Re: [asterisk-users] My spa has a mind of its own
I had a similar issue a while ago. Check your dial plan. Are you forwarding to your cell phone's V-Mail as fallback? I had the issue where I was getting callbacks from asterisk if one phone was on DnD and the calll wasn't answered. Becarefull of your dial() commands and the delays you use. Steve Edwards wrote: I have a Sipura SPA-841. It's developed a nasty habit. At random times, it likes to dial my cell phone voicemail number and play my messages to anybody who happens to be within earshot. Any clues where to look at what's going on? My voice mail number (extension 220 in my dialplan) is the only number being dialed. When this happens, show channels looks like this: IAX2/NuFone-1(None) Up Bridged Call(SIP/spa841-09f083 SIP/spa841-09f08388 [EMAIL PROTECTED]:5 Up Dial(IAX2/mumble:mumble which looks the same as if I dial it myself. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- It is completely one's own responsibility to think outside the cucumber. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
Zeeshan, First off, if your fear of being sued is what stops you from doing business then get out of the industry or get over it. Its a risk we all take everyday (not just in VoIP). You build up a core of Insurance and Defensive Patents to protect yourself. Risk is just part of doing business. Elements of the Asterisk that are clearly incompatible with the Dual License model are not included in the regular distribution. You may find them as add-on modules or in Trunk (If it supports a free development/education license) but not as a part of the regular distribution. To address the real issue... In the USA in recent years companies have been granted broadly worded patents. People at the patent office are clerks and not engineers. Plus, they have to deal with ALL INDUSTRY (e.g. Medical, Aviation, Computer Science, Earth Science, Early Childhood Development, Mining, Agriculture, Automotive, Maritime, Textile, Nuclear Physics, Beauticare, Electronics, Chemistry, Mechanics, Pharmaceutical, etc...etc...etc...) not just Telecom. It is quite literally impossible to understand enough about everything to make clear judgments as to what is truly patentable and what is not. The patent office position is basically Spell it out to us and let the courts figure out the rest. While most broadly worded patents are unenforceable it still takes a legal process to get the patents dismissed as too vague. That process can be VERY costly for the person sued as well as the suer (sp). For a large telecom is all just part of the cost of doing business. Most smaller companies (e.g. us guys) are forced to settle because we haven't the millions of dollars needed to defend ourselves. Now that being said where does the g729 patent (and the like) fit in? A patent like g729 is actually VERY specific about what it does and how to do it. Sure its a software patent but there is little room in the wording about what it accomplishes, by what means and the limitations of the patent. Plus the price is very reasonable at $10/channel (non-transcoding pass through requires no License). Additionally, g729 is not the only game in town when it comes to low-bandwidth codecs. (Personally I like to use g726-32 its lightweight and transcodes to/from uLaw easily...but I digress). This varies from some other software patents for One-Click-Checkout or Online Shopping Cart. They are both patented and every challenge has been settled out of court, thus they still stand a viable patents. Ultimately the question comes down to...Do you want to stay home and hide or would you rather come out and play? Just my input, Mark C. Zeeshan Zakaria wrote: Hi everybody, As the Asterisk community is getting larger and larger, I was wondering that the features which are provided in Asterisk and are programmed by the open source community under GPL, or GUIs like FreePBX which also come loaded with wonderful features and uses same Asterisk, are they anywhere violating any patent laws? Most of the features work the same way as Nortel, Avaya and other PBX systems. Is there anyone who owns these features and will come one day to claim his royalties? When I deploy an asterisk soultion for a customer, is there any violation of any patent or copyright laws anywhere? Of if I use my own Asterisk server to provide services to some customers, am I violating any patent laws by not paying the royalties to some patent owners? I heard people saying that IVR technology is patented and google search for patents also say so. But we all are using IVR for ourselves and our customers without paying royalties to anyone. But when it comes to using g729, all of a sudden royalty issue comes in. So what is right to use and what is not? -- Zeeshan A Zakaria -- As I slowly sip my coffee I feel my humanity start to slip back into me and realize what a foul beast humanity really is. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Steve Totaro wrote: What if a train derails and slices through the main fiber connections. OK, so you have XO, Global Crossing, Verizon, and UCN all for redundancy. Well guess what? They are all most likely running over those strands of fiber. You better have a VSAT connection too! That's why I lease a few servers in a data center on other side of the country. Setup in a hot stand-by state. Its that peace-of-mind you can't buy any way else. So it costs a few hundred dollars a month (actually less then $500). It kicks in to take-up capacity when my main servers gets real busy or go off-line for maintenance. Its instant and automatic. Ok sure it took a lot of planning to get it right, but that's what I get paid to do. Single point of failure should NEVER completely disable your company. Yes outages happen and backhoe's cut fibre all the time. From within this stuff can make one's life rather difficult, but from the outside it should be almost unnoticed. When was the last time you noticed an outage at Google, Microsoft or the DoD? Do you think they don't happen? Its not that difficult or all that expensive if planned and implemented properly. Mark C. -- As I slowly sip my coffee I feel my humanity start to slip back into me and realize what a foul beast humanity really is. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Stephen Bosch wrote: Of course not -- but how many hundreds of millions have been invested in their infrastructure? You missed the point. The standard formula I use is 5 days out or more precisely 2% of gross revenues each year. For google its still a kings ransom, but for a small business it not too hard to implement. Case-in-point I have half a dozen cell phones I bought at the beginning of the summer (V3RAZR). Costs me nothing (after rebates) to setup other then an afternoon at the cellophane store, but I like toy shopping anyway. The residual cost is an additional $60/month. I now have a practical backup to my phone system. My regular phone stuff (office DiDs and 1-800 number) is setup to forward to the cells on a simple phone call. Its not a perfect solution but I've planned something which is better then most. Most people's way of handling an outage is to go home and let the techs fix it for tomorrow. -- As I slowly sip my coffee I feel my humanity start to slip back into me and realize what a foul beast humanity really is. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Steve Totaro wrote: Setup those cell phones to use chan_mobile and you have a very nice solution. Unless the phones are assigned to people who use them as their own. You could possibly add some lines on a family plan $10/mo extra w/T-Mobile and use those strictly as PSTN fialover lines. Thanks, Steve Actually the chan_mobile is a really good idea. I'll need to look into that. I was just gonna have them on hand for grab and go use. With a means to fall back in the event my PSTN or VoIP failure. Mark C ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Log analizer software
Hello all, I'm looking for software for my asterisk logs that will compile the information into nice web-based charts and graphs. Something that works similar to webalizer for apache. I want to be able to spot trends of usage, call volume levels, disconnect/failure levels, and basically see exactly where my system has been at over the past day/week/month, etc. . I would prefer for the software to work with 1.2 and 1.4 but 1.4 is the more important version for me these days. I have found FOP to be a great tool for a current snapshot of where I'm at but I have no indication of where i've been unless I'm watching it (a little to busy these days to just watch my pbx). Your input is greatly appreciated. Mark C http://www.psh-inc.com http://www.alohafreewifi.com http://www.coccimiglio.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RF to IP bridge
Per Jessen wrote: Radio-amateurs have done phone-patching for decades (where allowed) - there must be someone who can point you in the direction of an easy solution. /Per Jessen, Zürich The BIG problem here is that most Radio Amateur software and hardware operate in a half-duplex manner. I don't think that would be what you want. If half-duplex is ok then most radio makers (Icom, Motorola, etc.) have complete turn-key solutions. If you want it cheap then your will have to build it yourself. I don't see $200/channel happening in either case for VHF/UHF. Please share more info and maybe I can help. Mark C ( N3WHX ) [EMAIL PROTECTED] sip:[EMAIL PROTECTED] (VoIP) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
Eric ManxPower Wieling wrote: Michael Collins wro Except that for some users 1.2.18 is NOT stable. I've had to roll back to 1.2.15 on my production servers in order to prevent core dumps at least once per day. No, I am not willing to turn my production servers into testing servers to solve this. Doing so would make me a former consultant for these customers. You seem to miss the idea here. You work with a version that supports your feature needs and find the sub-version that provides the most stability for your deployments. Lets face it these boxes should go in and run for weeks, months or even years without much intervention (assuming the mission of the box does not change). I'm running a 1.2.7.something (i think) that has been running almost nonstop since installing. Very reliable and stable for my needs. Compared to a Merlin or Nortel or any other system out that I feel I have a much better product. Could I benefit from a newer sub-version? Maybe. Will I upgrade the box in it current roll? No. Unless the application I use the box for has a major change (or the hardware dies) I'll just let it keep on running as it is. For my future deploys I am working closer with 1.4. The reason is clear. 1.4 is the future of asterisk. When 1.6 or 2.0 comes out I'll investigate into migrating in that direction at that time because that will become the future of asterisk. smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Matthew J. Roth wrote: In fact, it seems that somewhere between 200 and 300 calls, the two servers start to exhibit similar idle times despite one of them having twice as many cores. Sounds like you are running into the hardware limitations of your systems PCI or Front Side Bus (FSB) and not necessarily an issue of asterisk. In short there is a limited amount of bandwidth on the computer's PCI Bus (33 MHz) and the FSB (100-800MHz). One thing to remember is that ALL cores and data streams need to share the PCI and FSB.Asterisk is very processor and memory intensive. At the extreme level of usage more cores won't help if data is stuck in the pipe. So the performance planing you described would be expected. Mark C. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
Tzafrir Cohen wrote: What I say is that you have the worse of both worlds: - downtime of at ~1/2 a minute (avarage, if a cron runs every minute). In the case a restart is all it take. - A bigger downtime in case a restart is not what it takes. Because your logs will be flooded. - And a most unpredicatable behaviour. So why would Asterisk crash? And if so: would a simple restart really do? Tzafrir; Well I would seriously disagree. Assuming that asterisk just hit a bump in the road and needs a restart then an average downtime of 30 seconds is minimal. I'm assuming that scripts like safe_asterisk include restart delays anyway, just to let things settle before trying again. So there is really is no difference here. As for bigger downtime, well if asterisk won't restart your startup method isn't gonna change anything. Most monitoring scripts will keep trying just as cron will the difference is that the script can be set to stop after a certain number of time (maybe a dozen or so depends on the config). So your logs will be just as full anyway. And the behavior is in no way unpredictable. If anything it is predictable with clock-work efficiency (please excuse the pun). Lets face it that in a production environment if asterisk dies the manager that is on will be calling from their cell phone to get someone to fixit ASAP. I did say originally that this was a simple and less then perfect way to handle things, but it does work well. If you don't like it that is just fine. I have found that it is a reliable way to insure that asterisk will restart on a stable system that does not support inittab. Mark C. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
Tzafrir; Actually I have found this config to work really well. I prefer to use a script run from inittab but Ubuntu doesn't work like Redhat or BSD. On a production box keeping asterisk up and running is THE TOP priority. If you would rather check every five minutes then replace the first * with */5. I will address your points as it seems that you haven't really thought about this. 1) In a production environment you should NOT be messing with the config. That's what test hardware is for. 2) The answer to this question is: crontab -e its really not that hard. I'm not running asterisk every minute. I'm looking to see if asterisk is running and then act accordingly 3) If asterisk fails believe me a full mailbox is the least of my worries. As for full logs I'd rather have more informationgrep awk are your friends. I prefer to keep things as simple as possible. Sure scripts like safe_asterisk are nice and do some really neat things but lets face it how often do you actually sit at the console of your asterisk box. My main PBX is located about 7 feet from my office desk and I still mostly use ssh (not even telnet) to get into the box. Mark C http://www.psh-inc.com Tzafrir Cohen wrote: On Fri, May 04, 2007 at 01:59:41PM -1000, Mark Coccimiglio wrote: What I do is add an entry in the crontab file as such: * * * * * if [ ! `/bin/pidof -s asterisk` ] ; then /usr/sbin/asterisk; fi Its simple and it works. Additionally if asterisk crashes then cron restarts the server in about a minute. Just be careful with your configs. It will not Just Work, because: 1. you may want to give Asterisk other command-line parameters (-p, -U) and not do that through asterisk.conf . 2. You may have your own reasons for wanting to stop asterisk occasionally. Having it run every minute from a cron job is a source for problems. 3. In case running asterisk generates an error, you get a very ugly flood in your logs and your mailbox. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
What I do is add an entry in the crontab file as such: * * * * * if [ ! `/bin/pidof -s asterisk` ] ; then /usr/sbin/asterisk; fi Its simple and it works. Additionally if asterisk crashes then cron restarts the server in about a minute. Just be careful with your configs. Mark Coccimiglio IS Manager Payroll Services Hawaii, Inc. http://www.psh-inc.com Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudspeaker
Just run down to your local Radio Shack...and KISS. http://www.radioshack.com/product/index.jsp?productId=2062696 Mark C. Klaverstyn, David C wrote: This is what I want. Do you have any URLs to such a device as I cannot find any. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cb Sent: Monday, 16 April 2007 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
These are the patent numbers in the lawsuit (Thanks Pat and Sal) 6,137,869 6,430,275 6,104,711 6,282,574 6,359,880 6,128,304 6,298,062 Mark C. Yuan LIU wrote: From: Kenneth Padgett [EMAIL PROTECTED] Date: Mon, 9 Apr 2007 23:49:31 -0400 [good stuff sniffed] I'm not doubting that patents exist, I'm just betting that you'd have to have some seriously drunken vision to interpret them as the exact business processes Vonage uses. I think if Verizon thought for a second they had solid ground to stand on, they would disclose which patents they're referencing so the public could decide. I bet you can access court records under some public information access laws. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linux Kernel Timer Frequency and Asterisk
Ok here is a real geek question, I building my own linux kernel for my asterisk system and came across the kernel setting for the timer frequency. I have one of 3 hardcode choices 100Hz, 250 Hz and 1000Hz. From what I understand the default Freq was changed from 100Hz in kernel 2.4 to 1000Hz (1KHz) in kernel 2.6. Timing is a BIG issue in asterisk with all the TDM and zap channel stuff. My guess is to go with the lower 100 or 250 Hz option but that is only a guess. The 1KHz sounds like it will conflict with the Zap 1khz timer (or am I wrong about that). Does anyone know what the prefered settings are for Trixbox or AsteriskNOW (or the asterisk code forks e.g. OpenPBX)? Please let me know what your experience has been. Aloha, Mark C IS Director - Payroll Services Hawaii, Inc. http://www.psh-inc.com FWD: 293625 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WellTech 380x Gateway
Ok this is a simple question... What has been your experience with the WellTech 38xx series (I'm looking specifically at the 3802) VoIP gateway? I'm looking for a good (and hopefully not too expensive) VoIP/T.38 gateway for my office. Asterisk intergration is not a major factor at this time but may be later on. How well does it work? Is Echo a problem? Do the T.38 capablities actually work? Please share what experience you have had. Also any experience (good or bad) with other T.38 gateways/ATAs. Thanks a lot. Mark C http://www.psh-inc.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3K to SPA3K DTMF issue
My experience has been to be consistant. The only time I have had problems with DTMF is when I am not using the same DTMF encoding technique on all hardware. Your choices are: INFO, RFC2833 or INBAND. Some equipment also has an AUTO option but I would not recomend it. Stick with INFO or rfc2833 and be consistant across the enterprise. Mark C IS Manager http://www.psh-inc.com [EMAIL PROTECTED] wrote: Hi all, Has anyone faced an issue when sending DTMF from FXS of one SPA3K to FXO of another SPA3K through asterisk? Im not able to send it properly. Wanna be sure if its an issue faced by all.. If you have a fix for it, pls guide me. Thanks Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a low cost cell phone base station for asterisk ?
M.Hockings wrote: I don't really know the name of what I want to look for but maybe someone could tell me if it would be available. I have a number of old analogue cell phones laying about here and I was thinking it would be useful if I could set up a short range base station for them that would cover maybe an acre or so. What I would like to be able to do is use it to connect into Asterisk and this way have a useful wireless extension-phone range. I do know that there are WiFi IP phones available but based on the connection range to our WiFi access points it seems limited as is our existing wireless handset (POTS). Any thoughts, suggestions ? Mike You have a few options... Firstly I would suggest throw away or donate the old phones. There is much better technology then Analog Cellular. Simple Choice 1: Get new GSM phones subscribed on the same carrier and a GSM terminal. Make sure the phones all have free in-network calling (assuming that option is available in your country). Also setup the GSM terminal on the same group and hook it up to your asterisk server (think of it as a cellular extension). Lock the phones so that they can only call each other and the GSM terminal. Cost: (assuming 5 phones 1 terminal) ~$2000 to start and $150-200/mo. YMMV More complex choice 2: Get an RF engineer to design you a real WiFi coverage footprint and Wifi phones. Cost: $4000-7000 (or more) for consultation, hardware and setup. No reoccuring charges (hopefully). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)
Mark, Do these 1600 series Cisco routers you mention that you find on eBay for $50-$150 support Layer3 routing? I have a managed switch setup on my home network with several VLANs defined. (work subnet, home subnet, VOIP subnet) I currently have to use a Linux box to route between the VLANs. I'd like to move to Gigabit routing, but I'd need to replace the Linux box(more processor power and new NICs) and that gets expensive. I'd much rather have a router or smart switch for that matter that does Gigabit Layer3 routing all in one unit. Do you have any recommendationsthat wouldn't break the bank? Thanks, Ed Ed, Layer3 routing is a fundamental function of a router which is supported by the Cisco 1600 series (1605R specifically) router. However VLAN definitations are not supported in the 1600 series. You would need to moveup to the 1700 or 2500 series for that function. As for Gigabit support the 1600 and 1700 series do not support that high speed interface. These router are designed around WAN style routing operating at ~1.5Mbps. Gigabit routing is a rather cutting edge capablity that is only seen in newer hardware. I would checkout a Cisco Catalyst 3500 series for something like that. Be carefull and look closely some systems only support 2 ports on 1000baseT and the rest are 100BaseT. Good luck and happy hunting, Mark Coccimiglio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)
shadowym wrote: Regardless of the 1600's spec's which are outdated in many ways by todays standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY MANY hours on it. Sure, they are built to last but they do not last forever. I would consider ANY of these boxes as somewhat unreliable for high availability requirements. Buzzwrong answer! Don't answer on things you have no idea. and stop providing bad information. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)
Jon Pounder wrote: you should take your own advice - an acre is 200ft x 200ft - what idiot would pay a consultant $7000 to tell them they need one access point in the middle. I have a BA in Electronic Engineering, a Masters in Computer Science and I'm an FCC licensed radio operator. I think I know what I'm talking about. Life isn't always as simple as that. What if its a warehouse that is 60x800ft. still about an acre (I've seen this one myself). How will the system perform once the empty space is occupied with inventory? How will metal shelving effect performance. What hardware should you use? Netgear, dLink, Linksys, Cisco (they are different), Alvarion, Proxima? If its an outside area an AP in the middle is not necessarly practicle. You can't just use any antenna combination you want There are rules governing use. Are you certified to assemble and test such a system for Part 15 compliance? Do you know the specs and ERP limits? Who has presidence FCC or OSHA regs? What about other ISM bands? How long can you make your ethernet runs or should you use Fiber Optics? These are the types of things that an Engineer addresses. ...one access point in the middle. It may work it may not. One thing for sure is that the system probably won't perform as you expect it. Mark C ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)
Marty, Where are you paying $1000 for a 1600 series Cisco? I can get you 20% off that price on any quantity (note: Sarcasam). Its not the 1990's anymore. You can get them on eBay ($50-150) for only slightly more then the Linksys. The performance is rock solid. Three-quarters of the world have used them for decades. I know of units running 2 and 3 YEARS between reboots. The power company reboots my equipment more then I do. Ok it is true that Cisco does not support the models anymore, but you can't buy a services contract for a linksys router either. It can sometimes be a little difficult to configure without any technical knowledge but that is what most of us get paid for. It does impress the customer when you bring in the grey box labled Cisco. As for performance just try to put 50 people behind a linksys/netgear/dlink. I've used 1605R supporting +100 users. Not even a blink. Finally, untill everyone is using 10Mps FTTH the broad band link is still the slowest part of the connection. Not to shabby for antiquated technology. Mark C Martin Joseph wrote: On 2007-01-06 00:48:11 -0800, Mark Coccimiglio [EMAIL PROTECTED] said: Mike I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router with Fair-Weight queueing enabled. Works great. The nice thing about Fair-Weight queueing is that it dynamically adapts to lower the priority of higher demand traffic (e.g. large downloads). If you want quality stick with quality stuff. Mark C Reread the subject line please. $1000 (US) isn't inexpensive by any stretch. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best inexpensive home office router for VoIP (QoS with maybe PoE)
Mike I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router with Fair-Weight queueing enabled. Works great. The nice thing about Fair-Weight queueing is that it dynamically adapts to lower the priority of higher demand traffic (e.g. large downloads). If you want quality stick with quality stuff. Mark C Mike wrote: Hi, I'm looking for opinions on the best value router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source IP (i.e. an Asterisk server) or MAC address of the phones. Ideally with PoE, but at this point it's just a bonus. What are people on this list using? I've found that the mention QoS on a box doesn't guarantee any real QoS functionality. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
Try setting in sip.conf: nat=route This tells asterisk to send all responses back to where the inquiry came from rather then from the info contained in the sip packet. Good luck, Mark Coccimiglio IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com Elpidio Ramos wrote: This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux: My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC) I got the ports open in fedora core 3 (5060 and 1 thru 3) for both interfaces. I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP. When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop. This is my sip.conf file: [general] context=ramosoft allowguest=no realm=ramosoft.com bindaddr=0.0.0.0 bindport=5060 srvlookup=yes pedantic=yes tos=184 tos=lowdelay maxexpirey=3600 defaultexpirey=120 disallow=all allow=ulaw allow=ilbc allow=gsm musicclass=default language=es relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 useragent=RamoSoftPBX regcontext=ramosoft localnet=10.10.10.0/255.255.255.0 rtcachefriends=yes [authentication] [311] type=friend regexten=311 username=311 secret=311 callerid=Elpidio Ramos 311 host=dynamic nat=yes canreinvite=no Is there anything I am missing here to get two way voice? Thank you in advance all ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Colomachine TE405P
I was wonder if anyone is rumming this combination of hardware: Colomachine.com: CM62 Digium Card: TE405P I need a rackmount to send to a data center and this combination fits my budget. Has anyone else used colomachine with asterisk? how has it performed? I plan to run the latest trixbox on it with the RAM upgrade to 1GB. Should I go to the 2GB RAM? Will this combination handle all 4 spans at full load? I'm mostly intrested in realworld experiences. I hear a lot of don't trust because of the cheap price but when I talk to actual users they seem quite happy with these servers. Your opnions please. Mark C. IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXP2000, Linksys RV082 Firewall / NAT, Registrations
Do you have STUN Enabled? I had similar when I had STUN turned on. I found it better to turn off stun and place in sip.conf nat=route. Also use NAT Keep-Alive on the ATA that is NAT Timeout on the Router. Good Luck, Mark Coccimiglio IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com [EMAIL PROTECTED] wrote: Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they keep unregistering and re-registering every few couple of minutes (between 4 and 10 minutes). We have checked the client's Internet connection and it's not bouncing as well as their local network is working stable. So, the only thing we can think of why these phones are re-registering so often is their firewall device since it's the only difference between this client and our other clients that use the GXP2000. Does anyone have any idea why this is happening or how this can be resolved? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXP2000, Linksys RV082 Firewall / NAT, Registrations
I prefer to keep my NAT Timeouts short ( ~5 minutes) and lets the applications be responsible for keeping the connections open. **Most** consumer grade routers use a timeout interval of 1 hour to 1 day. A safe figure to start with is 600 seconds (10 minutes) and see if anyone complains. [EMAIL PROTECTED] wrote: Did I forget to mention I had STUN enabled? :) Well, that did it. Your suggestion worked perfectly. Does anyone know what a reasonable NAT Keep-Alive to use, if you don't have access to their firewall/router configuration? Thanks, Daniel -Original Message- From: Mark Coccimiglio [EMAIL PROTECTED] Sent: Mon, December 18, 2006 2:27 pm To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] GXP2000, Linksys RV082 Firewall / NAT,Registrations Do you have STUN Enabled? I had similar when I had STUN turned on. I found it better to turn off stun and place in sip.conf nat=route. Also use NAT Keep-Alive on the ATA that is NAT Timeout on the Router. Good Luck, Mark Coccimiglio IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com [EMAIL PROTECTED] wrote: Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they keep unregistering and re-registering every few couple of minutes (between 4 and 10 minutes). We have checked the client's Internet connection and it's not bouncing as well as their local network is working stable. So, the only thing we can think of why these phones are re-registering so often is their firewall device since it's the only difference between this client and our other clients that use the GXP2000. Does anyone have any idea why this is happening or how this can be resolved? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] connect Sipura with Asterisk - both behind NAT
I have 2 SPA-841s and an SPA -3000. I have found that setting externip to be usefull, but what really helped the most is to set the SPAs to use distinctly different ports. SPA841(#1) uses ports UDP 5066,5067 and RTP 16391-16393. SPA841(#2) uses UDP 5068,5069 and RTP 16395-16397. Finally I have the SPA3000 on UDP 5070 and RTP 16399-16401. I don't use STUN (tends to cause more problems then it solves). On the Server side I have the NAT firewall/gateway forwarding UDP port 5060 and RTP 16393-16401 to *. In sip.conf set nat=route for each NAT client. Hope this Helps, Mark Coccimiglio [EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] Joseph wrote: Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TigerJet PCI PPG FXO Card
I bought one over a year ago along with the USB phone. Was never able to get the card to work properly with anything (even the software it came with). For less money I got am X100P clone on ebay and that works great. Leo Ann Boon wrote: Anyone has any experience with these cards? Looks suspicious like the X101P. http://www.cuphone.com/products/ppg/index.htm smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad BRI card
From what I understand the B410P us intended for use OUTSIDE North America. I contacted them a little over a month ago looking for a USA compatable card and was told that there isn't sufficient market for the hardware. Oh well, so if you are in Europe or most other places you will have a Digium option soon enough. Otherwise the Diva server cards are a good option (extensive, but come highly recomended from most that I hear). Good luck and happy hunting. Mark C VoIP: [EMAIL PROTECTED] Wayne Gemmell wrote: Hi all Does Digium make a quad BRI card? I can't see anything of the sort on their page but I thought they might call it something else in the States. Failing that, can anyone recommend a make/model that would handle 4 BRI ports? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 911 @ Zap Channel Breakin
Ok here is one for you. I know we all do the this for 911: exten = _911,1,Dial(Zap/1/911) exten = _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a least cost routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is durring a 911 call test if the Zap channel is Available (probably using ChanIsAvail() ) to test the line. IF the channel is in use I want to barge in with an announcment saying that the line is needed for an emergency and the call we be disconnected. Then immediately drop the call capture the line so noone else can use it, wait about 5 seconds for the telco to clear the far end and place the 911 call. Is this possible? Thnaks Mark C [EMAIL PROTECTED] FWD: 293625 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk BRI in the USA - Episode 2 The Phantom Sales Rep
Hey all here's an update. I do care to thank everyone for your information on BRI interfaces that operate in USA/NA. I know the responses are were limited, but the selection of hardware is also limited. (Shame because BRI would fit my needs perfectly). To continue, it's now been over 4 weeks since I last talked to the ILEC sales rep about pricing and plans. Unfortunately there has been no response. Which does not surprise me. I had an issue with DSL with the ILEC basically looking to migrate to bigger pipe and was shined on then. So i've decided to take the plunge into VoIP. Got asterisk up and running, works wonderful. Beats an SPA-3000 by a long shot (but that's the difference between a real system and a simple FXO gateway...no surprise here). So now I'm looking into VoIP providers that service Hawaii. I would need 1 DID on each of the islands and possibly an 8xx toll free number. I am looking into IAX.CC (SixTel). I wanted to know peoples take on this company. Hard price to beat, but I wonder what the network is like. Has anyone here had much experience with them? Can anyone make a recommendation for a quality ITSP that has a/c 808 DiD's? Is FoIP possible with any of these guys? As always your (non)professional opinion matters. Aloha, Mark C. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] www.SavaJe.com
Sounds like Propaganda to me. Dean Collins wrote: While not strictly on topic I think this could be an interesting opportunity for the Asterisk development community. As some of you might already know JavaOne will be happening in San Francisco in 2 weeks time http://java.sun.com/javaone/sf/ I wanted to draw your attention to an interesting company that will be exhibiting there called www.Savaje.com more details here http://pr_sun/exhibitor_info?id=5051 In summary Savaje is a brand new Java mobile phone OS that is highly programmable and customizable it has been awarded device of the show http://www.savaje.com/show_device.html There are a number of undisclosed hardware manufacturers who have signed on to offer hardware and undisclosed carriers will announce their involvement at the show. Now what does this have to do with Asterisk; lots. Basically at the show Savaje will be offering the first JavaOS handsets and appropriate SDKs for developers who want to build their own Java applications to run on these handsets. You can now write an application that will run on your mobile phone and be able to communicate over the data channel to external devices and applications. I think with the wealth of development talent in the Asterisk community we should be able to build some interesting apps that not only interoperate with Asterisk but also operate stand alone functionality. For far too long mobile functionality has been a walled garden without the ability to implement perfectly beneficial custom applications I think that this company might be able to offer us something interesting and exciting. Important point to note I dont work for Savaje I cant provide any more information than I already have. I will however be working on the Savaje stand for the 3 days of the exhibition. I cant say who but Im working with an as yet un-nameable Savaje development partner. If you are going to the show please feel free to come up and say hi when you visit the Savaje stand. Hope to see you there. Regards, Dean Collins www.Cognation.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?
Ok, I have to agree here. IF my simple fax server log/tiff archive is not enough to satisfy a client that the fax is genuine I would not want them as my customer. I don't care how much money they spend. Business is business and what I do is what I do. There has to be at least a little bit of trust between me and my clients otherwise we spend too much time bickering about stuff. Have you ever tried to bill "b*tch-time"? I have worked in "High-CYA" positions before and refuse to have that in my current company. Aloha, Mark C Scott Gifford wrote: I don't see the advantage to this; the client still has to trust that all of this is done correctly, and if they don't trust the fax recipient to put the correct fax in the paper file or keep the correct TIFF, why would they trust them to do this? Using a third party to receive and relay the fax, one which is trusted by both the client and the fax recipient, would solve the problem; the third party could create a document with the caller information (ideally from ANI, which is harder to forge), the time, and the message itself, then digitally sign it. This might even be an interesting business plan, for some applications where confirmed document transmittal is important. But it's hard for me to imagine this isn't overkill; if a client and a service provider distrust each other so thoroughly that they have to communicate through a third party to verify integrity, probably they just shouldn't do business with each other. Scott. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huawei EP201S
I have 2 of the Sipura SPA-841 (now Linksys/Cisco) and they work great once I got the latest firmware. They cost ~$100 each. Tomislav Parčina wrote: Has anybody used Huawei EP201S IP phone? I need 9 SIP phones that cost around 100USD, and those phones are one of options. Can anybody suggest anything else that costs around 100USD? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk BRI in the USA
I'm seeing Diva Server V-BRI running close to $1K/card. There are other Diva cards running around $700. A little pricy but not impossible to do. I remember back in the 90's I had ISDN into my home for internet access. The netgear router I used cost me about $350 back then, and it worked great. I still have it as a matter of fact. However internet access is not what I need. I'm still waiting for the ILEC (HawaiianTelcom) to get back to me to find out if it is even possible to do BRI into my office. The nearest ISDN capable CO is located a bit of a distance from my office (actually its closer to my home). The local CO dosen't have BRI capablities. From what I'm hearing when you bundle together all the costs BRI PRI are gonna be close in price (from a H/W point of view.) Maybe I should just look into going the PRI route and try to find some people willing to buy on my extra DiD's? Any one what a phone number in Hawaii? :) Its such a shame I can't leave well enough alone and suck it up on POTS (eck). I'll keep you informed as to my progress (or lack there of). Mark Coccimiglio n3whx @amsat.org sip:[EMAIL PROTECTED] Walt Reed wrote: I'm in a similar situation. Being on the end of a long loop, POTS sucks - echo / static / crappy calling features. Paying around $2K-3K for BRI solution is a non-starter though. It needs to get down to the $200-400 / port level (more ports = cheaper per port) to be viable. Soho / Very small business (under 12 people) is definately a 1-2 port market which my guess would be the bulk of sales for BRI. It would be awesome to see a Sangoma BRI card. It's hard to say what the market would be since the US telco companies have really tried to kill BRI service. Considering what a full PRI costs, there is also a point where too many BRI ports no longer makes sense, but that number is probably 4-6 BRI's. I was in a situation where I really only wanted 4 BRI's, but had to look at a PRI instead which ended up wasting a lot of money in the long run. POTS was a non-option. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk BRI in the USA
I guess what I need to find out first if there is anyone out there using Asterisk BRI in the USA? If so what hardware have they been able to use. I no longer want to hack around with analog circuits. BRI has the potential of PRI with only 2 B channels. A great idea for a small office such as my own. VoIP may be an option, but I would need a ITSP that would allow calls to transfer from my asterisk box to the remote phone set. My link to the internet is fast, but its pointless to route a call into the office just to stream it back out. More work more work more work. Mark Coccimiglio n3whx @amsat.org sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk BRI in the USA
My needs are simple and clear. I have an office with 2 employees, a sublet (independent business) and myself. I work p/t from the office and p/t from home (telecommute). My wife works p/t from home (telecommute exclusively). My other employee telecommutes from 350miles (we fly her in every 3-4 months as the needs fit). My sublet works full time from the office. We have an assortment of POTS,VoIP,cellular,fax mixed in popouri of technology. I want to unify my TC. POTS--FaxServer and BRI--Asterisk--VoIP Phones. I want to have the TC Unity of an office without the need to have everyone in the office at the same time. (Am I a dreamer?). BRI w/DiD through Asterisk would do that nicely. So would BroadVoice (Vonage has no DiD's where I live). I'm not affraid to spend a little to get a system that will do whatever I want. I currently use POTS --SPA-3000 -- SPA-841. Works well, but really only supports 1 user, depends on ILEC for V-mail and has no room for growth. A BRI would support 5-6 people without stretching the technology. A PRI is just TOO much, unless I could get it cheap and wholesale the extra DiDs. Am I heading the right way? Your suggestions would be greatly appreciated. Mark Coccimiglio [EMAIL PROTECTED] sip:[EMAIL PROTECTED] Lacy Moore - Aspendora wrote: Mark, I was in the same situation. Our current system uses BRI for almost all lines. I looked for some kind of solution and finally gave up. The BRI products here just seemed way too expensive for me. I then started checking on a PRI and ended up getting an integrated PRI with 7 channels for voice and the rest as a fractional T1 for data. It hasn't been installed yet, but once it goes live, we'll end up saving about $1000 a month from what we were paying with BRI and POTS. Most of this savings comes from needing multiple phone numbers, but not that many lines. We had 10 BRI circuits (equates to 20 voice lines). We're also getting rid of our DSL since this includes data. It kind of depends on what you have, and what you need (as far as voice and data), whether this will work for you. On 4/10/06, Mark Coccimiglio [EMAIL PROTECTED] wrote: Hey all, It such a shame that BRI technology is such a flop in the USA.For a small office such as mine it would be a great product.So her goes my questionWhat is a known asterisk working BRI card that will operate in the USA.I need to weigh price/quality.I need to do DID/DDI (or what ever you want to call it).Asterisk will do everything else I need.The ILEC has at the other end a DMS-100.I have been having all kinds of problems using POTS lines that I will consider it an investment to move to a more digital connection. I am considering going the VoIP route (Vonage, Broadvoice, etc...) but before I commit either way I'm exploring all my options. Your opnion matter here to please let me know. Mark Coccimiglio [EMAIL PROTECTED] sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk BRI in the USA
Hey all, It such a shame that BRI technology is such a flop in the USA. For a small office such as mine it would be a great product. So her goes my question What is a known asterisk working BRI card that will operate in the USA. I need to weigh price/quality. I need to do DID/DDI (or what ever you want to call it). Asterisk will do everything else I need. The ILEC has at the other end a DMS-100. I have been having all kinds of problems using POTS lines that I will consider it an investment to move to a more digital connection. I am considering going the VoIP route (Vonage, Broadvoice, etc...) but before I commit either way I'm exploring all my options. Your opnion matter here to please let me know. Mark Coccimiglio [EMAIL PROTECTED] sip:[EMAIL PROTECTED] smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
