunk to carrier --- Carrier's CLASS 5 --- PSTN This particular customer is seeing echo in all three scenarios. Hence the reason I'm looking into the AEC and AES features of the Polycom. Bill - Original Message - From: John Lange [EMAIL PROTECTED] To: asterisk@uc.org Cc: [EMAIL PROTECTED] Sent: Friday, July 13, 2007 4:38 PM Subject: Re: [on-asterisk] Polycom Echo Problems Echo *always* comes from the far end point. The amount a given person perceives the echo is determined by how loud and how delayed the echo is. Volume and delay are influenced by a number of factors along the call path. Echo is a very complex issue but I'll try and give a brief explanation. In the situation where you have a Polycom phone connected to an Asterisk server which is in turn connected to the PSTN talking to a residential wireline customer, e.g.: Polycom -- Asterisk -- PRI -- Wireline Handset If the Polycom customer hears echo it's coming from the wireline handset (and/or the hybrid but I'm trying to keep this example simple). Most consumer handsets just don't care about generating echo because its never been a problem. So echo is normal on all local wireline calls but you don't perceive (hear) echo because the echo is not delayed. Now when you throw Asterisk in the mix the act of encoding and decoding the voice adds delay. This added delay causes you to perceive echo even though the volume of the echo is roughly the same. Technically, to solve echo you fix the endpoint that's causing the echo. But since you can't replace every wireline phone ever made and the telco certainly isn't going to help you
Re: [on-asterisk] Rogers My5 and free incoming/outgoing
With my experience they are only delivering it on outbound calls...not inbound, unless its from another Rogers cell phone. Case in point I just called my Rogers cell phone from my residence and I only get CallerID Number, no name. - Original Message - From: Kevin, Legends To: David Steele Cc: asterisk@uc.org Sent: Monday, July 09, 2007 10:16 PM Subject: Re: [on-asterisk] Rogers My5 and free incoming/outgoing As Rogers doesn't seem to support caller ID names (only numbers), ... -- Yes, it support. Rogers Wireless is the only mobile provider in North American supporting caller ID name. It happened that I was involved to deliver this service to Rogers. :) On 7/9/07, David Steele [EMAIL PROTECTED] wrote: Hi everyone, I've just changed my cell phone plan with Rogers to take advantage of their new My5 plan - free incoming and outgoing to 5 Canadian DIDs (plus long distance fees, where applicable). I've got my Asterisk box set up with a DISA service (authenticated!), so all of my outbound can now be free of charge - if I can be bothered to key in the phone number each time. As an aside - anyone have any neat DISA dial-plan logic that makes initiating a new call to frequently called numbers easier? Or should I simply pre-pend my numbers in my cell phone with the DISA number, password, etc? Inbound is the kicker, though, and the real reason for this post. Can someone let me know if there is any way around the following problem: to qualify as a free call, all calls to my cell that I initiate or forward from my PBX will need a My5 caller ID number. As Rogers doesn't seem to support caller ID names (only numbers), this means I won't be able to tell who is calling, or whose calls I have missed. Can anyone see a way around this? Thanks, Dave.
Re: [on-asterisk] Rogers My5 and free incoming/outgoing
I pay for it. I get caller name from other Rogers Wireless Customers I get caller name from other Rogers Home Phone or Business Customers I get caller name from a call originated by a Rogers PRI I don't get caller name for anything else. - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: asterisk@uc.org Sent: Tuesday, July 10, 2007 12:42 AM Subject: RE: [on-asterisk] Rogers My5 and free incoming/outgoing Name Display is an additional service on Rogers which you have to subscribe to, and your handset needs to support it as well. Doesn't work for me (my handset does not support it nor do I pay for the additional service), but when calling from our PRI to a friend's Rogers Blackberry Curve, they do see the name sent from our PRI, even when I change it from call to call. Nabeel -Original Message- From: Bill Sandiford [mailto:[EMAIL PROTECTED] Sent: July 9, 2007 11:20 PM To: Kevin, Legends; David Steele Cc: asterisk@uc.org Subject: Re: [on-asterisk] Rogers My5 and free incoming/outgoing With my experience they are only delivering it on outbound calls...not inbound, unless its from another Rogers cell phone. Case in point I just called my Rogers cell phone from my residence and I only get CallerID Number, no name. - Original Message - From: Kevin, Legends mailto:[EMAIL PROTECTED] To: David Steele mailto:[EMAIL PROTECTED] Cc: asterisk@uc.org Sent: Monday, July 09, 2007 10:16 PM Subject: Re: [on-asterisk] Rogers My5 and free incoming/outgoing As Rogers doesn't seem to support caller ID names (only numbers), ... -- Yes, it support. Rogers Wireless is the only mobile provider in North American supporting caller ID name. It happened that I was involved to deliver this service to Rogers. :) On 7/9/07, David Steele [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi everyone, I've just changed my cell phone plan with Rogers to take advantage of their new My5 plan - free incoming and outgoing to 5 Canadian DIDs (plus long distance fees, where applicable). I've got my Asterisk box set up with a DISA service (authenticated!), so all of my outbound can now be free of charge - if I can be bothered to key in the phone number each time. As an aside - anyone have any neat DISA dial-plan logic that makes initiating a new call to frequently called numbers easier? Or should I simply pre-pend my numbers in my cell phone with the DISA number, password, etc? Inbound is the kicker, though, and the real reason for this post. Can someone let me know if there is any way around the following problem: to qualify as a free call, all calls to my cell that I initiate or forward from my PBX will need a My5 caller ID number. As Rogers doesn't seem to support caller ID names (only numbers), this means I won't be able to tell who is calling, or whose calls I have missed. Can anyone see a way around this? Thanks, Dave. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] surprising google purchase
Cash burning holes in their pockets.MUST SPEND !!! - Original Message - From: Simon P. Ditner [EMAIL PROTECTED] To: asterisk@uc.org Sent: Wednesday, July 04, 2007 9:48 AM Subject: [on-asterisk] surprising google purchase http://googleblog.blogspot.com/2007/07/all-aboard.html I still can't believe it, Google bought GrandCentral for $50m! For what? Certainly not IP... - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Calls not connecting on PRI span
Correct you are looking for a DS1-X cable. If I recall, T1 cables are RJ48c spec, so in a crossover scenario pin1 goes to pin4 and pin 2 goes to pin5 and vice versa. If you had the wrong cable, the carrier wouldn't even come up, let alone the B or D channels. - Original Message - From: Dave Donovan To: asterisk@uc.org Sent: Wednesday, July 04, 2007 2:01 PM Subject: Re: [on-asterisk] Calls not connecting on PRI span On 7/4/07, McQuiggan, Mark - Broadridge (Toronto) [EMAIL PROTECTED] wrote: Dave: Try a straight-through cable. You need a straight-thru to connect a pri_cpe to a pri_net. Maybe I misunderstand, but what you're saying doesn't seem right. I've never connected Asterisk to Asterisk but I've connected Asterisk line side to a bunch of different things (Avaya and Nortel PBXs). I've alway used a T1 crossover cable and then had one side act as NET and the other as CPE. My understanding was that the crossover cable addressed the electrical requirements and the CPE/NET settings addressed signalling requirements. Am I wrong? I would expect that if Dave had the wrong cable, he wouldn't get anything at all in terms of call setup. In fact, I would think the B and D channels wouldn't initialise. Dave Donovan
Re: [on-asterisk] 1004Hz and silent termination
Jim: After the last email that I sent you off-list, I found these numbers in my notes...they are from about a year and a half ago so I don't know if they all work still, but you can give them a shot. Milliwatt Test Numbers Seattle- 206-345-0020 Toll Free - 877-250-0600 Sprint 905 - 905-290-0102 Sprint 416 - 416-916-8102 Sprint 647 - 647-430-0102 - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: asterisk@uc.org Sent: Tuesday, June 26, 2007 4:37 PM Subject: [on-asterisk] 1004Hz and silent termination Anyone know the numbers for 1004Hz and silent termination? They used to be NXX-1185 and NXX-1191, but they don't work anymore. Jim -- Jim Van Meggelen Core Telecom Innovations [EMAIL PROTECTED] www.coretel.ca 416-425-6111 x6001 877-CORETEL x6001 (Canada) IAX2:[EMAIL PROTECTED]/6001 www.oreillynet.com/pub/au/2177 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.8/869 - Release Date: 25/06/2007 5:32 PM - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
NXTcomm next week in Chicago
Is anyone going to NXTcomm next week in Chicago? For those of you that aren't familiar with NXTcomm, it is by far the largest Telecommunications show in North America, and probably the world. I personally attended last year (when it was called Globalcomm). It was by far the largest trade show that I had ever been to. I spent 2 full days (10am-5pm) and I still didn't have enough time to get through all of the tradeshow floor. Details on the show can be found at http://www.nxtcommshow.com/ All of the big VoIP and telephony companies are there. There was a lot of vendors with asterisk related content there as well. Digium will be there. I am going driving down again this year with a colleague. We are leaving the Toronto area around 1pm on Monday in order to get there late Monday night so we are ready to hit the show floor Tuesday at 10am when it opens. If anyone is interested in following us down or meeting up with us there let me know. We are planning on heading back around noon on Thursday. For those of you that haven't been to a show this size before, apart from all the free swag there is to be had, all the major vendors are their with their HUGE marketing budgets. This means FREE FOOD, FREE BEER, FREE BOOZE, and tonnes of FREE partys. As a new CLEC I already have invites to most of the big parties and I can get anyone that tags along in to those parties as well. For example, Wednesday 6pm - midnight dinner and drinks at the legendary Chicago House of Blues...open Bar. I can tell you that last year in the 3 days that we were there we didn't pay for a single meal. The recruiting company dice.com started handing out free beer at 10am. I also have plenty of URL links to complimentary tradeshow passes from vendors for those that don't want to shell out $150 USD to get in the door (even though it is worth it) Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
Disable DND on Polycom
Does anyone know how to disable the Do Not Disturb button/function on a Polycom SIP phone. We have far too many staff members that are putting their phone on DND and forgetting to take it off. I want to remove the ability for them to put it on DND
