RE: [Asterisk-Users] Pricing for DS3000P
Keep in mind that softswitch hardware + licensing costs ~$25k-35k/DS3, at least on a decent platform. Yearly support is on average, $10k/year. Meta has good rates. $4k is cheap, especially for this market. Now will they ever bother to produce compactpci boards? Probably not any time soon. This question has been raised several times in developer conferences on 996. If they could produce compactpci boards+software to run on cpci cpu's, redundancy capabilities etc, this is a cheaper replacement for softswitches. but no CLEC (no ILEC would ever buy cheap) would replace stable hardware+software with unknown reliability factors tied in. You can't just 'reboot' a switch's software (technically, you can... but I wouldn't want to be the one to do it). -m On Sat, 4 Jun 2005, Peter Svensson wrote: On Sat, 4 Jun 2005, Tom Fanning wrote: What's so special about Digium cards that makes them this expensive? $4000 for a PCB is extortion IMO! I'd say low volume and high development and certification costs. A contributing factor is what the market is willing to pay. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pricing for DS3000P
FYI: TNTs are $6-7k, fully loaded DS-3s including all DSP's on ebay. On Thu, 2 Jun 2005, izo wrote: On 6/2/05, Andrew Latham [EMAIL PROTECTED] wrote: I don't know, but pricing it per line whould be safe. Say $100 per line that would be $67,200.00. So anything less than that would be great. I think it will be about $20 bucks a port. 672 * 20 = 13,400 come on it must be cheaper ! for that price you can get Lucent MAX TNT Lets look at digiums cards 4xE1 = 1500 USD so 1500/120 = 12.5 per port if you consider the scale effect imho it'll be like 10 $ per port so end price somewhere about 6-7k USD regards m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pricing for DS3000P
Cisco's somewhat reliable firmware, of course. ;) On Sat, 4 Jun 2005, Pavel Jezek wrote: what's so special about eg. ci$co cards... WS-X6608-E1= Catalyst 6000 8 port Voice E1 and Services Module USD 19,995.00 ;-) PJ Tom Fanning wrote: What's so special about Digium cards that makes them this expensive? $4000 for a PCB is extortion IMO! Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pricing for DS3000P
4 to 1 ratio... is the industry standard for voice. 10 to 1 is dialup. Some raise it. I'm, personally, waiting a year or so to hear about the complaints from the lists before I bother. -m On Sat, 4 Jun 2005, trixter http://www.0xdecafbad.com wrote: On Sat, 2005-06-04 at 09:50 +0100, Tom Fanning wrote: Agreed, those are the figures we were able to get from Digium... I'm still waiting for a confirmation, but I'm being safe with a $4k estimate.. snip What's so special about Digium cards that makes them this expensive? $4000 for a PCB is extortion IMO! A ds3 does 672 channels, normally on 2 strands of coax (and there are bnc connectors on the pic on digiums site). The port cost is then about $6/port. That is really cheap in all honesty. 672 ports can in theory support about 10,000 customers (given the rather dated 7% of people use the phone at any given time - that figure I think was accurate in the early/mid 90s and I am sure its higher now but I havent checked any reliable sources for an update. I did read a more recent study that suggested that the average person usese the phone 6 minutes a day, I use it for hours a day my parents maybe 16 mintes 3 times a week, so who knows). Even if its 5000 customers (ie calling is 2x higher, people stay on 2x longer, etc) that is still much more cost effective than the 28 individual DS1s that it would take to fill a DS3. There is a EU standard that afaik is framed basically the same but instead of 4 DS2s which are 7 DS1s (logical framing a DS2 always exists on a DS3 physically afaik) its built upon E1s, so there are slightly fewer E1s since they are 30 DS0s instead of 24. Not to mention that a DS3 circuit normally costs about what 12 DS1s cost so its like getting 16 free. This makes everything cheaper in the long run, thus companies are able to offer better rates for PSTN interconnection which can be passed to the consumer. I am curious on cpu load, if all dsp functions are done via software instead of offloaded onto a specialized processor (DSP board) that has to have some effect on call processing, meaning a more beefy machine to handle the load, and the real possiblity of not having a single board do everything (application, media gateway, VoIP, etc). While it makes sense on that type of a system (high capacity) to spread it out for load balancing and redundancy and all that stuff that gives you a warm fuzzy feeling, it may now be more of a requirement. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pricing for DS3000P
On Sat, 4 Jun 2005, trixter http://www.0xdecafbad.com wrote: On Sat, 2005-06-04 at 05:48 -0700, Matt Klein wrote: 4 to 1 ratio... is the industry standard for voice. 10 to 1 is dialup. Some raise it. As I said my numbers were dated, and I didnt know what they were now. 10:1 is horrible for dialup, busy signals abound at higher than 7:1, or at least they used to. I havent worked for an isp that did dialup for 8 years, for the most part dialup has no money in it now. $10/mo accounts are also the reason the contention rate for a modem is up. You've got to be kidding me, dialup is huge. *Still!* Especially, when you have recip comp with the ILEC! Rural areas w/o Wireless, Cable, DSL.. what do they use? Dialup, *still*. Agreed, 10 to 1 is harsh in *some areas*. Rural, 10 to 1 is too high. You gotta know your busies to determine this figure, 7 to 1 is entirely fair in a non-rural area. I'm, personally, waiting a year or so to hear about the complaints from the lists before I bother. Could you elaborate on what exactly you are waiting for? Perhaps its my lack of sleep that is making it a bit harder for me to comprehend that sentence. What will be on what lists and from whom? Pricing, Bug Reports (i.e. all of the problems associated with previous digium products, google for them), and it'd be from you, who would shell out bug reports, spending $4k to tell me not to buy -- yet. :) clip -m ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pricing for DS3000P
Mike, ebay for Carrier Access, CAC, Widebank -- this turn DS-3s into T-1s. $500-$600 -- current market rate. -m On Sat, 4 Jun 2005, Michael D Schelin wrote: Are you kidding! $4000.00 is cheap for a ds3 board! Even if you don't use all of the 28 t1's it's better because you will now be able to put in as many T1's as you will most likely need. Expansion will be just simple configuration change. Also as I've read in these forums, the interrupt issue should go away as this should only need 1. Don't let the term DS3 scare you. I have herd there are DS3 to T1 adapters out on the market for as little as $500. If you need more than 1 4 port T1 card you should buy the DS3 card unless of course you only need 5 T1 ports. Jay Milk wrote: What's so special about two tons of steel and a little plastic and leather that you'd pay at least $20K for it? How come Adobe gets away with charging $300 for a simple CD, when you can buy a stack of 100 for less than $20? Content matters... And someone needs to pay for the development cost, testing, certification, etc... Or there wouldn't be any peripherals. -Original Message- From: Tom Fanning [mailto:[EMAIL PROTECTED] Sent: Saturday, June 04, 2005 3:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Pricing for DS3000P Agreed, those are the figures we were able to get from Digium... I'm still waiting for a confirmation, but I'm being safe with a $4k estimate.. snip What's so special about Digium cards that makes them this expensive? $4000 for a PCB is extortion IMO! Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pricing for DS3000P
OK, at first, I thought you were kidding me. Now I know, you are kidding me. PC hardware, at $12k? Give me a break. Besides, I already have estimated cost quotes from digium. Joker. -m On Sat, 4 Jun 2005, Andrew Latham wrote: Thanks Mike. existing DS3 card - http://imagestream.com/PCI_921-CDS.html For the list I will repeat that the cost should be ball park of $12K. Why you ask. If you can afford a DS3 then you can afford an extra $12K. I do not know anything extra other than the fact that this card will allow some users to drop their Cisco equipment totally. The opening editors note of the latest Linux Journal talks about the trouble caused by developers trying to interoperate with proprietary software. Its a game of catch up and not a winning one. We need to create or direct the future of the market to be as open as it can. Knowing that you have an option for that DS3 is a great feeling. Some real discussion would be about optional OC3 or a optical DS3. My local CO tech friend sees more optical than copper due to support costs. Powering and repeating a fiber line is cheaper than that of copper. Final question, is anyone from Image Stream on this list? http://imagestream.com Andrew Latham ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pricing for DS3000P
Andrew, I have never attacked, you or your statements, smartass. I can give you details on my HW, if I want, but if you're in the know-how, you apparently didn't even bother reading all of my e-mails, you didn't even bother reading that I'm in complete agreement with you on the DS-3000 pricing. Besides, Carrier Access Channel banks, 24 ports, $150 ebay per 24, Mainstreet/3624 blah blah channel banks, less than a hundred, ebay. It's cheaper to go DS-3. *ALWAYS*. Read my emails to the list you jackass. Want to invest in something fun? Call me, 541-312-4251. I just may answer my phone, this time. Otherwise, leave a message. -m On Sat, 4 Jun 2005, Andrew Kohlsmith wrote: On Saturday 04 June 2005 17:31, Matt Klein wrote: Mike, ebay for Carrier Access, CAC, Widebank -- this turn DS-3s into T-1s. $500-$600 -- current market rate. Ok smartass now what do you use to terminate those 28 T1s? Let's see... $1500 per quad T1 card, 7 cards required... 4 systems required (2 cards per system)... oh HELL YEAH, you just saved a pile of money. Who do I call to invest in your company? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pricing for DS3000P
kk ;) On Sat, 4 Jun 2005, Andrew Kohlsmith wrote: On Saturday 04 June 2005 18:15, Andrew Kohlsmith wrote: Ok smartass now what do you use to terminate those 28 T1s? Let's see... $1500 per quad T1 card, 7 cards required... 4 systems required (2 cards per system)... oh HELL YEAH, you just saved a pile of money. Who do I call to invest in your company? And of course here's where I stick not only one, but both feet in my mouth; I read too quickly and replied even more quickly. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pricing for DS3000P
$4k is beautiful get them to cpci it, though, and you're looking at double. -m On Sat, 4 Jun 2005, Forrest W. Christian wrote: On Sat, 4 Jun 2005, Tom Fanning wrote: What's so special about Digium cards that makes them this expensive? $4000 for a PCB is extortion IMO! $4K for a channelized DS3 card isn't all that bad. We've been paying ~2K for a free-framed DS3 card. Component-wise yov'e got upwards of $1K if not $2K on-board. Factor in RD time and some reasonable profit, $3K or even $4K isn't that bad. Now if you want to discuss whether or not the prices for the IC's and other components are extortion or not, then I might be willing to agree with you. -forrest ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pricing for DS3000P
Unofficial: Digium guesses that their DS3 card will be $3k - $4k. -- tack on a k or two to be safe. Later this year is my guess... from what I heard. -m On Thu, 2 Jun 2005, Nathan wrote: Does anyone have an estimate for the pricing on the DS3000P DS3 PCI card by Digium? How about a timeframe? Thanks, Nathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pricing for DS3000P
Agreed, those are the figures we were able to get from Digium... I'm still waiting for a confirmation, but I'm being safe with a $4k estimate.. timeframe wasn't given to me, but I was told this year (later this year). All of it, of course, unofficial. Hardware specs have already been discussed on this list for this board (a month and a half ago?), and from what I remember, they should be no more than the current hardware specs for a 4 port or two. Some of the software stuff has been moved to the board. I think all DSP is still done in software, as is echo can, but I think chanellization has been moved to the board -- someone correct me please. But to me, the real question is, when's a good DSP board coming out w/ * support??! -m On Thu, 2 Jun 2005, Jason Walker wrote: I called Digium about a week ago asking about ETA and initial pricing. The support person I spoke to said that they are shooting for a September release (Fall '05) and a price around $3,500 US. Take it for what it's worth - but I hope this is the price. Either way - I hope hardware specs come out soon on requirements. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, June 02, 2005 7:30 PM To: izo; Asterisk Users Mailing List - Non-Commercial Discussion; Andrew Latham Subject: RE: [Asterisk-Users] Pricing for DS3000P Yep anything over $7k makes it more feasible/reliable to go for multiple server multi-card solution. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of izo Sent: Thursday, 2 June 2005 8:21 PM To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Pricing for DS3000P On 6/2/05, Andrew Latham [EMAIL PROTECTED] wrote: I don't know, but pricing it per line whould be safe. Say $100 per line that would be $67,200.00. So anything less than that would be great. I think it will be about $20 bucks a port. 672 * 20 = 13,400 come on it must be cheaper ! for that price you can get Lucent MAX TNT Lets look at digiums cards 4xE1 = 1500 USD so 1500/120 = 12.5 per port if you consider the scale effect imho it'll be like 10 $ per port so end price somewhere about 6-7k USD regards m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 267.4.1 - Release Date: 6/2/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What do you name yours
Mine is named spike... On Thu, 12 May 2005, Paul Hales wrote: We bought one of those books on the worst cars ever made...every page has great names... PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Thursday, 12 May 2005 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Andrew Latham Subject: Re: [Asterisk-Users] What do you name yours On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P does not fit in motherboard
It's you. Get a 3.3v supported Motherboard. -or- Grab a Hacksaw. Others will post instructions. ;) On Wed, 4 May 2005, Daniel Salama wrote: I'm trying to install a TE410P on a Gigabyte motherboard (8S661FX) and I'm just noticing that the TE410P does not fit in the PCI slot. It seems as if the little opening in the PCI is on the wrong side. Has anyone else seen this or is it just me and I'm too stupid to do something as basic as this? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P does not fit in motherboard
Vice Versa. http://www.digium.com/index.php?menu=wildcard_te410p The TE405P is for 5v slots. To counter, there are, and I won't suggest them, ways to make the TE410P work in a 3.3v slot. This would basically entail cutting it to fit.. and has been proven to work, but it is not recommended by Digium. There are instructions on Google. -m On Thu, 5 May 2005, David Phelan wrote: Corect me if I am wrong, but the TE410P is for 5v PCI Slots.. I think you need to be using the TE405P (3.3V PCI) Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, 5 May 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TE410P does not fit in motherboard I'm trying to install a TE410P on a Gigabyte motherboard (8S661FX) and I'm just noticing that the TE410P does not fit in the PCI slot. It seems as if the little opening in the PCI is on the wrong side. Has anyone else seen this or is it just me and I'm too stupid to do something as basic as this? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.5 - Release Date: 4/05/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P does not fit in motherboard
Hey, I remember those commercials, too! On Thu, 5 May 2005, David Phelan wrote: [Oooops] Corect me if I am wrong, but the TE410P is for 5v PCI Slots.. I think you need to be using the TE405P (3.3V PCI) [/ooops] Got that back-to-front Maybe I should Have had the Scrambled Eggs instead of my Brain... Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, 5 May 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TE410P does not fit in motherboard I'm trying to install a TE410P on a Gigabyte motherboard (8S661FX) and I'm just noticing that the TE410P does not fit in the PCI slot. It seems as if the little opening in the PCI is on the wrong side. Has anyone else seen this or is it just me and I'm too stupid to do something as basic as this? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.5 - Release Date: 4/05/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Freak incidents, who's to blame?
If you're going through a CLEC for your lines, they can probably set the Glare Preference to be You or the Telco. I'm not sure if the Baby Bells would add that preference option for you. -m On Tue, 3 May 2005, Eric Wieling aka ManxPower wrote: Ryan Courtnage wrote: Hello all, Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? It's called glare. http://home.intekom.com/scotland/cookbook/146.htm http://www.authorizedcom.com/lines_trunks.asp http://www.beagle-ears.com/lars/engineer/telecom/bizphone.htm http://www.zvon.org/tmRFC/RFC3064/Output/chapter4.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI
Max TNT's are pretty cheap they'll need to price it accordingly. On Thu, 28 Apr 2005, David Josephson wrote: Your reference picture is fine ... but note that Asterisk can be the TDM/VoIP gateway, particularly when Digium releases their DS3 card (644 voice channels!) working, a lot more cheaply than a standalone box from some hardware vendor. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static and echo on PRI
Mark, Call your upstream and ask for good echo can, to start, issue a trouble ticket regarding static and echo on your T1. -m On Tue, 26 Apr 2005, Mark Johnson wrote: Andrew Kohlsmith wrote: On April 26, 2005 06:19 pm, Mark Johnson wrote: Does anyone have some suggestions on how to get rid of this static on my Digium card? I am supposed to go live tomorrow night and will get shot if it's like this!! Lack of planning on your part does not constitute an emergency on our part. There were a number of suggestions given to you over the past week or so and a great number of them (including some given by myself) have gone unanswered. Perhaps you should read over this thread and make sure you haven't missed anything. -A. Um... If you read my orginal post, this was unplanned as I had a Cisco hardware failure. I have been working on building Asterisk for over 6 months and don't have the luxury of forking out over $5,000 for a test T1. I also have noticed that in looking through this particular thread that I have never seen your name in it. Just double checked the archives and, nope, you aren't there... I have tried every suggestion and replied my results. If you don't have any facts to share, please don't bother. I am desperate and don't have alot of time left and am begging for the list's advice. I left probably the largest post this month with EXACTLY what I have tried, the results, debug information, etc... I have removed drivers, swapped cards, changed IRQ's... I am open to any suggestions. If you tell me to go buy a different card, I will do that. You guys know more about than I do. What do you suggest, exactly? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static and echo on PRI
ask your upstream. On Wed, 27 Apr 2005, Mark Johnson wrote: Andrew Kohlsmith wrote: Try these things: Software: - don't play with gains on PRI or T1 unless you have echo or too loud/quiet. Static isn't caused by screwy gains and on digital circuits it technically shouldn't ever need to be adjusted - turn echocancel off for now - I notice you've got span=1,0,0 -- if you're talking to the telco make sure you're synchronizing the clock to them. Use span=1,1,0. - remove all modules except those absolutely necessary - Have you tried span 2, 3 or 4 instead of 1? Also is this a *stock* kernel or some distro-enhanced version? Grab a stock kernel of the same version from ftp.kernel.org. Finally, don't use the agressive canceller unless you REALLY can't get rid of it any other way (I seem to have very good performance with MARK2, using the MMX-friendly implementation (zconfig.h) and making sure my CFLAGS for the zaptel code was optimized for my processor (-march=pentium4). Also see http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html. Hardware: - *remove* the TDM22P from the system. Don't just unload the modules. - pull the TE405P out and put it in another (not same motherboard) system. I've seen this clean things up several times. Wetware: It's getting a little late for this now, but you paid for support from Digium when you bought the card; You might want to give them a call. Unfortunately I don't think this is an issue they will be able to solve over the phone, and their likely recommendation would be to replace the system. I'd love to know what they do find, if you try this route. Again, my apologies, for blasting you; I had you mixed up with someone else. -A. This is perfect stuff!!! Thank you!! I actually pulled the TDM22P today, removed all of those drivers and get the same results. I have built another box and am installing asterisk as we speak. I tried the span=1,1,0 with the same results and have been running that line for a day now. What I find strange is this... If I speak at a normal tone, it sounds OK. I still get static noise when the other person speaks. If I talk louder, I start to get what sounds like a partial echo. If I yell, I get a definite echo. Have not tried a different slot on the quad, will try that tomorrow. When monkeying with the echo cancel, I never really noticed a difference. I would even reboot the machine between changes to see if it made a difference. I am running this on Fedora Core 1. I will try any OS you recommend, but I have always had great luck with RH type distro's. I keep 400 and 500 day uptimes on those machines and they run many, many services. Uptimes would be higher but it seems whenever I find a good place to work, they close up or I move. Admittedly, I don't use RPM's for the core services, I typically compile those myself. I also shut down every module and service I don't need. I did alot of reading and it seemed like Digium cards were the real deal and I also found many users that had luck with the same setup. Should I try a different approach/OS/system? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static and echo on PRI
Mark, Call your local Telco and issue a Trouble Ticket and specifically state that you have Static and Echo on your line and need assistance. Most likely, they can give you Echo Can for free. Call Manager has always produced better quality sound. -m On Wed, 27 Apr 2005, Mark Johnson wrote: Matt Klein wrote: ask your upstream. Not sure what you mean. This T1 is in good working order with a different system. Do you mean call the telco or Digium? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK PROGRAMER
$4,172.38 USD and I'll programin anything you want for asterisk server. On Sat, 23 Apr 2005, Franz wrote: PLEASE CAN SOMBODY HELP ME PROGRAMIN AN VoIP ASTERISK SERVER Atentamente, Franz Schuverer Arrue GLOBAL GROUP, INC. www.telefoniaglobal.net [EMAIL PROTECTED] Tel. (504) 221-4062 (Honduras Tel. (507) 322-2259 (Panamá) Tel. (866) 978-0976 (U.S.A.) CONFIDENCIALIDAD. El contenido de esta comunicación, así como el de toda la documentación anexa, es confidencial y va dirigido únicamente al destinatario del mismo. En el supuesto de que usted no fuera el destinatario, le solicitamos que nos lo indique y no comunique su contenido a terceros, procediendo a su destrucción. CONFIDENCIALITY. The content of this communication and any attached information is confidential and exclusively for the use of the addressee. If you are not the addressee, we ask you to notify to the sender and do not pass its content to another person, and please be sure you destroy it. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Sábado, 23 de Abril de 2005 11:00 a.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 9, Issue 209 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Cisco 7960 won't register as SIP device (List Receiver) 2. Re: if outgoing call fails with provider 1 then auto try provider 2 (Thomas Miller) 3. Re: if outgoing call fails with provider 1 then auto try provider 2 (Thomas Miller) 4. RE: Cisco 7960 won't register as SIP device (Robert Webb) 5. RE: Re: Hotel billing in IPSwitchBoard (Mathew McKernan) 6. Re: Hotel billing in IPSwitchBoard (Steve Rawlings) 7. RE: Cisco 7960 won't register as SIP device (Robert Webb) 8. RE: Cisco 7960 won't register as SIP device (List Receiver) 9. Re: Quadbri bristuff: can * respond only to 1MSN and leave 1 number to other ISDN phones ? (Michiel van Baak) 10. Re: Hotel billing in IPSwitchBoard (tgj) 11. RE: Hotel billing in IPSwitchBoard (Chris Mason (Lists)) 12. [Fwd: FW: [Asterisk-Users] IAX help] (Michael DiMartino) 13. Re: Re: Hotel billing in IPSwitchBoard (tgj) 14. Re: OctoBRI and 2.6kernel (Michael Bielicki) 15. Re: [Fwd: FW: [Asterisk-Users] IAX help] (Peter Bowyer) 16. Re: Re: Hotel billing in IPSwitchBoard (David John Walsh) -- Message: 1 Date: Sat, 23 Apr 2005 08:23:32 -0700 From: List Receiver [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii The DNS servers are valid. I configured the phone via .cnf files. The following are the sip.conf and sipMAC.cnf files. [tycisco] type=friend username=username secret=secret qualify=200 ; Qualify peer is no more than 200ms away nat=yes ;insecure=no host=dynamic; This device registers with us ;defaultip=24.18.147.95 canreinvite=no context=fullaccess dtmfmode=inband ;mailbox=101 disallow=all allow=ulaw allow=alaw allow=g729 .cnf: # SIP Configuration File (start) # Proxy Server proxy1_address: asterisk.mastermindpro.com proxy2_address: proxy3_address: proxy4_address: proxy5_address: proxy6_address: # Line 1 Settings line1_name: tycisco ; Line 1 Extension\User ID line1_displayname: 101 ; Line 1 Display Name line1_authname: username ; Line 1 Registration Authentication line1_password: secret ; Line 1 Registration Password # Line 2 Settings line2_name: ; Line 2 Extension\User ID line2_displayname:; Line 2 Display Name line2_authname: UNPROVISIONED ; Line 2 Registration Authentication line2_password: UNPROVISIONED ; Line 2 Registration Password # Line 3 Settings line3_name: ; Line 3 Extension\User ID line3_displayname:; Line 3 Display Name line3_authname: UNPROVISIONED ; Line 3 Registration Authentication line3_password: UNPROVISIONED ; Line 3 Registration Password # Line 4 Settings line4_name: ; Line 4 Extension\User ID line4_displayname:; Line 4 Display Name line4_authname: UNPROVISIONED ; Line 4 Registration Authentication line4_password: UNPROVISIONED
Re: [Asterisk-Users] ASTERISK PROGRAMER
The funniest part is, he thought I was serious. I'd be dumb if I didn't at least charge $4,172.39 USD for the job. On Sat, 23 Apr 2005, Gary Stimson wrote: On Saturday 23 April 2005 19:23, Bob Goddard wrote: On Saturday 23 April 2005 19:13, Matt Klein wrote: $4,172.38 USD and I'll programin anything you want for asterisk server. You are too stupid for the job. Quoting the 1200-line long Asterisk Digest message in your reply and adding one single line to it, where you just insult someone who was making a joke and add nothing of value is also stupid. People who live in glass houses shouldn't throw stones... Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] US$200 bounty for * paging feature
cool. see, no need to fight anyone. you people are crazy. luf... On Wed, 20 Apr 2005, trixter http://www.0xdecafbad.com wrote: you did a great parody of him completly ignoring what I was saying and going off on something unrelated to what I say just to get MS bashing in. Gotta love people who disregard what is said thinking that it has to be all or nothing. You say that in some way a company did something that is good beyond themselves and all of a sudden people attack you for saying that everything the company did is great, which was never said. I wonder what makes people snap that way. Is it sheer stupidity and inability to read or do they live in a total fantasy land. Now to make this more asterisk, I will be releasing code within a week that is a better than festival TTS engine. Caching support, better than speek and spell v1.1 voice, infact the engine supports a few languages, male and female speakers and even US UK english dialects (as well as a couple dialects of spanish and a few other languages). On Wed, 2005-04-20 at 15:36 -0400, Race Vanderdecken wrote: Wow! What a great fight! Let me egg you guys on. Furthermore, (if you knew your history) MS had been doing funny things with DOS / and windows to make it difficult for other windowing systems and DOS clones to work with MS-DOS / Windows, further cementing their market dominance. As someone who worked under DOS. And by under I mean we loaded first, then loaded DOS on top of us so DOS would make the pre-NETBIOS world calls and file calls to us. And as one of the Original Windows 1.x, 2.x, 3.x, 95, 98, NT, Windows 2000, XP developers I can tell you some stories. Neither DOS nor MS ever did anything funny to trick anybody. The Code was just poor code. Unless you actually meet and worked with Aaron, one of the original MS DOS guys, you have a clue. Come on. Does anyone really think that a developer would try to cheat people? It was those business clowns who lied; not the developers. Why is it that the conspiracy guys are all lousy developers or spaceship probed Red Necks? Long live Linux! Screw Apple. I hope MS goes broke. Race the tyrannical ludite Vandedecken http://en.wikipedia.org/wiki/Luddite -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed Sent: Wednesday, April 20, 2005 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] US$200 bounty for * paging feature On Wed, Apr 20, 2005 at 09:01:56AM -0700, trixter http://www.0xdecafbad.com said: On Wed, 2005-04-20 at 09:36 -0400, Walt Reed wrote: On Tue, Apr 19, 2005 at 06:24:09PM -0700, trixter http://www.0xdecafbad.com said: as a whole. I enjoy cheap computers, if it were not for microsoft creating windows, making computers easier to use for everyone, the mass production and highly competitive hardware market would not exist. If that didnt happen the $300 computer of today would likely not exist, and if it did it would cost more like computers did 20 years ago, $2000+ for a bare system. rantmode Um, that's total bullshit. Low computer prices and ease of use would have existed if MS was never around. You completely dismiss billions of man hours of hard work by those outside MS making advances in hardware and software around the world. To make a statement like that, you show a total lack of knowledge of the industry. and hoiw many operating systems were so popular during the 80s and early 90s? What operating system shipped on almost every computer during that period? BTW, in the 80's, it wasn't windows - it was DOS (I know, well before your time.) Again, nobody could really compete with the IBM / MS / compaq x86 platform dominance, so the ONLY real choice on that platform was Dos, although there were a few specialty OS's and extensions (OS/2, QNX, Desqview/X, etc.) I realize you wouldn't know about them, comming into the game rather late. It wasn't until Windows 3.1 in the early 90's that there was a relativly stable (if you could call it that) windowing system from MS (despite that other companies had been doing it for many years.) Bundling and restrictive contracts made it impossible to compete. Furthermore, (if you knew your history) MS had been doing funny things with DOS / and windows to make it difficult for other windowing systems and DOS clones to work with MS-DOS / Windows, further cementing their market dominance. I dont think I lack understanding of the industry I think that I remember clearly that windows was shipped on that, I think that whether or not it resulted in an anti-trust conviction microsoft did make it easier for people to use computers and thus more sold. Again, your lack of experience with and knowledge of other OS's shows otherwise. I am sorry that you are so bigioted to think that other operating systems dominated the market during that period, and cant accept that windows was the #1 operating system by a clear margin in terms of installed systems. Did I say they
Re: [Asterisk-Users] US$200 bounty for * paging feature
stop wasting my bandwidth plz On Wed, 20 Apr 2005, trixter http://www.0xdecafbad.com wrote: On Wed, 2005-04-20 at 22:14 -0500, Dan Perik wrote: Michael D Schelin wrote: Ok you guys enough. The debate will go on forever. Agreed! At the risk of wasting bandwidth myself Please, guys stop wasting my precious bandwidth. If you want to private message your flames, great but leave this list to Asterisk, please. Thanks! - Dan Interesting that so many people are coming out to say stop, even to reply to others saying stop and holding precious bandwidth up as the reason. I love your logic. To jump on the bandwagon stop waasting my bandwidth telling people to stop wasting your bandwidth. Its only fair. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] US$200 bounty for * paging feature
fight fight fight fight! On Wed, 20 Apr 2005, Race Vanderdecken wrote: Wow! What a great fight! Let me egg you guys on. Furthermore, (if you knew your history) MS had been doing funny things with DOS / and windows to make it difficult for other windowing systems and DOS clones to work with MS-DOS / Windows, further cementing their market dominance. As someone who worked under DOS. And by under I mean we loaded first, then loaded DOS on top of us so DOS would make the pre-NETBIOS world calls and file calls to us. And as one of the Original Windows 1.x, 2.x, 3.x, 95, 98, NT, Windows 2000, XP developers I can tell you some stories. Neither DOS nor MS ever did anything funny to trick anybody. The Code was just poor code. Unless you actually meet and worked with Aaron, one of the original MS DOS guys, you have a clue. Come on. Does anyone really think that a developer would try to cheat people? It was those business clowns who lied; not the developers. Why is it that the conspiracy guys are all lousy developers or spaceship probed Red Necks? Long live Linux! Screw Apple. I hope MS goes broke. Race the tyrannical ludite Vandedecken http://en.wikipedia.org/wiki/Luddite -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed Sent: Wednesday, April 20, 2005 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] US$200 bounty for * paging feature On Wed, Apr 20, 2005 at 09:01:56AM -0700, trixter http://www.0xdecafbad.com said: On Wed, 2005-04-20 at 09:36 -0400, Walt Reed wrote: On Tue, Apr 19, 2005 at 06:24:09PM -0700, trixter http://www.0xdecafbad.com said: as a whole. I enjoy cheap computers, if it were not for microsoft creating windows, making computers easier to use for everyone, the mass production and highly competitive hardware market would not exist. If that didnt happen the $300 computer of today would likely not exist, and if it did it would cost more like computers did 20 years ago, $2000+ for a bare system. rantmode Um, that's total bullshit. Low computer prices and ease of use would have existed if MS was never around. You completely dismiss billions of man hours of hard work by those outside MS making advances in hardware and software around the world. To make a statement like that, you show a total lack of knowledge of the industry. and hoiw many operating systems were so popular during the 80s and early 90s? What operating system shipped on almost every computer during that period? BTW, in the 80's, it wasn't windows - it was DOS (I know, well before your time.) Again, nobody could really compete with the IBM / MS / compaq x86 platform dominance, so the ONLY real choice on that platform was Dos, although there were a few specialty OS's and extensions (OS/2, QNX, Desqview/X, etc.) I realize you wouldn't know about them, comming into the game rather late. It wasn't until Windows 3.1 in the early 90's that there was a relativly stable (if you could call it that) windowing system from MS (despite that other companies had been doing it for many years.) Bundling and restrictive contracts made it impossible to compete. Furthermore, (if you knew your history) MS had been doing funny things with DOS / and windows to make it difficult for other windowing systems and DOS clones to work with MS-DOS / Windows, further cementing their market dominance. I dont think I lack understanding of the industry I think that I remember clearly that windows was shipped on that, I think that whether or not it resulted in an anti-trust conviction microsoft did make it easier for people to use computers and thus more sold. Again, your lack of experience with and knowledge of other OS's shows otherwise. I am sorry that you are so bigioted to think that other operating systems dominated the market during that period, and cant accept that windows was the #1 operating system by a clear margin in terms of installed systems. Did I say they dominated? No. Please work on your reading comprehention. There was competition on the OS front, but it's hard to knock out the market leader, and impossible when they won't play fairly (legally.) I have worked for over 10 years in the software development industry and Then you entered the industry far too late to know the real history of computing, have read too many MS revisionist history books, or were hiding under a rock. I started using computers in 1976. I dont think I entered too late. As for reading MS revisionist history books, no but I think that you have been readiung too many anti-MS revisionist history books. The popularity of a personal computer in the home was not made with cp/m it was not made with coherent (a unix for the pc before linux was around). It was not made by os/2, it was not made by any mac. Computers did not fully become so incredibly popular until windows. look at any historical sales reports and see when the numbers started increasing dramatically. Again, bundling,
Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card
Kevin, Keep in mind that according to Wiki there are no DSP's on the board. -m On Tue, 12 Apr 2005, Kevin P. Fleming wrote: [EMAIL PROTECTED] wrote: In other words, a PCI-based co-processor would double the PCI bus bandwidth necessary. And with a latency-sensitive product like voice, bus contention is not something you want to add to! :) It only 'doubles the bandwidth required' when compared to a single-board solution, which does not exist. When compared to doing the transcoding and echo can in the host CPU, it would be a major win :-) Also, keep in mind that a DS3 is _only_ 45 megabits per second. Any PCI bus (even lowly 33MHz 32-bit PCI) can easily handle 90 megabits per second of traffic. People looking a DS3 cards are also likely to deploy them in servers with multiple independent PCI buses, which would then allow for even more bandwidth. The mind boggles at the possibilities! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card
Kevin, Mmm. Yep. -m On Tue, 12 Apr 2005, Kevin P. Fleming wrote: Matthew Boehm wrote: So, no hardware encoding on this beast? The announcement on the website makes no mention of transcoding, echo cancellation or toast-and-jam making, so at this time, no, there is no hardware transcoding apparently included. (Besides, would you really want a board that could only ENcode? G) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Number Ports
One of our local carriers charges 17 cents per ported DID MRC, no port/non recurring charges. I've seen in the neighborhood of $15 per 10 ported numbers as an LSR charge from other carriers NRC.. and as low as 5 cents MRC per Month. I've also seen cases with no MRC per DID per month, but an NRC per number. -m On Thu, 7 Apr 2005, Damon Estep wrote: Anyone out there (in the US) using a CLEC to do third party local number ports? Let me be more specific; Our inbound calls come in via inbound only PRIs from a local CLEC, our outbound calls go via SIP termination to a wholesale VoIP carriers softswitch. On the inbound numbers we use the carrier of record is the CLEC that we buy the PRI from, not us. When we bring a number on to our system via local number portability the number is actually ported to the CLEC that provides us the wholesale PRI. This is know as a third party LNP. Anyone doing it now? The real questions is what are you paying per number port? We have no reference for what this should cost and therefore do not know if the proposed rate is competitive and fair. Any input appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Discussion Forums provided by Digium
uNF On Mon, 4 Apr 2005, Kevin P. Fleming wrote: The recent discussions about mailing lists vs. forums have resulted in Digium management deciding to offer a forum site on a provisional basis, to determine if it will benefit the community. You will find a brand-new set of phpBB forums at forums.digium.com. Membership and posting are open to the public. Note that Digium, Inc. is _not_ officially participating in the forums, nor will the forums be used as a technical support channel for Digium products. Any Digium employees who choose to participate in the forums will do so on their own and not as representatives of the company. To those who are vehemently demanding that forums be provided, here is your chance to show us how you think they will work more effectively than the mailing lists do. We are willing to let the forums run for a period of 90 days, at which time we will re-evaluate their effectiveness and any problems or issues that they have caused. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a way to get inserted into an LEC's CLIDB? (fwd)
- Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. -- Forwarded message -- Date: Tue, 22 Mar 2005 19:16:09 -0800 (PST) From: Matt Klein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Is there a way to get inserted into an LEC's CLIDB? NeuStar also offers CNAM db services, but VeriSign pays you for your cnam listings as they receive reciprocal compensation for their databases, probably charging rbocs, clecs etc per query.. I'm not sure about NeuStar or how they handle this, but I'm almost positive that they provide cnam updates as well... I'd look into both to get comparitive pay-outs. One of the two may also have limits, meaning you'll need to list 50+ or 500+ lines.. and not just 10. I'm not sure on that either. Verisign seemed pretty easy going in that respect. Simple stuff. If you leave a voicemail for verisign sales, they generally contact you within 30 min during business hours. Don't bother filling out the online inquiry page it's a dead end. -m - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Tue, 22 Mar 2005, Kevin P. Fleming wrote: Tom Samplonius wrote: I had be using a group of two PRIs for more than a year on a Nortel PBX. After I started testing with Asterisk through a AS5300 gateway, I quickly noticed that I could present any calling number. Yes, we all know we can do that (and do it every day). The poster's question was not about that, though. Presumably he has numbers assigned by a provider that does not provide CNAM database records, and he wants to get his number listed in the master CNAM database so that a proper calling name will show up. Entirely different situation :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a way to get inserted into an LEC's CLI DB?
Verisign, CNAM http://www.verisign.com/products-services/communications-services/intelligent-database-services/cnam-calling-name-database/page_001662.html Look there. -m - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Tue, 22 Mar 2005, Jed Stafford wrote: We offer that service for our termination customers, however we can only provide it for (206) area code numbers. So what we find is people who don't care as much about the number and more about their callerid lookup such as businesses and call centers opt to utilize it. We can even change the name with 1 business days notice, it's pretty cool. -Jed --- Robert Goodyear [EMAIL PROTECTED] wrote: Robert Goodyear wrote: Does anyone know if there's a service out there to -- for a fee -- inject our DID into the LEC's CLI database so a called party gets our associated name? No, only if the LEC servicing the number offers it to you. It is the responsibility of the operator running the switch that the number routes to via SS7 to manage the LIDB, CNAM, ALI and other database entries related to that number. I wonder if that's an exploitable space given the oncoming avalanche of VoIP adopters... some sort of gateway subscription service to provide a batch feed to all the individual LECs for a fee. I'd certainly pay x dollars a month per DID to have a name better than Los Angeles Call identifying my business to the called party, especially since I'm more than an hour south of LA and *not* being in LA is a specific business differentiator for us. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a way to get inserted into an LEC's CLIDB?
NeuStar also offers CNAM db services, but VeriSign pays you for your cnam listings as they receive reciprocal compensation for their databases, probably charging rbocs, clecs etc per query.. I'm not sure about NeuStar or how they handle this, but I'm almost positive that they provide cnam updates as well... I'd look into both to get comparitive pay-outs. One of the two may also have limits, meaning you'll need to list 50+ or 500+ lines.. and not just 10. I'm not sure on that either. Verisign seemed pretty easy going in that respect. Simple stuff. If you leave a voicemail for verisign sales, they generally contact you within 30 min during business hours. Don't bother filling out the online inquiry page it's a dead end. -m - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Tue, 22 Mar 2005, Kevin P. Fleming wrote: Tom Samplonius wrote: I had be using a group of two PRIs for more than a year on a Nortel PBX. After I started testing with Asterisk through a AS5300 gateway, I quickly noticed that I could present any calling number. Yes, we all know we can do that (and do it every day). The poster's question was not about that, though. Presumably he has numbers assigned by a provider that does not provide CNAM database records, and he wants to get his number listed in the master CNAM database so that a proper calling name will show up. Entirely different situation :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STOP NOW not responding
Try ps -auxww, find the process and kill it. Or, if you're on a system that supports killall, just killall asterisk. safe_asterisk should restart the * process automatically, or if you're not running safe_asterisk, then just start the process after you've killed it.. I have no idea why STOP NOW doesn't work every time, but I experienced the problem with a 2004 CVS-HEAD update. -Matt - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Tue, 8 Mar 2005, Wiley Siler wrote: Has anyone had any new information about STOP NOW hanging? I am using [EMAIL PROTECTED] 0.6 and today my system just stopped responding. I issued the usual STOP NOW command and it just returns to the CLI. I have found a lot of info regarding others having this happen but nothing that addresses resolution. I do not do a lot of calls so I am not sure why this would occur. Any info woud be appreciated. I am continuing to search the Wiki and list so if I find something I will post it. Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI HDLC Abort (6) Errors
ztcfg - works, too.. after a timing source change... power cycle works. - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Fri, 4 Mar 2005, Steven Critchfield wrote: On Fri, 2005-03-04 at 15:27 -0700, Tom wrote: Hello, I have searched and searched, and come up with nothing. I am running Asterisk with a wcte110p configured for t1. Our PRI is staying up, and we can make calls however our service provider's logs are flooding with errors and we are getting lots of HDLC Abort (6) on Primary D-Channel Errors. Our provider says it looks like our box is trying to be the master timer on the circuit (which is not correct they are providing the timing) we have tried both span=1,1,0,esf,b8zs and span=1,0,0,esf,b8zs in zaptel.conf both produce the same problems. The problem is not in Asterisk per se as the errors start happening as soon as I modprobe the driver and run ztcfg. As soon as the circuit comes up the errors start on the provider's end. Did you make sure to power cycle afterwords? Sometimes the zap cards don't change critical settings like timing once configured. We are running CVS Asterisk/zaptel/libpri from March 2nd 2005 on Fedora Core 3 fully patched as of last night, I was thinking the problem was with the 2.6 kernel getting preempted and therefore the driver not being able to do its timings right, however fc3's kernels have preemption disabled by default. Does Digium hardware really need/expect a real time OS to run properly? Like I said previously I think the problem is in the driver itself not in asterisk. Any help would be appreciated, and I can code a bit in c so if someone can point me in the right direction I might be able to fix it myself... You probably want to dump the FC kernel like a bad habit. Get a plain vanilla kernel and see if that fixes your problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simulated dialtone like in other PBX
from the console, show modules - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Sun, 20 Feb 2005, Anton Krall wrote: That app_disa is new to me... Is there a list of available apps? Im still quite new to asterisk but I guess you can find out which apps you have by using a show applications but my question would be more of how to make new apps or download/get new ones, is this possible? Also, is there a list of command that can be used in a dialplan or are they just apps like dial()? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Sent: Domingo, 20 de Febrero de 2005 04:48 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simulated dialtone like in other PBX Anton Krall wrote: I think it would be your last suggestion.. When I pickup the phone I hear a tone, the sip phone box tone... Then I hit 9, no tones :) and enter the whole phone number and it starts to ring on the other side.. So no outside dialtone get heard ever.. I was wondering if it could be possible to make it so that after hitting 9.. The tone would change to something else letting the user know that they are dialing on an outside line. Yes, you can do this, stick a extension in your dial plan for 9, then point that to app_disa... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simulated dialtone like in other PBX
And go here: http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Sun, 20 Feb 2005, Anton Krall wrote: That app_disa is new to me... Is there a list of available apps? Im still quite new to asterisk but I guess you can find out which apps you have by using a show applications but my question would be more of how to make new apps or download/get new ones, is this possible? Also, is there a list of command that can be used in a dialplan or are they just apps like dial()? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Sent: Domingo, 20 de Febrero de 2005 04:48 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simulated dialtone like in other PBX Anton Krall wrote: I think it would be your last suggestion.. When I pickup the phone I hear a tone, the sip phone box tone... Then I hit 9, no tones :) and enter the whole phone number and it starts to ring on the other side.. So no outside dialtone get heard ever.. I was wondering if it could be possible to make it so that after hitting 9.. The tone would change to something else letting the user know that they are dialing on an outside line. Yes, you can do this, stick a extension in your dial plan for 9, then point that to app_disa... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
Agreed, Steve. Iq, Maybe it is for your voice, but speech to text is a long ways away from being as advanced as you think it is. Check out dragon speek, and see what it takes to train a voice... -m On Sun, 13 Feb 2005, Steve Underwood wrote: Iqbal wrote: Hi I dont know jack about speech recognition, however since this topic came up anyonw know how spinvox do speech ercognition, in fact its so good it converst the speech to text and sends the voicemail as a SMS, I think a awesome addone to the sms module in asterisk. If it works really well, there is probably a human operator involved. A number of systems that try to look automated actually rely on human operators. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with perl script/agi for ringback
kill the line breaks?
On Sun, 6 Feb 2005, taf taffey wrote:
Hi,
I'm trying to write a simple perl script that will run
the following:
Action: Originate
Channel: local/[EMAIL PROTECTED]/r/n
Exten: 1234
Context: callback
Priority: 1
Extensions.conf
exten = 500,1,agi,callback.pl
callback perl script:
use Net::Telnet ();
$mgrUSERNAME='fred';
$mgrSECRET='bloggs';
$server_ip='127.0.0.1';
$tn-print(Action: originate\nExten: 1234\nContext:
user\nChannel: local/[EMAIL PROTECTED]/r/n\nPriority:
1\nCallerid: 1234\n\n);
$tn-waitfor('/Event: Newchannel.*/') or die Unable
to determine call status, $tn-lastline;
# wait for asterisk to process
$tn-print(Action: Logoff\n\n);
I'm not a programmer (as u can probably tell) so any
pointers would be much appreciated.
Cheers,
Taff.
___
ALL-NEW Yahoo! Messenger - all new features - even more fun!
http://uk.messenger.yahoo.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and Cops knocking on my door
Try dialing 591-2079 and see if you're trying to make a call to 91-2079 instead of 591-2079. -m On Wed, 2 Feb 2005, Andrew Niemantsverdriet wrote: Hi, I am quite new to asterisk so I am not sure what is needed to figure out this problem. If more information is needed and not provided I will gladly provide it. I have a very basic asterisk setup. 1 x100p card and a grandstream handytone 286. I can make calls fine to most phone numbers from the handytone device the trouble seems to come when I dial this number 591-1079. It puts me through to the local 911 dispatch. Causing the police to show up at my doorstep and check to make sure everything is alright. I can see why I think; 5 911 079. But I don't understand why it is being handled this way. Can somebody offer me some guidance on how to get this to stop? TIA _ /-\ ndrew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and Cops knocking on my door
looks like an ignorepat problem on the first *number* (single) dialed (i.e., trying to ignore the number 9 on an outbound call.) try to make a call to 591-2079. - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Wed, 2 Feb 2005, Andrew Niemantsverdriet wrote: Hi, I am quite new to asterisk so I am not sure what is needed to figure out this problem. If more information is needed and not provided I will gladly provide it. I have a very basic asterisk setup. 1 x100p card and a grandstream handytone 286. I can make calls fine to most phone numbers from the handytone device the trouble seems to come when I dial this number 591-1079. It puts me through to the local 911 dispatch. Causing the police to show up at my doorstep and check to make sure everything is alright. I can see why I think; 5 911 079. But I don't understand why it is being handled this way. Can somebody offer me some guidance on how to get this to stop? TIA _ /-\ ndrew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys PAP2 / RT31P2 + multiple G.729 calls
of course it won't. neither can the ata. they're cheap, it was a licensing decision. i look forward to v2. -m On Wed, 2 Feb 2005, Matthew Boehm wrote: Holy Crap I have just verified this! The linksys PAP2-NA will NOT SUPPORT 2 SIMULTANEOUS G729 CALLS! And I just got off the phone with some super-level technician at linksys and he said they knew this all along!! What bastards! Anyway, he told me they are comming out with the PAP2-NAv2 in a few months which WILL allow 2 simul G729 calls. -Matthew - Original Message - From: Leonardo Gomes Figueira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 01, 2005 12:21 PM Subject: [Asterisk-Users] Linksys PAP2 / RT31P2 + multiple G.729 calls Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the others. If I disable G.729 on sip.conf for both peers they can't establish a second call (ring but drop after answer). If there is allow=ulaw on sip.conf I can establish 1 G.729 call and 3 G.711 with RT31P2-NA (using three-way calling). In PAP2-NA if I mark Use Pref Codec Only and there is one call established, when I call the PAP2 it replies with 488 Not Acceptable Here. Thanks, Leonardo -- Leonardo Gomes Figueira [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and Cops knocking on my door
yep, post your conf. On Wed, 2 Feb 2005, AJ Grinnell wrote: post your dialplan from extensions.conf On Wed, 2 Feb 2005 14:15:28 -0700, Andrew Niemantsverdriet [EMAIL PROTECTED] wrote: Hi, I am quite new to asterisk so I am not sure what is needed to figure out this problem. If more information is needed and not provided I will gladly provide it. I have a very basic asterisk setup. 1 x100p card and a grandstream handytone 286. I can make calls fine to most phone numbers from the handytone device the trouble seems to come when I dial this number 591-1079. It puts me through to the local 911 dispatch. Causing the police to show up at my doorstep and check to make sure everything is alright. I can see why I think; 5 911 079. But I don't understand why it is being handled this way. Can somebody offer me some guidance on how to get this to stop? TIA _ /-\ ndrew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Terrible inbound call quality vs. outbound
Try voicepulse in a different area w/ 800 service. With 800 you can jump POPs. If you're stuck with local DID service w/ them, I feel sorry for you. I've seen several recent complaints, and have experienced my own problems, with voicepulse inbound service. They need to upgrade the POP (wherever you are) to add more bandwidth and/or more lines (depending on if you're getting choppy sound, or if you're getting fast busies, etc). Personally, I'm cancelling service in a week or two. Call them and complain, if you think it'll do any good. -m - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Tue, 1 Feb 2005, Robert Goodyear wrote: Oh I've tried all manner of packet shaping and QoS tagging... it's certainly not an issue with the ISP. I think Gene Willingham may have the right answer, that VoicePulse cannot handle the load. Anyone else have any thoughts? Maybe I need to find another IAX service provider to test a different DID in my area. /rg On Feb 1, 2005, at 3:52 PM, Miguel Ruiz Velasco Sobrino wrote: I've had similar problems but with dial-up modems. ISP's mantain large queues in the inbound side of your connection to maximize download speed, but that same hurts latency on your side. You may be saturating the BW and thus the queue makes it's job. Use the bw conditioner that is described in the advanced linux routing howto, in the cookbook, that is named a thing like the ultimate bw conditioner, fast downloads and uploads and blablabla. Modify it by putting the ports that the RTP or IAX stream pases, assigning them with a filter to the interactive class. Also don't forget to put the correct uplink and downlink values, or you will be putting a bw restrictor. The thing that is very weird is that only inbound calls are affected, I would think that both inbound and outbound calls were affected. --- [EMAIL PROTECTED] wrote: Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be jumbled up voice packets?) Miguel Ruiz Velasco __ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Voicemail ...
MySQL RealTime: Retrieve SQL: SELECT * FROM
voicemail_users WHERE mailbox = '201' AND context = 'default'
There is no column 'context'.
Add a column 'context' to your voicemail_users table, default value of
'default', make sure the column is filled with 'default' as a value. Or
edit the source code to modify the SQL call, to ignore querying
the AND context = '.
Do this and your login should work.
-
Yeah, we rocked the vote all right. Those little
bastards betrayed us again.
- Hunter S. Thompson on the 2004 election.
On Tue, 18 Jan 2005, Vamsi Pottangi wrote:
Hi,
Realtime SIP and Extensions are working fine but facing some problems
with Voicemail.
Added an entry to extconfig.conf
voicemail = mysql,asterisk,voicemail_users
Created the corresponding table and an entry for mailbox 201.
This is also reflected in the CLI as shown below.
CLI realtime load voicemail mailbox 201
Column Name Column Value
uniqueid 1
customer_id 201
mailbox 201
password 201
fullname Mailbox 201
stamp 20050118164309
CLI
When I try to log into the Voicemailmain, it cribs for incorrect login
as shown below.
Where am I going wrong ?
Jan 18 17:49:12 DEBUG[5502]: MySQL RealTime: Retrieve SQL: SELECT * FROM
extensions_table WHERE exten = '8500' AND context = 'default' AND
priority = '1'
Jan 18 17:49:12 DEBUG[5502]: MySQL RealTime: Everything is fine.
Jan 18 17:49:12 DEBUG[5502]: MySQL RealTime: Retrieve SQL: SELECT * FROM
extensions_table WHERE exten LIKE '\_%' AND context = 'default' AND
priority = '1' ORDER BY exten
Jan 18 17:49:12 DEBUG[5502]: MySQL RealTime: Everything is fine.
Jan 18 17:49:12 VERBOSE[5502]: -- Executing VoiceMailMain
(SIP/vamsi-0c3c, ) in new stack
Jan 18 17:49:12 DEBUG[5502]: Scheduling timer at 160 sample intervals
Jan 18 17:49:12 VERBOSE[5502]: -- Playing 'vm-login' (language 'en')
Jan 18 17:49:12 DEBUG[5502]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 101:
Found
Jan 18 17:49:14 DEBUG[5502]: Manager received command 'Command'
Jan 18 17:49:14 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:14 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:16 DEBUG[5502]: MySQL RealTime: Retrieve SQL: SELECT * FROM
voicemail_users WHERE mailbox = '201' AND context = 'default'
Jan 18 17:49:16 DEBUG[5502]: MySQL RealTime: Everything is fine.
Jan 18 17:49:16 DEBUG[5502]: Scheduling timer at 160 sample intervals
Jan 18 17:49:16 VERBOSE[5502]: -- Playing 'vm-password' (language
'en')
Jan 18 17:49:17 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:17 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:19 VERBOSE[5502]: -- Incorrect password '201' for user
'201' (context = any)
Jan 18 17:49:19 DEBUG[5502]: Scheduling timer at 160 sample intervals
Jan 18 17:49:19 VERBOSE[5502]: -- Playing 'vm-incorrect-
mailbox' (language 'en')
Jan 18 17:49:22 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:22 DEBUG[5502]: Scheduling timer at 0 sample intervals
Thanks,
~Vamsi
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] Cisco 7690 Voicemail Problem
Yep that was going to be my suggestion -- though, I've had user reports that this doesn't always fix the problem in every case. Ask your users to turn down their volume, and the sound should stop coming out of the speaker. And as Paul wrote, a good longer term solution would be gain boosts.. -m On Thu, 30 Dec 2004, Ryan O'Connell wrote: On 30/12/2004 19:01, Paul A Brown wrote: Anyone? :-) If you turn down the volume on the phone slightly (Just one or two units) it goes away. I assume the output volume is overloading the phone and the DSP isn't clever enough to clip it. A longer term solution would be to boost the gain of whatever input you're using so that people don't have their phones turned up so loud. - Original Message - From: Matt Klein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 7:37 PM Subject: Re: [Asterisk-Users] Cisco 7690 Voicemail Problem a faint scratching sound of your voice coming out of the speaker? or loud and clear? I would say a medium crackly version..Actually its the voice from the vmail system ( ' The person at extension blah blah blah') So not too loud but not really clear either ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] Cisco 7690 Voicemail Problem
On Thu, 30 Dec 2004, Paul A Brown wrote: Anyone? :-) - Original Message - From: Matt Klein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 7:37 PM Subject: Re: [Asterisk-Users] Cisco 7690 Voicemail Problem a faint scratching sound of your voice coming out of the speaker? or loud and clear? I would say a medium crackly version..Actually its the voice from the vmail system ( ' The person at extension blah blah blah') So not too loud but not really clear either Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7690 Voicemail Problem
a faint scratching sound of your voice coming out of the speaker? or loud and clear? On Wed, 29 Dec 2004, Paul A Brown wrote: Happy New year to you all... I was wondering if anyone can help. I have a couple of 7690's working with the latest SIP image and they call to each other just fine. The problem I have is when I get someones * voicemail. If I have the handset in my hand and am about to leave a message, I get my voice coming out of the 7690 hands free speaker Any Ideas? Thanks Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Originating Network identity
focus on npa-nxx (area code-prefix) if the call is coming from a non-ported number, then http://telcodata.us/docs/queries.html may help you -- see the example files.. there are also a couple other sites out there.. but i've found this one to be my favorite thus far. -m On Sun, 26 Dec 2004, oi geli wrote: I am not sure if it is the right list for the post. Please excuse my lack of expertise, if it is a bad post. Is there anyway to detect the originating network identity of the call in Asterisk? For example, if the Asterisk gets a call from Cingular Network, is there anyway to find out that the call came from a Cingular subscriber. Thanks __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Originating Network identity
http://www.illuminet.com/docs/lidb/
-
I believe there are more instances of the abridgment
of the freedom of the people by gradual and silent
encroachments of those in power than by violent and
sudden usurpations. - James Madison
On Sun, 26 Dec 2004, Lyle Giese wrote:
That's good to get a general idea, but number portability only tells you
which carrier has the block. It does not let you know about specific
numbers :-{
Lyle
- Original Message -
From: Matt Klein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 26, 2004 5:11 PM
Subject: Re: [Asterisk-Users] OT - Originating Network identity
focus on npa-nxx (area code-prefix)
if the call is coming from a non-ported number, then
http://telcodata.us/docs/queries.html may help you --
see the example files..
there are also a couple other sites out there.. but i've
found this one to be my favorite thus far.
-m
On Sun, 26 Dec 2004, oi geli wrote:
I am not sure if it is the right list for the post.
Please excuse my lack of expertise, if it is a bad
post.
Is there anyway to detect the originating network
identity of the call in Asterisk? For example, if the
Asterisk gets a call from Cingular Network, is there
anyway to find out that the call came from a Cingular
subscriber.
Thanks
__
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Originating Network identity
More specifically, see the data sheet about lidb:
http://www.verisign.com/stellent/groups/public/documents/data_sheet/001944.pdf
You could go that route, or get a switch, or... there's a variety of other
options. But if you're looking for a full number lookup, you're looking
for lidb access..
-m
-
I believe there are more instances of the abridgment
of the freedom of the people by gradual and silent
encroachments of those in power than by violent and
sudden usurpations. - James Madison
On Sun, 26 Dec 2004, Lyle Giese wrote:
That's good to get a general idea, but number portability only tells you
which carrier has the block. It does not let you know about specific
numbers :-{
Lyle
- Original Message -
From: Matt Klein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 26, 2004 5:11 PM
Subject: Re: [Asterisk-Users] OT - Originating Network identity
focus on npa-nxx (area code-prefix)
if the call is coming from a non-ported number, then
http://telcodata.us/docs/queries.html may help you --
see the example files..
there are also a couple other sites out there.. but i've
found this one to be my favorite thus far.
-m
On Sun, 26 Dec 2004, oi geli wrote:
I am not sure if it is the right list for the post.
Please excuse my lack of expertise, if it is a bad
post.
Is there anyway to detect the originating network
identity of the call in Asterisk? For example, if the
Asterisk gets a call from Cingular Network, is there
anyway to find out that the call came from a Cingular
subscriber.
Thanks
__
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
Being optimistic, I think it's a great idea.. putting on the pessimistic hat, getting * to work under those conditions w/ the # of ports (48) you're discussing.. I think is probably your biggest headache. I wrote 4 other paragraphs about what I think, and deleted them. Interesting, let me know where you go with this. -m - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Greg Boehnlein wrote: On Wed, 15 Dec 2004, Matt Klein wrote: who said anything about a computer? :) computer, $$extra on both. may be less on the pm3 side due to resource needs. In the scenario I envision this being used in, there is no computer. The PM3 runs (On it's x86 w/ 4 or 16 megs of ram) a stripped down, embedded version of Linux + Asterisk. With a TE405P you need a PC to house the cards in. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
they've mentioned interest in making it a channel bank, really, FXS/FXO to SIP or IAX or another protocol, delivered via tcp/ip, and your input would be interesting regarding the hardware capabilities of the boxes. - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Bob Knight wrote: On Wed, 15 Dec 2004, Matt Klein wrote: 3) good luck getting the firmware source is the firmware source freely available, -- I've been asked by others. All the other (excellent, thought provoking) conversation aside, Jake Messenger from Portmasters.com has been granted a license by Lucent for ComOS. http://www.portmasters.com/pipermail/comos/2004-August/41.html That contains a link to the license the source is under. It isn't free as in GNU, but I don't think that really matters much. I had to give up following this list too closely, because it just sucks up too much time. But I did just stumble onto this thread about portmasters. I worked at Livingston and wrote the drivers on the portmasters. That source code is easy to find and even compiles on a linux box these days (we used to use SunOS). If you come up with anything interesting to do with the boxes, please let me know I may be able to help. Contact me off list is best. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
Ethernet Channel Bank Hmm. Caught my attention for more than 34 hours, you win. I'm getting 24 port carrier access channel banks for $100, Digium 4 Port Cards for about $800 (T400P) a card.. Meaning a blended cost of ~$12.50 per channel NRC. I can mux up 96 channels for a cost of $12.50 per channel all day long. And easily sell it at $25 to cover the cost of the box per port, T-1 channel per port, and channel bank per port non recurring costs. Why am I looking at this post? Your cost is higher. Why would I bother? Give me influence. Why would I think about buying a $400 2T portmaster for twice the price to achieve a lesser result as a 4 port and 4 T carrier access channel banks? 4 ports and 4 banks cost me about $1200, or about $12.50 NRC not including the associated hardware. Add on about 6 bucks a port. Even then, these costs still result in the same Hardware costs outside of the Channel Bank costs of a Portmaster. $400 is a rip for Channel Banks. The CLECs are already hurting and need a quick solution, an 8 year return plan isn't going to help anyone. Make it cheap and you'll win cash. And have we discussed gr303 for oversubscription capabilities which * supports? You need oversubscription capability, if you don't include this in your design, you will fail. Period. My two cents. I can have SS7 access pretty easily, can provide colo, and can provide a machine, with capability for testing if desired. I think I also have PM3s in inventory somewhere, email me. Lemme know. Dunno. I'm also interested in the DSP's ability to provide codec trans and echo can. Actually, I'm really concerned with that as a primary, everything else is kinda noise to me ATM. If you can get the original source for the comos you can probably get the layout of the cards, which means they can be reassembled for compactpci capability if that doesn't currently exist, which extends my interest. I am, however, interested in further comment on this thread, including PM3, Ethernet - DS0 bridging... continue discussion PLZ cPCI is my current undying interest. I have facilities if anyone wants to play. bkw input? blah from 1am. wherd. -m - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Sun, 12 Dec 2004, Greg Boehnlein wrote: On Fri, 10 Dec 2004, nik martin wrote: news.gmane.org wrote: Allied Telesyn VoIP Access Device http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf This is a 24-port FXS 1u device, conveniently presented as a single RJ-21 TELCO connector. yeah, but those are expensive as crap. i was thinking about something more competetive with a channel bank You know, if someone had some time on their hands, was good at hardware/software hacking and had the will, the old Livingston/Lucent PM3 platform would make an awesome 48 port IAX2 - PRI/T1 channel bank. Basically, the PM3 has 2 T1 ports that can be configured for ISDN PRI. The core of the system runs on an AMD x86 CPU. The plug in Modem cards have Lucent DSP's on them (up to 50 in a box). Flash size is 4 megs, and RAM is usually around 4 megs. That is still quite a bit of horsepower, and the boxes are under $400 now. The DSP's could be used for Codec Translation, if neccessary, or for echo cancellation. And, we can get access to the original Lucent ComOS Source code. Anyone game? :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
W/ Portmaster to Ether Portmaster Qty 2 T1$400 Portmaster Qty 2 T1$400 Carrier Access Qty 4 $400 Total $1200 W/ T400P to Ether T400P Qty 1 (4 T1)$800 Carrier Access Qty 4 $400 Total $1200 1) And T400P is already coded for. But Maximum Cards per Chassis is a prob. 2) Portmaster could potentially support more ports per cost I.E. 256-512 simultaneous being routed through one machine after translated to SIP, codec'd and shipped?. as opposed to 96-256 calls on one machine via 4 port cards 3) good luck getting the firmware source is the firmware source freely available, -- I've been asked by others. - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Matt Klein wrote: FXS vs FXO, listen. I really didn't mean to present that as a perspective, only a challenge to shoot down an inflated price model for FXS ports. If you mention FXO, I must mention $100 will not buy you what I am talking about. I am talking FXS ONLY. I am too, interested, in an Ethernet Channel Bank. I will not mention PAP2-NA's for 2 Line VOIP FXS ports at $56 standard retail each, nor will I mention SIP which comes standard with Asterisk to ship VOIP calls to, and which passes G.729 across a T-1 or DSL to a device which you can buy, such as a 7905, 7940, 7960 and I definitely won't point you to buy a 729 license from http://store.yahoo.com/asteriskpbx/asteriskg729.html. or 'borrow' one from elsewhere. Besides, GSM compression is pretty close. etc etc etc Big Difference between FXS and FXO. And yes, point made, rural fxo bonding could be more cost friendly with that type of a device.. as could already established lines. No installs etc. Let me know if you every come up with a $400 48 Port FXO device. With that, if as an FXO device, and not looking at FXS, the $400 is interesting.. considering only 2 points 1) Call Answered is at Channel Bank level 2) Call Delivered is only via Ethernet. PSTN - | PM | - * Otherwise for twice the cost, I can do 4 ports @ $800.. vs 2 ports at $400. Matching cost and supporting open source software. Otherwise this is a null and void topic. Same cost per T? Am I wrong here? -m - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Rich Adamson wrote: Ethernet Channel Bank Hmm. Caught my attention for more than 34 hours, you win. I'm getting 24 port carrier access channel banks for $100, Digium 4 Port Cards for about $800 (T400P) a card.. Meaning a blended cost of ~$12.50 per channel NRC. I can mux up 96 channels for a cost of $12.50 per channel all day long. And easily sell it at $25 to cover the cost of the box per port, T-1 channel per port, and channel bank per port non recurring costs. I'm also interested, but not from an I-can-buy-a-channel-bank-cheeper- then-you-can perspective. Rather, an ethernet channel bank would make it very easy to pick up a flexible number of pstn-fxo lines at remote locations where I already have Internet presence. I don't need any T1 cards to extend the reach into small towns and cities. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
good point, let me know too, 24 FXO less than $400. -m - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Andrew Kohlsmith wrote: On December 15, 2004 05:29 am, Matt Klein wrote: If you mention FXO, I must mention $100 will not buy you what I am talking about. I am talking FXS ONLY. You can't find an FXO channel bank for $100. FXS sure, FXS channel banks are a dime a dozen. FXO ports are damned expensive, especially if they have far-end disconnect supervision. I will not mention PAP2-NA's for 2 Line VOIP FXS ports at $56 standard retail each, nor will I mention SIP which comes standard with Asterisk to ship VOIP calls to, and which passes G.729 across a T-1 or DSL to a device which you can buy, such as a 7905, 7940, 7960 and I definitely won't point you to buy a 729 license from http://store.yahoo.com/asteriskpbx/asteriskg729.html. or 'borrow' one from elsewhere. For 2 ports the PAP2-NA's fine. Do you really want 12 of them, a switch and a wall full of wall warts (plus the associated cabling and rise in temp over ambient) that goes with this solution? Let me know if you every come up with a $400 48 Port FXO device. Hell let me know if you ever find a $400 24 port FXO channel bank that does far-end disconnect supervision! -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 12.50$ per port ???
Shoval, Interesting Mention. I agree, most people don't have CO exp. And I wish daily I had enough. Understand that what I mean by my e-mail is consumer side FXS ports, in broader terms, I mean, customer picks up a phone line, it signals a channel bank which signals *. 24 of those channels. Not channels equipped to Send Signal to the CO that a loop has been made.. meaning FXO. 24 FXS, $100. I don't deal with FXO since I deal w/ PRI.. and do not need FXO ports. My thought here was related to downstream customers.. which in the this world implies FXS. Talk was mentioned for DSPs.. Echo Can and Codec Management, this interests me because of the unique hardware requirements of the 4 port cards.. mainly in interrupts and CPU usage. Fit 10 cards to one system, that's 40 ports.. fit double density, that's 80.. this interests me on the cPCI platform. Others know what I think about this and why. Beyodn that, my point here is 1) Buy a T400P for $800 and 2) Buy 4 Carrier Access Channel Banks for $100 and 3) You'll have a 96 Downstream Port solution for $1200, meaning $12.50 a Port NRC. If you can find a cheaper *OR* easier solution, let me know. Because it'll save me money and you'll be my friend. ;) As for FXO, my best solution was Mainstreet Newbridge 3624s from Ebay for about $150-$300 a box w/ 12 port FXOs a while back. Then I moved to PRI for the capability of setting Caller ID easily (em was a pain in the..). wherd -Matt - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Shoval Tomer wrote: -Original Message- From: Matt Klein [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 15, 2004 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Ethernet Channel Bank idea Ethernet Channel Bank Hmm. Caught my attention for more than 34 hours, you win. I'm getting 24 port carrier access channel banks for $100, Digium 4 Port Cards for about $800 (T400P) a card.. Meaning a blended cost of ~$12.50 per channel NRC. I can mux up 96 channels for a cost of $12.50 per channel all day long. And easily sell it at $25 to cover the cost of the box per port, T-1 channel per port, and channel bank per port non recurring costs. Matt, A 24 port channel bank for $100 ??? Where? If you can get one that's working and has 8 FXO ports and 16 FXS ports, and is one of the brands that work well with * (support all or most features) for $100, I'll buy it from you for $200. Now, as to why we're interested in an Ethernet channel bank, you should keep in mind that some of the people on the list have no telco experience. And that's fine by me (especially since I'm one of them). You can manage a Cisco call manager, and other VOIP systems that originated from the IP side of the VOIP world without being a telco expert. VOIP systems that originated from the VO side, like ones from Panasonic, LG, etc. are cryptic at best to us non telco guys. We don't have the experience and knowledge necessary to setup channel banks (it's not like they have 24 RJ45 ports on them, right?) An Ethernet channel bank will probably be much easier for us to set up. I can understand and respect telco guys, who've setup channel banks for TDM PBXs for years not being afraid to try and hook it up with *. Me? I'm scared as hell Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
FXS vs FXO, listen. I really didn't mean to present that as a perspective, only a challenge to shoot down an inflated price model for FXS ports. If you mention FXO, I must mention $100 will not buy you what I am talking about. I am talking FXS ONLY. I am too, interested, in an Ethernet Channel Bank. I will not mention PAP2-NA's for 2 Line VOIP FXS ports at $56 standard retail each, nor will I mention SIP which comes standard with Asterisk to ship VOIP calls to, and which passes G.729 across a T-1 or DSL to a device which you can buy, such as a 7905, 7940, 7960 and I definitely won't point you to buy a 729 license from http://store.yahoo.com/asteriskpbx/asteriskg729.html. or 'borrow' one from elsewhere. Besides, GSM compression is pretty close. etc etc etc Big Difference between FXS and FXO. And yes, point made, rural fxo bonding could be more cost friendly with that type of a device.. as could already established lines. No installs etc. Let me know if you every come up with a $400 48 Port FXO device. With that, if as an FXO device, and not looking at FXS, the $400 is interesting.. considering only 2 points 1) Call Answered is at Channel Bank level 2) Call Delivered is only via Ethernet. PSTN - | PM | - * Otherwise for twice the cost, I can do 4 ports @ $800.. vs 2 ports at $400. Matching cost and supporting open source software. Otherwise this is a null and void topic. Same cost per T? Am I wrong here? -m - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Rich Adamson wrote: Ethernet Channel Bank Hmm. Caught my attention for more than 34 hours, you win. I'm getting 24 port carrier access channel banks for $100, Digium 4 Port Cards for about $800 (T400P) a card.. Meaning a blended cost of ~$12.50 per channel NRC. I can mux up 96 channels for a cost of $12.50 per channel all day long. And easily sell it at $25 to cover the cost of the box per port, T-1 channel per port, and channel bank per port non recurring costs. I'm also interested, but not from an I-can-buy-a-channel-bank-cheeper- then-you-can perspective. Rather, an ethernet channel bank would make it very easy to pick up a flexible number of pstn-fxo lines at remote locations where I already have Internet presence. I don't need any T1 cards to extend the reach into small towns and cities. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] list broken again?
?? list is working. -m - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Greg - Cirelle Enterprises wrote: Sure, why not? You know, like how your PHB emails you to let you know the mail server is down. -- Tracy Reedhttp://copilotcom.com PHB ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
who said anything about a computer? :) computer, $$extra on both. may be less on the pm3 side due to resource needs. - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Andrew Kohlsmith wrote: On December 15, 2004 06:13 am, Matt Klein wrote: W/ T400P to Ether T400P Qty 1 (4 T1) $800 Carrier Access Qty 4 $400 Total $1200 Plus computer. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
