Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!
On Tue, 2006-07-18 at 06:02 +1200, Matt Riddell (NZ) wrote: :) Which applications exist that have been disclaimed, well coded, are patent unencumbered and are not accepted? res_js for example, which in my experience on a more or less fair comparison (the javascript dialplan has more error control, better error checking, and slightly more functionality but other than that it does the same stuff) uses LESS ram and LESS cpu on the same hardware with the same asterisk version when compared to extensions.conf dialplan processing. That is just one example of something that could easily be placed in asterisk-addons, was disclaimed, and wasnt wanted. It has no patent and the license for the code it uses is mozilla spidermonkey which depends on the nspr stuff, both of which are tri-licensed - gpl,lgpl and mpl (more like BSD). There are more examples, but this is one that doesnt break. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tf.voipmich.com - Broken?
On Tue, 2006-07-18 at 13:10 -0500, Brent Torrenga wrote: Anyone notice that tf.voipmich.com (ENUM for US toll free service) will connect you successfully, but then disconnect after what seems like 30 seconds or so? Anyone know what might be going on here? I googled the hell out of voipmich and did not get very far. For North American tollfree there is always: sip:[EMAIL PROTECTED] iax2:[EMAIL PROTECTED]@voip.trxtel.com -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!
On Sun, 2006-07-16 at 23:57 -0700, Martin Joseph wrote: I think if you keep the older source in a separate directory, you can always cd back to it and do a make clean, make, make install. or if you are lazy, make takes multiple targets so you could do: make clean all install all on one like that way and if one target fails the others shouldnt proceed :) 'install' should have a dependancy on 'all' so if you just do make clean install it should work the same. It will use the newer zaptel if you dont do that as well, so if zaptel is the issue that causes you to want to go back then you will have to do a make clean all install there as well. This is also the reason I have avoided building from SVN, as I like the idea of being able to revert to an earlier working build if need be... you can have different SVN repositories on your local system as well, and still do that, or use a release tag to get a specific version, either way. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!
On Mon, 2006-07-17 at 19:21 +1200, Matt Riddell (NZ) wrote: It will sometimes tell you that there are modules inside /var/lib/asterisk/modules which were not compiled for the version you are compiling. If these are not asterisk-addons modules you will likely need to remove them. or modules from others that arent allowed to contribute to asterisk-addons or the tree itself for whatever reason, of which I have a few of those that have been specifically rejected for inclusion even though disclaimers are on file :/ politics at its finest. At least they work and it appears that some of them take less ram and cpu than default asterisk stuffs :) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: where is the error?]
On Mon, 2006-07-17 at 15:17 +0200, olivier.taylor wrote:
email message attachment (where is the error?)
SELECT\ left(Customer.balance,instr(Customer.balance,'.')-1)\ FROM\
Customer\ Inner\ Join\ subscriber\ ON\ subscriber.customer_id\ =\
Customer.id\ WHERE\ subscriber.username\ =\ ${CALLERIDNAME}
asterisk translates , to | then processes it. try \, instead see if
that cures your errors.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058 US WA +1 360 207 0479
US NY +1 516 687 5200 FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!
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Re: [asterisk-users] Asterisk stops abruptly
On Tue, 2006-07-11 at 08:20 -0700, Dan Brummer wrote: Hello, I'm recently having the problem where Asterisk just stops working. The console gets disconnected and the process appears to die. I am using Asterisk version 1.2.9.1. Anyone have any ideas on where I should be looking for the cause of my problem? Also, I notice there is a /var/log/asterisk/messages log file but it doesn't contain any information that I can use to help troubleshoot the application crashing. Is there a way to put more debugging in the log file? Yes take a look at logger.conf. There is a default of 'full' which will create /var/log/asterisk/full for example, and will have more info, but you can add the individual elements to the messages one if you would rather. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provider UNREACHABLE
On Tue, 2006-07-11 at 09:08 -0700, Ira wrote: At 04:59 AM 7/11/2006, you wrote: I am repeatedly getting a UNREACHABLE and then REACHABLE about 10 sec apart most of the time and then sometimes for about 45 - 74 minutes I have tried a reload and sip reload but neither bring the provider back ? I see this with sipdiscount and it's brethren occasionally. I'm guessing it has something to do with latency in the outside world, like why is the internet blazing fast one minute and painfully slow 5 minutes later. Here(Los Angeles) it tends to happen late at night, mostly when I'm asleep. with sipdiscount its mostly due to the fact that their sip server doesnt respond in the predefined time set by 'qualify' in the peer definition. If you do like qualify=2000 it will only display them if they cant respond within 2 seconds. Which at night in LA is daytime where they are located (Europe) and they probably see a higher call volume and thus their servers arent as speedy to respond. Note this number isnt the network lag by itself, its also the time it takes for them to respond to sip messages, as well as the network lag, so it can be midleading to performance by itself. These types of errors are generally harmless, if they bother you set the qualify to a higher number or turn it off with qualify=no. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provider UNREACHABLE
On Tue, 2006-07-11 at 14:10 -0400, Rick Smith wrote: teliax had a 2.5 hour outage today. I wouldn't call that short. its all relative, nufone had a 30 day outage :P -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate or rank ITSP
There is also the issue of origin of the caller. Not just geographically, but which network provider they use, and to some degree when. Most ISPs see higher traffic volumes when school gets out for the day (abnout 3pm) continuing for a few hours, then gradually declining until its later (11pm or so). As a result, a call made at 10am may have totally different characteristics from one made at 6pm (when both school kids and working adults tend to get home). As such the problem may not be the VoIP provider but with the network of the person sending the call, or a tertiary provider in between the two. Therefore I suggest that if such a list is done, people will include who their provider is, and whether they have had problems in the mornings, evenings, dead of night, etc. That way the list can be as fair as possible to all providers. And if you can search for providers from your geographic location off your provider, you can filter out the ones that are known to be bad from your location and network. On Tue, 2006-07-11 at 21:03 -0400, mike wrote: i'll be very interested in that it would also be useful that every qos rate comply with some [snip] On Tue, 2006-07-11 at 09:40 -0400, Barry Fawthrop wrote: Hi There I know of wiki there is a list of VOIP providers, but is there a list or can we create / suggest one that will list VoIP providers, their location and quality of service ? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NuFone, please send the log file
On Wed, 2006-07-12 at 06:40 +0800, Ronald Wiplinger wrote: Dear NuFone, Without misunderstanding I ask you again, please send the log file and pay back my money! Not following this request results in the assumption that NuFone is cheating and I will post this info every hour on more Internet places. This should help that other people will not trap into a cheating company. no offense but this is hardly the correct forum to be making such requests. If they choose to ignore such a request made in this venue I dont think that anyone will hold it against them (but then some might who knows). You may also try irc.freenode.net and message jerjer if he is on and not idle, you may get a response there. I believe they have a ticket system (although I have heard aweful things about it from others) on their website, and I am fairly sure they have a phone number, possibly even a tollfree so it wont cost you anything to call (and if it does you should look at http://www.trxtel.com the VoIP provider that pays you who will pay you for each minute you call a north american tollfree - yes its a shameless plug :) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NuFone, please send the log file
On Tue, 2006-07-11 at 20:16 -0400, C F wrote: I find this hard to believe, half a truth is a whole lie. First you just say the screwed you out $3k, not saying how, letting everyone assume thru phone service, then you change the story, you lied before, how do we know you are saying the truth now? Elaborating on a story isnt changing it, its providing additional details. That is hardly a lie. The fact that he claims he lost $3000 from nufone did not in anyway change, the fact that you assumed something then found out that you were wrong in your assumption does not make him a liar either. But you get an A for effort. In any case it doesnt make sense anyhow, why would you pay $3k for just setting up a server, when others here on the list will do it for much less. Also they never did the work? just took the money? thats an accusation that doesn't make sense. Just becuase others will do it for less doesnt mean that he didnt opt for them to do the work. I dont see that as not making sense, there are many lawyers out there that charge a premium for their services when you can get one much cheaper. Yet the expensive ones still get clients. The same is true for virtually every industry and even products. People will buy name brand foods instead of the white label generics even though they might be the same. People buy brand name drugs instead of the generics (which are identical in chemical composition) yet the generics are cheaper. It happens every day in virtually every good and service. As for someone taking the money and not doing the work, that happens all too often as well. As such a claim that it doesnt make sense that someone wanted something for nothing isnt so easily dismissed. NuFone placed a ton of international calls on behalf of a customer, then later they tried to not pay for the service that was provided according to their contract. Why? Because the calls were to a destination that had a high billing rate and NuFone didnt charge enough for the call. They signed the contract to pay for those services, they billed the customer for those calls, yet they didnt want to make good on it (jermey was all too eager to talk about this to forbes and other major publications too, so its Jermeys claim that they didnt want to pay for service provided). As such the something for nothing issue is something NuFone has an admitted history of. That tends to lend credibility to the claim that they did indeed do this, although it doesnt prove it, and the 'doesnt make sense' argument is no better than 'chewbacca'. Again you get an A for effort. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NuFone, please send the log file
On Tue, 2006-07-11 at 20:51 -0400, C F wrote: While I don't disagree with you, look at what my point was, just accusing them for such without any documentation doesn't make sens. He only brought that up after people started questioning it. So I dunno. And lets face it, this is the internet there is really no proof of anything. Screen captures of a webpage? That is easy enough to forge. Invoices? They too are easy enough to forge. Even if someone states they had horrible call quality you have no proof, but that is generally accepted that that one person experienced that. And where does that leave you? You have to either take a chance on your own or go with those that you trust and/or whatever is said the most. So since its hard to get any sort of proof you kinda just have to accept that it happened or not and move on. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NuFone, please send the log file
On Tue, 2006-07-11 at 21:44 -0400, Andrew D Kirch wrote: Michael Workman wrote: Very Simple. I hired JerJer to Have a SER and Asterisk setup with Acounting... JerJer told me to Talk to Shido6 and he would do it... He told me it Would cost me $3000 and he do it. He demanded the $ first and never did the work. I have said on several occasions that I think that Jeremy's a bit of a jerk, and tends to berate the less than clueful. I think I have demonstrated the same attributes tonight (which I make no apology for), HOWEVER, Greg (shido6) is honest. I'd like some further background information on this, as it is perhaps an issue that can be straightened out. (note well the following: I'm a third party with no stake in this and that could really care less but I'll lend a hand if I can.) IIRC about a month or so ago shido6 started asking around on how to set up SER. When did this occur? Perhaps its a recent thing and its not that they never delviered just that they havent yet delivered. There was a post elsewhere, where it seems that someone complained that techsupport is not responding to an email sent, and headers on that email indicated it was sent 20 minutes before the posts to the list. So it may be a similar thing, who knows. Then again if it was something much older it may be a different story. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] NuFone, please send the log file
On Wed, 2006-07-12 at 00:41 -0400, Alexander Lopez wrote: taken off line. Please respect the wishes of those that fund the list. ___ --Bandwidth and Colocation provided by Easynews.com -- easynews? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] NuFone, please send the log file
On Wed, 2006-07-12 at 01:00 -0400, Alexander Lopez wrote: Lists.digium.com yeah easynews provides that. Thanks for being clear. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Encrypting the Conversation
On Mon, 2006-07-10 at 07:34 -0500, Mike Bates wrote: Are you talking about ZiPhone a USB device ? Mike zphone is phil zimmermans (creator of pgp) encrypted rtp system. Unlike SRTP this does not rely on the server itself to provide the encryption. It also lets you be reasonably assured that if the numbers displayed match then not only is no one listening now, but they havent since you paired both endpoints. There is a drawback that SRTP can solve however, zphone only works on voip networks where the media proxy does not alter the data stream, it cannot be used to bridge to different channel types and codecs. This means that if you want to call out on the PSTN, SRTP can encrypt over the internet where zphone cannot. So it has its benefits, but also its drawbacks. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the story with X10*P FXO cards?
On Sun, 2006-07-09 at 10:22 +0200, Vincent Delporte wrote: Still, considering the number of people having similar problems with those cards, I was wondering what the problem is. Is it because the hardware, no matter what is advertised, is actually not identical from card to card so the zaptel driver doesn't work reliably unless they are among the few remaining authentic cards made by Digium before it stopped manufacturing them? digium didnt really make em. as to the reliability, some of that depends greatly on what chipset is on the card itself. There are a couple different ones that while cost about $5 couldnt be resold for $100 for good reason. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skype gateway
On Wed, 2006-07-05 at 12:18 +0200, Patrick wrote: I read the page about the Skype API at https://developer.skype.com/Docs/ApiDoc/Using_the_Skype_API_on_Linux Not being a programmer, I wonder if it's possible to use the API and the examples at the end of the page to come up with some way to interface Asterisk and Skype. Possible at all? Anyone tried that yet? Its not an API as most people think of the term, it technically is an API however ... What the skype API lets you do is remote control the skype GUI. This means that you *must* have a sound card and a GUI for the skype client to run. You can use the X virtual frame buffer to get around no GUI. It pretends to be an X server, but requires no video hardware, its all in memory. You can also remap the sound calls (skype uses OSS so remap open/close, read/write and ioctl to /dev/dsp which isnt that hard to do, I have code that does this already). This gives you a device which you can send/receive audio in a known format (the ioctl sets the sample rate, bits per second, number of channels, etc - and the program would know what those settings are). By doing all that you could interface skype fairly easily using dbus (how the skype gui is accessed in X). This however does add a real cost to doing skype, further you are limited to whatever the skype client lets you do, such as 1 channel or whatever. Now if you want to have a skype library that speaks the skype protocol that could happen, but it wont be easy. Skype uses a very weak, but still present encryption scheme on the binary (XOR with a fixed pattern on some but not all sections). It is decrypted in memory, and they do checks to see if a debugger is being used, if it is it refuses to run. People have documented how to disable this (which becuase it also verifies the checksum of the binary, you have to do it after loading the file). Gotta wonder why they went to so much effort to obfuscate the binary. Once you get access to the internals of the program you could potentially figure out how it communicates, look into the crypto routines and see how its dealing with keys and the like. And once you have mapped all that out and mapped out the protocol, and think you have a working library, you get to start all over becuase ebay/paypal/skype decided to change it on you and possibly sue you since that is a violation of their TOS to do (that is also potentially a felony in the US per 18 USC 1030 - the hacking statute, it wont be an automatic one becuase of the elements required but it could be fairly easily). In short, you gotta ask yourself, if they dont want to play nice with others is it really worth it for someone to try to play with them and constantly play catch up? Skype also uses wideband codecs, something asterisk has problems with, so you would lose the better sounding audio that skype offers if you run it through an asterisk box. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skype gateway
On Wed, 2006-07-05 at 16:01 -0400, Tigran Kocharyan wrote: Patrick, I think this is your answer: https://www.nch.com.au/skypetosip/index.html That oinly runs on windows, which isnt acceptable for some people. Further I have yet to hear from anyone that has used it with more than 1 channel, has anyone? can they share their experiences? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] please remove the autoresponder
it appears that scott has an autoresponder on the list saying he wil be gone for the next 2 weeks, it also appears that he is responding to himself, which is going to cause an exponential growth in the list volume. Infact it also appears that he will respond to this message as well, which will increase the list volume.. Can he be whacked until he returns? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Is there a search feature?
On Tue, 2006-07-04 at 23:10 -0400, David Beckerdite wrote: Is there an archive for this list that can be searched? If so, could someone tell me where it's located? google works... site:lists.digium.com asterisk-users your search query here -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More g729 calls than licenses?
On Tue, 2006-07-04 at 23:49 -0500, voiplist wrote: Any way to monitor this? Send an email to admins? Something? On 7/4/06, Thomas Kenyon [EMAIL PROTECTED] wrote: voiplist wrote: What happens when/if your Asterisk server is asked to handled more g729 calls than it has licenses? Does it fall back to an alternate codec or does the call get rejected? Well IME you get around 15 notices a second in the console stating that you have run out of licenses and there is no sound in either direction for the caller/callee. I believe with the digium codec you can do asterisk -rx 'show g729' and get data back that tells you how many are in use and how many are available. You could write a script that will look and send a report, however there are problems with this: 1. it cuases higher system load to run this often 2. calls can be bursty, a bunch of short lived calls can consume all your licenses, and your script didnt run in time and you didnt know until its too late. The real fix would be to hook into the SDP negotiation for sip (and whatever codec negotiations are used for any other protocols you use) to check if g729 loaded, if so check avialable licenses, if enough are available then allow that coded, otherwise dont list it as available. If that is done this becomes less of an issue since you wont have dead calls and warnings spewed to the screen. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: vserver with no /dev/tty* how to run asterisk -c for a colored CLI?
On Wed, 2006-07-05 at 08:32 +0300, Tzafrir Cohen wrote: ln -s /dev/pts/20 /dev/tty9 I got a terminal for asterisk *g* and now I have a colored CLI running :)) But how are you going to guarantee /dev/pts/26 will exist? Specifically: what happens when you end your current ssh session? maybe by making it part of a script with the 'tty' command? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone Tollfree Port
On Sat, 2006-07-01 at 11:33 -0400, John Kington wrote: I tried to get an update from NuFone but Has anyone gotten their tollfree number ported to another provider by NuFone? Should I just forget it and move on? Regards, John I have heard a lot of people have gotten em ported to asterlink within a day. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting a group of phones available channels
On Fri, 2006-06-30 at 13:31 +0400, Jean-Michel Hiver wrote: Hi List I have 10 separate SIP phones, and I wish to limit the simultaneous available channels to 5 maximum for these. How would you go about it without setting up a separate * box? Cheers, Jean-Michel. you can limit it to the provider end by doing a limit, read the page below for 1.2 notes, as the naming changed. You can do a setgroup/checkgroup in the dialplan putting all 10 people into the same group. Lastly, and probably the least effective, is you can watch channel usage and when someone exceeds 5 run over to their desk and smack them with a rotten fish. http://www.voip-info.org/wiki-Asterisk+sip+incominglimit -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Limiting a group of phones available channels
On Fri, 2006-06-30 at 10:44 +0100, Steve Langstaff wrote: trixter aka Bret McDanel wrote: Lastly, and probably the least effective, is you can watch channel usage and when someone exceeds 5 run over to their desk and smack them with a rotten fish. http://www.voip-info.org/wiki-Asterisk+sip+incominglimit I can't find the 'rotten fish' stuff documented anywhere on voip-info.org - was that some sort of red herring? Its an advanced asterisk management option. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
How many channels have you guys been able to get with this? The only problem I have with this is that it takes skype and a soundcard (virtual or otherwise) and the API is really executing commands on a running skype process. In my opinion its not worth it for 1 concurrent call per account. I have written code that works with skype in linux that simulates a virtual sound device. I have used that and successfully done calls out with this. I havent played with the dbus stuff (how you control the skype app from within linux) but since I have a soundcard that I know the audio format of it wouldnt be difficult to integrate this into asterisk, I could tweak chan_oss and make it into chan_skype fairly easily since that takes care of the other half of the equation. The only thing missing would be the events via dbus, which there are plenty of examples on so its not like all new code would have to be written. But its just not worth it if you have to have skype running for each call. And then you would potentially have to have a new username for each running process, and skype really wants X on linux so you would have to at least have the X virtual frame buffer (it works and acts like X but never displays anything or uses any hardware). That seems like an aweful lot of wasted resources on a box to connect to skype. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Wed, 2006-06-28 at 08:14 +, [EMAIL PROTECTED] wrote: Well, look at it this way: if you get the working, you can buy one of those tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard and a ethernet port. Run Linux off a CF card and have it setup to *only* interface with Skype and Asterisk. Basically, make a Skype ATA, but it would convert Skype to SIP. I think that could still be considered an ATA, right? Or a gateway at least. it wouldnt need a real soundcard, that is part of the point. I remap all the calls the same way that I did for allowing instant porting of your digium g729 licenses (in another post, code is at my personal site http://www.0xdecafbad.com/ somewhere). Remapping those calls is trivial, there are very few things that are acutally done to a soundcard to set it up, ioctl() for setting the sample rate, etc and read/write/open/close basically. Really trivial code. It would however be nicer if you didnt have to run a seperate copy of the binary for each call. This has a direct cost against memory. It would be better if it didnt use memory to open a GUI (even with the virtual framebuffer for X it still takes all that memory even though it doesnt display for real). I also doubt that a 386 would cut it, with everything going on it would have to be faster and that pushes the cost up. If you are going to do that it might be cheaper to buy one of the 1,2,4 port FSX/FXO devices for integrating with a phone system or something (some plug into wall jacks others into phones). The 4 ports are about $750 which is steep. The 99 port one which is unclear how you use it exactly is $1500 or so. Actualy looking at the 99 port model it appears that its just a usb soundcard that has a FXS port on it, which is a silly way in my opinion, and still requires a system running skype to work :( Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you I dont, the overhead is insane. As as for a price for 'them' it would just be a software program. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Wed, 2006-06-28 at 13:15 +0300, Tzafrir Cohen wrote: Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you You would be violating the terms of usage of their API if you want to use (let alone sell) such a device. I am unsure if all the hardware devices are basically usb soundcards or not, havent really looked, but if they arent then it would seem to me that its possible to do. Further I dont think it would be against their api to write sofeware that uses their api. That is what was being discussed when this comment came out, so ... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki Voip Phone reviews
On Wed, 2006-06-28 at 19:10 -0700, Mike Fedyk wrote: Hi, We have a page on the wiki just for phone reviews, but I think it needs a bit of format change. Instead of individual reviews for each phone, I think each person should review all phones they have worked with and list the phones they have had access to and rank them in relation to each other. Also each review should have a date so the reader can see how fresh the data is to current. instead of replacing, why not create two pages, where generally one person lists their phones and rates the phones they used then someone who feels like it can sumarize it into a per phone thing. I think it can be better to see everyones opinion on a specific phone but harder for people to input that info since you aparently dont have a relational database to play with to cross reference everything. The other alternative is to find osmeone who can host such a database enabled site and link to that where people can search by rating, vendor, etc. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with callerid in sip to isdn gateway
On Tue, 2006-06-27 at 15:00 +0200, Morten Isaksen wrote: Hi! I have this setup: PABX --ISDN30-- Asterisk 1 --SIP-- Asterisk 2 --ISDN30-- TELCO Digium TE410P is used in both Asterisk 1 and 2. When I set the CLIR bit on the PABX the Callerid / ANI is removed somewhere between the SIP interface on Asterisk 1 and the SIP interface on Asterisk 2. Have you used a packet sniffer to ensure that its actually sent to asterisk 2? If it isnt then that may be the entire problem. Before trying to diagnose anything on the isdn side I would make sure that it is infact being sent correctly. Alternatively you can try some noops() on asterisk2 for when a call is received to display the caller id to the console, that may be easier for some than reading sip headers. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP
On Mon, 2006-06-26 at 13:16 -0400, Brian Capouch wrote: It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html and they have been working with cisco on ice (which is standards based, although ice is more of an extension to sip than anything else). But shhh that doesnt help the people that want to bash for no other reason than they can! -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Multi Call Generation
On Thu, 2006-06-22 at 13:16 +0200, olivier.taylor wrote: sipsak is ok for that Olivier sipp.sf.net is also not a bad product. They both work slightly differently so it depends on what exactly you need. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Add Country to CDR's
On Tue, 2006-06-20 at 10:49 -0400, William Piper wrote: List, Does anyone know how to add the dst Country to the CDR's via Macro (preferably). cdr(userinfo)? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Add Country to CDR's
On Tue, 2006-06-20 at 11:14 -0400, William Piper wrote: Thanks Bret, but how about an example or webpage? I'm not finding anything on google about this command for asterisk. What about AppendCDRUserField()... would this work? that seems to be the same thing. the userfield lets you stick arbitrary data into your cdr records. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOS Scores and LCR
On Sat, 2006-06-17 at 10:16 +0200, Florian Overkamp wrote: There are ways to guesstimate MOS scores on a call by continuously getting some decent statistics from the jitterbuffer. We've had an intern do some work on this using IAXclient. http://www.speakup.nl/en/opensource/jitterbuffer/ yes and I suggested that however, MOS is an opinion, so its totally subjective and not based on anything 'real'. That was kinda my point earlier. Personally I think that its better to isolate the network/cpu issues and correct them to get what a given implementation of a codec is supposed to be rated at (ideally the two would be intertwined). By looking at the jitterbuffer (assuming you have one, if not you may have to get some code that will inspect packets on the network which can be done but isnt as easy, its still not that difficult), latency, dropped packets, etc and generating stats based off that you can probably make a better guestimation of call quality without actually listening and then scoring the calls. The work that you have done so far is a great step towards a product that many people might find useful. In a nutshell the concept I am thinking about is a tool that you drop onto your network and it will monitor the data (presumably not just iax but sip, h.323, whatever) and generate live stats of the call and possibly even have an alarm system that would send off a page or something if conditions get too far from 'normal'. The approach to wait for customer complaints before reacting to voice degradation is likely not going to work for many. A tool like I described, which it seems you are well on your way towards even if that wasnt your intention, could add a more proactive approach to it and increase reliability and quality for many. I do not mean to trivialize network management and do understand that sometimes problems are outside your control (ie packets on the uncontrolled internet may experience problems because of a tertiary provider between you and the remote end) and as such this is just one peice of the puzzle as I see it. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOS Scores and LCR
On Sat, 2006-06-17 at 12:52 +0200, Florian Overkamp wrote: The work that you have done so far is a great step towards a product that many people might find useful. In a nutshell the concept I am thinking about is a tool that you drop onto your network and it will monitor the data (presumably not just iax but sip, h.323, whatever) and generate live stats of the call and possibly even have an alarm system that would send off a page or something if conditions get too far from 'normal'. Yes, that would be excellent indeed. Problem is that the location of measurement will influence the scoring :) If you have good ideas towards this we'd be very interested in participating. off the cuff I am thinking libpcap and possibly some of the open source libs that allow you to read RTP frames and other goodies. Configure the switch to replicate the traffic to the monitoring box so that you dont have any delay associated with the monitoring box itself. Granted packet sniffers arent 100% and can drop frames themselves, even though they are sent and received, but its a start. I really havent given this much thought, but think that something along those lines wouldnt be terribly hard to construct, I just dont know how accurate it would be nor any way to really make it more accurate. This way it would be generic for more than one particular softswitch. Integrating a process like this into the softswitch itself might be a better approach, just not as portable. The measurements would most likely be more accurate (given that frames dropped by the sniffer but were actually received wouldnt show up as a 'loss'). You also want to have a time value associated with the loss. Dropped packets 5 hours ago means nothing for calls going through right now, but it can be important for overall network monitoring so you dont just want to discard that data. I *think* opal is doing something like this internally, however I am unsure. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOS Scores and LCR
On Sat, 2006-06-17 at 23:25 +0800, Steve Underwood wrote: Calling MOS totally subjective is rather strange. Telephony only has to meet subjective goals. In reality, MOS is pretty objective, as it is a carefully controlled experiment across enough subjective individuals to filter out a reasonably objective answer. it is subjective to the particular test, which afaik MOS doesnt have standards on what types of people should form such a test group. As such you can have different results based on the combination of the sample sounds and the group that is listening to it. This is why many people are typically involved, but that is not a guarantee that the group will be diverse enough and objective enough when rating stuff, it is just a mitigating factor against all out bias. They exist, but current ones cost a fortune. Used with understanding they do a fine job. Sadly people who don't understand them tend to read far to much into the answers they give. Sometimes they are seriously out of line with perceived quality, but they usually do well. Yeah I wasnt clear I meant something that is either open sourced or priced reasonable. I do not mean to trivialize network management and do understand that sometimes problems are outside your control (ie packets on the uncontrolled internet may experience problems because of a tertiary provider between you and the remote end) and as such this is just one peice of the puzzle as I see it. For those reasons may think VoIP will always suck. They might well be right. Anyone that would say VoIP sucks becuase of that I would have to question a bit, simply becuase VoIP doesnt mean voice over the open internet, although to listen to marketing types that is often what they want to claim. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOS Scores and LCR
On Sat, 2006-06-17 at 01:26 -0400, Daniel Salama wrote: Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random calls so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? MOS (Mean Opinion Score) is generally a bunch of people sitting there listening to audio and rating it 1-5 (there is a newer method that is twice as good becuase it goes 1-10, basically all values are double). Its their opinion. This generally cant be dont automagically and still be MOS. You can try to track frame drops and other things on your end to rate call quality and try to come up with something, but that technically isnt MOS. AFAIK asterisk doesnt keep statistics of jitter, frame drops or anything else, that might be a good project for someone to take on, especially if you have multiple providers so you can rate quality in a more meaningful way. The human ear really isnt the best tool for much of this. http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci786677,00.html http://www.tmcnet.com/tmcnet/articles/2005/voice-quality-measurement-voip-alan-clark-telchemy.htm http://channels.lockergnome.com/it/archives/20050715_voipqos_mos_mean_opinion_score_explained.phtml -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)
On Wed, 2006-06-14 at 19:58 -0400, Daniel Salama wrote: Mainly GXP-2000 (with silence suppression off) and Eyebeam (with Enable microphone noise reduction off) its safe to ignore that too, it just means that asterisk doesnt support a sip feature that your phone does and its telling you hey I know that the feature exists but I really dont support it. If you disable CNG in the phone you wont see the messages anymore, alternatively if you ignore the messages they wont bother you anymore :) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Bounty Doubling program
TRX Teleocmmunications the VoIP provider that pays you would like to assist those that make asterisk better. To that end we are setting up a program where the community itself can help double the bounty for all of the outstanding code that is wanted but not yet present. TRX will match any bounty paid on any new code that gets put into tree in response to a bounty listed at http://www.voip-info.org/wiki-Asterisk +bounty. There are some rules for this doubling program which are available at http://www.trxtel.com/index.php?page=Asterisk_Bounty in short we will donate a small bit of money for each minute that each person is on the phone to a tollfree north american number (we will be having the same program for inbound DIDs soon as well). When a bounty is claimed, we will match what is paid. Asterisk gets more features, developers get more incentive, and the community as a whole can help make that happen. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com The VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf
On Wed, 2006-06-14 at 00:50 +0400, Jean-Michel Hiver wrote: Actually i've done 50,000+ line dialplans using my Asterisk::LCR dialplan generator, and asterisk has been just fine with it. I have you beat, I have done over 500k when loading my country list that I no longer maintain which is now at http://www.astbill.com. It worked although reloads and initial loads are noticably slow, and asterisk seemed to be memory hungry. I no longer do it that way and have had better luck with it. There are people that load a per user set of contexts out of a database, so that they only have what is being used not everything known to man, they have even put a expiry on the dialplan entries so they are freed from memory if they arent used in X minutes. I do not know if these patches are public or not, but I do know they exist, this was for someone who easily exceeded 2M lines and it was unmanagable otherwise, given they have many boxes which all need that info and the way asterisk processes dialplans, which becomes more evident under load. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question setting up a bat phone extension.
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote: Ideally I would have liked the pap2 to have done the same as 'immediate' when talking about fxo, capi, misdn, etc, but I couldn't get it to automatically dial nothing. A '0' was the best I could do. If anyone knows how to put it into immediate mode to come into asterisk as an 's' extension, I'd love to hear about it! sip targets arent limited to numerics, have you tried to dial an 's' instead of '0'? That is valid in sip, I just dont know if the pap2 supports it. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] revisit to legacy PBX and CID over PRI
On Thu, 2006-06-08 at 16:49 -0400, Steven wrote: My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. how long is the caller id string? I believe the spec used on the pstn is 15 characters, while asterisk supports much longer callerid strings that may be a problem. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Prices of g729 codec
On Fri, 2006-06-09 at 01:52 +0100, Chris Bagnall wrote: I think the issue for many people here is not the cost of the licence itself, but the very frustrating lockdown to specific pieces of hardware without any real reason. I say without any real reason because anyone who doesn't care about the licencing of g729 has an easy alternative in the form of the downloadable g729 binaries. They aren't exactly difficult to find - hell, they're linked to from voip-info and a google search for g729 binary asterisk always gives that page in the first few results. Is the ridiculous hardware locking down something that was imposed on Digium by the rights holder? its even more rediculous how trivial it is to bypass. Again I came ot the same conclusion about using that demo program for this purpose, anyone who has the licensed codecs does it because they want the licenses not just for the codec. I think of it more as a 'soft hardware lock' since it is so trivial to bypass ... nuisanceware :) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Fri, 2006-06-09 at 04:49 +0300, Tzafrir Cohen wrote: On Fri, 2006-06-09 at 01:52 +0100, Chris Bagnall wrote: I say without any real reason because anyone who doesn't care about the licencing of g729 has an easy alternative in the form of the downloadable g729 binaries. They aren't exactly difficult to find - hell, they're linked to from voip-info and a google search for g729 binary asterisk always gives that page in the first few results. Which are actually not legal for you to use with your PBX without paying Intel a bit, IIRC. according to intel.com its legal for you to use while in development, the same as the patent itself. Distributing intel code requires payment of a couple hundred, so there is a bunch going on both sides of that. However it is still done, which means that there are some that dont care. These people are less likely to try to break a licensing scheme, however trivial becuase they just want to use the codec and not pay anything. As long as both are available the people that get one with a license want the license itself not just the codec. its even more rediculous how trivial it is to bypass. Indeed. It would have to be much more evil (limiting) if it were to withstand such trivial attacks (and make you resort to more complicated hacks to work around it) Not perfect. But then again: in a perfect situations we would not have such an acute patemts problem. There is always the option of staying away from those codecs. that is likely what the majority do, however there are some for interopability reasons they want to use that codec over others that might be similar in bandwidth concerns (ie speex) but arent in most every voip hardware device. You end up having to pay somewhere, in a tdm-voip world if you use g.711 you have to buy extra bandwidth but less cpu, use g.729 you have to buy extra cpu and licenses but less bandwidth, some stuff in the middle is a compromise between the two. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Prices of g729 codec
On Wed, 2006-06-07 at 07:55 -0700, [EMAIL PROTECTED] wrote: For all the noise about this noone has mentioned one important thing. We should be gratefull that we have access to G.729a in Asterisk, whatever the mechanics of the licensing. It's obvious that its a pain in the [EMAIL PROTECTED] for Digium who absolutely not making ANY on it money for their efforts. It would be really easy for them to say no more and it wouldn't really impact their business at all, except to reduce their headaches. but they do in 2004 mark said it was one of their biggest revenue streams. Or do you mean that they dont make any money selling asterisk under their business edition line? Or maybe they dont make any money selling the hardware to people who buy it to 'support' asterisk development. I believe that cnet said they made over $10M/year in an article about an interview with mark. If $10M/year is not ANY money I would like to not make any money too. This will be especially true when they introduce their new hardware based transcoding engine. Why then should they gee just like sangoma (only sangoma anounced it first :) wonder if it will still use the zap interface and choke the system with more interrupts than required. I also wonder when asterisk will have better sangoma support so you can cut your interrupts from say 1000/sec to 50/sec. But that probably wont happen in tree. Again, we should be gratefull! It could very easily go away altogether. the codec? there will be alternatives for a licensed g729a and B (for those that want to do VAD when that is implemented) codec for asterisk. Those of you constantly complaining...this is supposed to be a open source community...don't just demand a better licensing scheme...design and implement one. That can be your contibution to the project. I'm not a code jockey or I'd have a go myself. its being done. Infact I am on the phone with some people talking about that right now. And have been for a little while (on/off for a couple months). Now you say that you arent a code monkey so you are unable to write one, but you can suggest what others should do. Specifically code something new. Hmm sounds like you just did the very thing you are complaining about. So I am lost are you complaining about your post now or what? All I have to say is that at least you can (aparently I still havent tested it with asterisk) port your digium licenses for which you paid when digium is closed but your business isnt :) and to show that I am not just suggesting a different licensing model but actually contributing here is the link to the BSD licensed code (whee its not gpl) for a trivial program and thus is my contribution. Note I have no disclaimer on file as the gpl is against my religion and as such am barred from contributing to asterisk directly. http://www.0xdecafbad.com/Remapping-function-calls.html -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Wed, 2006-06-07 at 11:17 -0400, Ben Klang wrote: On Monday 05 June 2006 15:41, Andrew Kohlsmith wrote: The (current) problem is that the registration program does not ask which ethernet card you wish to bind to, nor does it look at the Asterisk config and use the MAC address of the ethernet card whose IP address is referenced in bindaddr (as an example). It grabs eth0 and runs. Has anyone tried renaming the interfaces on the box? On all my systems I rename the ethernet interfaces to more friendly names (dmz, lan, ext) so there is no ambiguity. If the license verification code is really looking for eth0 it might be possible to juggle some interface names until the USB ethernet interface shows up as eth0. I havent used ifrename so I decided to install it and see, it appears to fully masq the name so that when you get the device by name it doesnt exist as the old (meaning its not an alias but a real rename). The device cant be up when this happens though, which may pose problems for some who dont want to down the interface. I dont know about the register tool, whether or not that takes more than one ethN device, but I have heard rumors that the codec itself will try eth0, eth1, etc until it gets an error back from the ioctl() saying that the device doesnt exist. If its pulling all those devices then it stands to reason that it will use them all when comparing the licenses/* files. But again I dont use the digium codec but instead the 3rd party one that as yet isnt released to the public, so I really cant say if this is true or not. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] show channel issue with 1.2.9
On Tue, 2006-06-06 at 11:37 +0600, [EMAIL PROTECTED] wrote: try asterisk -rx 'show channels' that is what I did try, yes I ommited the quotes in the email guess it wasnt understood that it returns only the header and not any information on what channels are in use nor any information on how many active or total calls are in progress. This works upto about 50 channels, after that it starts to break. This worked fine with over 200 channels with 1.2.4 however its very unreliable with 1.2.9. So something was changed. If I do just asterisk -r then type show channels it appears to always work, its just when its done from the shell prompt that it doesnt. It acts almost like a race condition that 'wins' when the channel count is low, but looses almost always when it gets to a moderate level. Why I was thinking it was a threading issue. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STNU spport
On Tue, 2006-06-06 at 15:17 +0800, Chen Fan wrote: Hi,all. There have any STUN spport for asterisk? thanks,,, where asterisk queries a stun server or where asterisk acts like a stun server? Becuase stun is totally self contained it would be silly (in my opinion anyway) to have a stun server built in. There are many free ones out there, stunner is one example. As for stun client support, afaik asterisk doesnt really do that yet. and it should have a periodic timeout if it does, but it would be a chan_sip addition. There is also an RTP patch that gets rid on 99% of NAT problems with SIP by technically violating the RFC but given the way networks work 99.99% of the time it would work perfectly (so if it were an option it would be good for everyone). What it does is on the RTP port if it receives a packet it will use that IP instead of whatever is specified. This fixes NAT problems on the other end at least, but for some reason it never made it into the source tree :/ -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STNU spport
On Tue, 2006-06-06 at 16:12 +0800, Chen Fan wrote: hi, We need STUN client support for asterisk... becasue the service provider only offer STUN interface,, so i can not connect asterisk to their server all stun does is resolve your external IP by sending data to a foreign server which looks at the IP and returns it back to you. It has nothing to do with the channel used other than SIP will then use that IP (which can be defined by either externhost or externip - dont forget localnet too in sip.conf). i have found that there someone is develop res_stun.c ..but still not release... likely that is just going to replace the externip value in the chan_sip driver. I cant imagine that it would do much more than that. Have you set both externip and localnet in sip.conf and checked to see if that works? If you dont do NAT on your end it wont even be required. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_flite.so
On Tue, 2006-06-06 at 15:41 -0400, Russell Handorf wrote: Hello all, I'm playing with app_flite, as I'm sure you've guessed. I have the sources compiled and running, headers and libraries in their respective places. I then compiled app_flite without any problems or errors. When I try to have asterisk execute the module, I get the following error Jun 6 15:29:06 WARNING[3197] loader.c: /usr/lib/asterisk/modules/app_flite.so: undefined symbol: new_voice Jun 6 15:29:06 WARNING[3197] loader.c: Loading module app_flite.so failed! asterisk doesnt do lazy mode when loading modules (specifically linux wont allow DLTD_LAZY the way it should but that doesnt matter). What this means is that it tries to resolve all the symbols when it loads the module. You are missing a library that asterisk was supposed to be linked against, something isnt in your LD_LIBRARY_PATH or /etc/ld.so.conf and ldconfig wasnt run, or whatever. My guess is that its the flite engine itself that has a library that isnt being found and loaded and thus is preventing the module from loading. try ldd /path/to/app_flite.so (I bet it doesnt show anything meaningful but it might) see if there is anything shown as 'not found' in the output. Make sure that you installed the flite stuff correctly, whatever app_flite requires to run, especially libraries. I'm using Asterisk version 1.2.1 you might want to upgrade that, there have been many things fixed (and possibly new things broken, hard to say right now). But if that works for you and you lock it down it should be ok. The one other tidbit is that when I ran make install for the flite sources, it errored with Installing mkdir -p /usr/local/bin mkdir -p /usr/local/lib mkdir -p /usr/local/include/flite /usr/bin/ginstall -c -m 644 include/*.h /usr/local/include/flite make: *** lib: No such file or directory. Stop. make: *** [install] Error 2 that is probably why it doesnt work, as mentioned earlier it really looks like a missing library, and it looks like there might have been a problem installing the libraries, you may want to investigate why that broke. Was it some prefix that got messed up, was it trying to install to /usr/local/lib which doesnt exist and it wont create the directory (linux distros normally, but not always dont do /usr/local however BSD ones normally do). -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zork and Asterisk
On Tue, 2006-06-06 at 13:09 -0700, John Todd wrote: http://www.boingboing.net/2006/06/05/play_zork_by_phone.html Let me preface this idea with one comment: I don't have the time to do this - I don't even have time to eat these days. But someone out there has the cycles to do this... and it would be very cool. zasterisk has existed for a while, it used the perl infocomm parser :) I recall looking into zasterisk last year sometime. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Zork and Asterisk
On Tue, 2006-06-06 at 13:47 -0700, John Todd wrote: While I love voice synthesis, I think that you'd gain legend status only if you had Allison (or, I hesitate to say, some other voice talent) do the dramatic readings. People respond so much better to real recordings - I tend to use synthesis only where it is impossible to have pre-recorded phrases, and since Zork is pretty much entirely canned personally given the genre of the game a brittish female with only a slight accent might do better than allison. Of course if you are clever (set(language) or whatever) and have the user select the voice that reads might be interesting. That way you can get a sean connery sound-a-like or a patrick stewart sound-alike or whatever for some callers and females for others ... Then if you get into translations so its in native languages to different callers ... Could really turn into something. But of course you would want to add more than just zork (and infocomm games for that matter). A MMRPG game played over the phone where people can go into a 'tavern' (voice bridge) and other stuff might do well too. That way you arent all alone in the virtual world. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Mon, 2006-06-05 at 09:49 -0400, Andrew Kohlsmith wrote: On Saturday 03 June 2006 16:57, trixter aka Bret McDanel wrote: but $10 only gets you one license, what if you are vonage sized and need to support a million customers? What if you accept that you can settle If you are Vonage and need to support a million customers I will bet you are not transcoding a million conversations on regular PC hardware. You will have AS5300/5400 boxes or MaxTNTs, which you have already paid WELL MORE than $2M for, which INCLUDES the AudioCodes patent license for g.729. You can't avoid it and stay legit. Sorry to burst your bubble. my bubble wasnt bursted as you proved my point. Thank you for proving that for me so I didnt have to. I was responding to the comments that said it was only $10 for a license, and not any comments on larger entities. for a 5:1 ratio, then its only 200,000 or $2M. Just for codec licenses, not to mention all the other costs of being a business. What if you are smaller than vonage, say 10k channels in use, then that smaller entity, probably without the hundreds of millions of VC that vonage got you would have to come up with $100k. Still more than $10. Again, 10k channels you'll have a half dozen MaxTNT boxes terminating DS3s. Your fixed costs will already be significantly higher and that little $10 license fee is included in that. Its not $10, which also goes along with something else I mentioned elsewhere. Digium charges $10 but the max cost for a g729 license is about $1.25. It goes down to about $0.10/license in quantity. As such it doesnt add a whole lot to the cost of the device once the initial code is in place (as that development does have cost since the license fee doesnt cover any implementation, only the right to sell that implementation). And again to clarify, since this was aparently lost somewhere, I was responding to the mentality that everyone is a home user and its only $10 for a license and that is all anyone ever needs to pay. You have proven me right, thanks again for that. When you get out of the home user mindset the cost goes up dramatically and the argument that I responded to that the cost isnt that high at $10/license was invalid, even though you seem to be saying that it is that same cost, which anyone who really knows anything about the licensing knows that isnt true. If you are going to bring businesses into it, at least accept that a business would most likely pay more than $10 for their licensing needs. Nonsense. The license fee that Digium charges is for onesie-twosie stuff. If you're making a real go of this as a business you will be paying that patent license fee either through Digium (if you're transcoding on PCs, in which case you are either doing something different or you're just plain stupid) or you are paying a smaller patent license fee which was included in the price of the hardware on your NAS equipment. My point again, thanks for the recap. The $10/license fee outside of the home user market is what I was contesting. Why do you keep proving my point in such an argumentative way? no its not that they want quantity becuase they will sell just one license, they only want to deal with people that implement the systems not the end users of the system. They claim the reasoning for this is to make it easier for end users to know that they have licenses - basically if you have it you are licensed. Even if that isnt the case. Check www.sipro.com for more info on g729 licensing. It's always easier to work with businesses and deal with quantity than it is to deal with the public or end customer and all the hassles with that. I'm positive that AudioCodes doesn't want to staff a customer service department to deal with Joe Sixpack who's cousin's friend's son is a computer whiz and hooked up this phone over teh intarweeb thingie for him but he just can't it working perfect. I never said they did, I replied to the notion that $10/license isnt that much and people should just pay digium because we owe them for beta testing the software for them that they sell commercially. I used a specific example designed specifically to show that the $10/lciense fee could actually be a considerable sum instead of only $10 which is what I replied to. I am now begining to think that you didnt follow the thread or even read what I replied to. Out of curiosity do you read slashdot? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE
Re: [Asterisk-Users] Prices of g729 codec
On Mon, 2006-06-05 at 10:00 -0400, Andrew Kohlsmith wrote: On Saturday 03 June 2006 04:05, Sahil Gupta wrote: We recently had around 60-80 licenses become useless because Digium refused to renew the keys on that. That was a bit of money kissed goodbye. Ok, that's a great fairy tale. Now tell us the true story. Well it may be it maybe not, I wouldnt call people liars without proof however. What I will do is state that I have written a tool that allows you to port digium licenses to different boxes. Lets say that digium is closed (2am, weekend, whatever) and your box caught on fire or whatever. You cant get them ported right then because digium is closed. My tool will let you convert that license file to something else. I will be making this program available publicly this week pending a final test to ensure that it doesnt itself cause your system to catch on fire :) When you buy licenses from Digium, you register them and they are branded with information from your machine (most likely MAC address of the NIC, but I'm not 100% sure nor do I particularly care for the details.) And I am relying on that fact to thwart piracy with my tool. If the license file is shared and later discovered to be shared it will be easy to track who leaked it and thus cause fewer if any people to leak it. That and the fact that if people pay for the digium codec they are doing it for the license not the codec itself since there are unlicensed ones out there if they dont want the license. The only reason to get the digium codec is infact the license that accompanies it, so piracy on any level should be rare if at all. If you upgrade hardware you may have to re-register them. Digium allows you to automatically re-register once without phone calls or any explanation. After that, you cannot re-register without calling Digium and making a case for it. This is a restriction placed on them by the patent holders of the g.729 codec. not really, that is conjecture. Having entered into a g729 license myself I can attest that changing the license like that isnt a requirement of my contract. So the TRUE story is that you had 60-80 licenses, registered, changed your hardware, re-registered, changed your hardware AGAIN and for one reason or another failed to convince Digium that it was a legitimate change to warrant a re-registration. That may be, which goes with what he said that you said was a faery tale. So which is it, faery tale or fact? You seem to be inconsistant. And if he could have made a case that he needed it changed, wouldnt that negate your argument that the patent holders wont let digium change it more than once? Is iut the patent holders (or their authorized agent sipro.com as the case may be) or digium that has the discretion? I have personally called Digium and provided sufficient reasoning to grant me a third registration. So honestly now, Sahil, what did you guys do that was so different? It *really* pisses me off when people like you give a half-assed, half-baked digium sucks post. If you've got an honest beef with Digium, then sure, lay it all out, but don't present half the fucking story and then bitch about how the big bad Digium beat you up and stole your lunch money. ... just like my kids... Wh, Joshua hit me! Yeah but you've been bugging the shit out of him for the last 5 minutes and he asked you nicely to stop twice. I'd have hit you too, Katie. I cant speak for you hitting your kids when they bother you for 5 minutes, however if it bothers you that people post 'I had a problem with digium' email and your explanation contradicts itself, even your claim that someone is being less than honest when you yourself state later that they were being honest seems dubious at best. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Mon, 2006-06-05 at 10:46 -0400, Paul wrote: I really doubt that Digium would insist on the $10 fee for a quantity buyer. no they do give some discount for quantity, people have mentioned that when they bought a bunch. However I think they said it was close to $8/license for 672 channels. Not a whole lot of a discount and certainly more than other solutions. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Mon, 2006-06-05 at 12:01 -0400, Andrew Kohlsmith wrote: On Monday 05 June 2006 11:03, trixter aka Bret McDanel wrote: Again, 10k channels you'll have a half dozen MaxTNT boxes terminating DS3s. Your fixed costs will already be significantly higher and that little $10 license fee is included in that. Its not $10, which also goes along with something else I mentioned elsewhere. Digium charges $10 but the max cost for a g729 license is about $1.25. It goes down to about $0.10/license in quantity. As such it doesnt add a whole lot to the cost of the device once the initial code is in place (as that development does have cost since the license fee doesnt cover any implementation, only the right to sell that implementation). Stop the presses: quantity purchases get price breaks! High enough quantities let you deal with the manufacturer directly! This is news how? again you are either intentionally ignoring what is said or unable to read, I dont know or care. You cant despite your claims go directly to sipro.com (the only authorized agent from the g729 consortium for licensing) if you are an end user. No matter what quantity you want to get. So your whole tirade misses the point. Again. What's my point? If you're willing to deal in real volumes, the $10/transcode license fee doesn't apply. You can either go directly to AudioCodes and negotiate a better fee ($1.25 is the number you're stating) or you have already paid the fee in fixed costs of the hardware you've got in order to be able to terminate that kind of call volume. audiocodes doesnt sell the licenses, Sipro lan telecom inc does. I wonder if that is where you went for your proof earlier that digium cant (despite kevins statement that digium does) change the licenses more than once. ... So we're arguing the same point? I dont know what you are trying to say, you keep commenting on stuff I am not saying. Your example was invalid, because no sane person running a business with that many concurrent calls will be transcoding them on PCs; they'll be terminating its invalid becuase everyone that runs a business of any size is sane? I disagree, but will let that go. Of course if the opening statement is invalid what does that say about the counter argument? You further are STILL ignoring what the context was that the original post was in reply to, and the reply to attempt to correct your bad information and in some cases even flat out wrong information (as disputed by digium employees). But hey you are entitled to your own delusions. You seem bent on proving me wrong even if that means misquoting, lying, making stuff up, or just being delusional. So I will let you have that victory, after all if you are willing to fight this hard to be right you must not be able to be right that often. You win I am wrong, digium is wrong and the g729 authorized agent (the only one) is wrong. I will even concede to the fact that the context I originally intended when I replied was wrong and that it was the undisclosed one that you brought up later. you win, let it go -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Size limitations of extensions.conf
On Mon, 2006-06-05 at 11:24 -0500, Moises Silva wrote: Asterisk support the concept of configuration engine, this means that you can write a configuration engine to get the data from anywhere. The default configuration engine is text_file_engine, that reads the configuration from text files. This engine does not have any limit in the code, so the only limit is the performance hit of starting or reloading. Actually some limits exists for the size of context names, nested includes etc, but no for number of lines. Why dont use database engine? instead of large files? Regards That doesnt solve the root problem. The configuration engine would be called at startup/reload and load everything into memory. A better approach might be what some have done in private patches to read in realtime each customers and only each customers information on a per needed basis. Lets say you have a 'cluster' of asterisk servers that all feed off the same database and in total you have several thousand hosted customers. Each customer has many lines, possibly several hundred lines that would form their extensions.conf entries. If you load that into memory you are consuming a lot of memory on each box of your 'cluster' that will mostly be unused because not all of your customers will receive calls on all the same systems. Sifting through all of this the way that is done also has a certain cost (although I dont know the actual cost of doing it in a DB). If you load each customers configuration on demand you lower the memory footprint and add the ability to have easier updates all at the same time. You can do some of this with realtime dialplans, but you cant get the faster processing available that the one patch that does all of this gets. The way the dialplan works has its own problems (internally to asterisk) with large extensions.conf. For reference the one I am talking about would be close to 200M if printed out, and a real pain to manage via any of the standard 'engines'. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] porting g729 licenses to another machine
On Mon, 2006-06-05 at 12:05 -0500, Kevin P. Fleming wrote: I have proposed that a number of times internally, only to be told (vehemently) that customers would never go for it. That includes responses from our distributors and channel partners, among others. It would also dramatically increase the cost for people buying one or two licenses, so it would have be an 'alternate' registration means if it existed. In addition there is the lag time between ordering and receiving the dongle. That could be worked around with a time expired software only license that is good for say 10 days so you can get running right away... Dongles arent free and they would add a bit, likely $5-10, to the total cost. For 1-2 codecs not worth it. However if you have a problem during digium off hours you can use the generic example program I wrote (for other reasons it just appears to work for this although I never tested it on a system with digium g729 codec installed nor did anyone else that I am aware of) will let you change the MAC address only for the instance of asterisk that you choose. This means that on a network view the system is using its real MAC addr, for all programs except the ones you selected it would use the real MAC addr, but for the one process you choose (my example uses ifconfig becuase that is easy to instantly verify if it works) it will use an alternate MAC addr. This is handy if you are on the other side of the world from digium and digium is closed when your hardware fails. The program itself was written as an example of how to do this type of remapping, it just appears to also work for this application. It is my belief that this tool will not aid in piracy at all because people that want a free g729 license will look at cisco (they run a project which has an open g729 implementation for non-commercial purposes) or the intel IPP stuff which is already ported to asterisk. They are buying the digium modules to get the license since the codec is already out there. That means they want the licenses rather than a free ride. I also dont know for a fact that it works on the digium codecs, only a hunch because there are only so many ways to get the MAC addr. Now that you have read this far here is the link :) http://www.0xdecafbad.com/Remapping-function-calls.html -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Mon, 2006-06-05 at 13:47 -0400, Matt Florell wrote: What are the reasons that people/companies/manufacturers use G729 instead of comperable codecs like GSM or Speex? Microsoft and Apple both support GSM in their software, and Speex is the same compression ratio as G729 yet is BSD-like licensed so no cost whatsoever. speex isnt in all ATAs and other things. So if its not there it offers worse compression since the call wont go through :P -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Tue, 2006-06-06 at 00:36 +0300, Tzafrir Cohen wrote: On Mon, Jun 05, 2006 at 04:32:29PM -0400, Cory Andrews wrote: Voiceage in Montreal is supposed to be working on an open source G.729A codec, although it mentions only that they allow developers to freely use their G.729(A) codec object code for non-commercial purposes. Source code availble does not mean open source. Free software and patents usually don't mix very well unless someone is willing to grant a generous grant. And we already have the source of the intel codec to stare at. voiceage is the backer of the open g729 prtoject as seen on vovida.org (cisccos open telephony platform) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Tue, 2006-06-06 at 08:41 +0800, Steve Underwood wrote: Cory Andrews wrote: Voiceage in Montreal is supposed to be working on an open source G.729A codec, although it mentions only that they allow developers to freely use their G.729(A) codec object code for non-commercial purposes. Lots of good codec related info here http://datacompression.info/Speech.shtml Voiceage have for several years had something called Open G.729. This is like the use of the term Open Systems in the early 90s - a term designed to distract attention fom the reality of it being totally closed. There is no source, and the last time I looked it would only run on Windows. It may only be used for experimental purposes, and is pretty much as open as a welded shut box. http://www.vovida.org/applications/downloads/G729A/ that links to voiceage.com http://www.voiceage.com/freecodecs.php there is a download button there for their open g729(A) which when clicked gives you pdf file, c files and a .lib and exe files. They give you a library but no code for that and their .c files call that library. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] show channel issue with 1.2.9
has anyone else noticed what appears to be a threading issue in asterisk 1.2.9 (it broke sometime between 1.2.4 and 1.2.9) where if you have about 50 calls and do asterisk -rx show channels it will display the header but nothing about channels, total calls, active calls, etc. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Sat, 2006-06-03 at 04:01 -0400, Chris Mason (Lists) wrote: I have no problem with paying Digium the $10 for G729 licenses, everyone has to make money. It's the administration of the licenses that sucks. I experiment with different hardware a lot, and make up demo machines to install for customers with available hardware. I have to put G729 licenses on them, usually $100 each time, and when I install the real hardware for the client, I can't transfer the licenses. If I scrap that machine or change the interfaces, that's a $100 loss. I believe when you buy a number of licenses, that should determine how many instances you can use, regardless of how you want to deploy them. In short, the method of enforcement is poor and leads to resentment from customers. Surely Digium can construct a better system? well you could patch your system to allow mac addr changes, which is not always a good thing since you may have the real mac address on a different system, you could write a library that is loaded with LD_PRELOAD or whatever that maps certain calls used by the codec to get the mac address so it always returns what you want (systrace is an example of remapping calls like this although that is likely overkill for this example). You could change the codec binary to bypass the check, its only a couple bytes. All of this lets you alter the licensing model, I am not saying to do this to avoid getting licenses in the first place (and in fact the first two methods require a valid license at some point) but it lets you decide which system that license runs on. I didnt see a EULA that forbids this, but then I didnt look either, so you may want to be careful with that if you choose any of these methods. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Fri, 2006-06-02 at 12:12 -0400, Andrew Kohlsmith wrote: The Intel g729 code is licensed for educational use ONLY. Commercial use is forbidden without paying the patent holder. $10 a port won't break the bank of any business with a shred of a hope of a chance of surviving, and you stay legitimate. but $10 only gets you one license, what if you are vonage sized and need to support a million customers? What if you accept that you can settle for a 5:1 ratio, then its only 200,000 or $2M. Just for codec licenses, not to mention all the other costs of being a business. What if you are smaller than vonage, say 10k channels in use, then that smaller entity, probably without the hundreds of millions of VC that vonage got you would have to come up with $100k. Still more than $10. If you are going to bring businesses into it, at least accept that a business would most likely pay more than $10 for their licensing needs. And the inten IPP g729 stuff isnt licensed at all, educational or otherwise. Read the information from intel on that. While it is generally accepted that educational uses can use patents without a license that doesnt always guarantee that fact. Try buying a legit g729 license from the patent holder if you're a home user or small business wanting to transcode g729. They only want to license hundreds of instances at a time, if not thousands. Digium negotiated a pretty damn good license fee so that they could offer the codec and sell it in onesie-twosie quantities to little guys like us at an affordable price. no its not that they want quantity becuase they will sell just one license, they only want to deal with people that implement the systems not the end users of the system. They claim the reasoning for this is to make it easier for end users to know that they have licenses - basically if you have it you are licensed. Even if that isnt the case. Check www.sipro.com for more info on g729 licensing. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Size limitations of extensions.conf
On Sat, 2006-06-03 at 16:03 -0500, voiplist wrote: Can someone tell me the size (or any other) limitations for the extensions.conf? We have managed to keep our file pretty small thanks to AGI but we are about to setup a bunch of call restrictions based on area and country code. One line per area code in the US alone adds a LOT of text to this file. Is it a bad thing to have 5 or 6000 lines of text in your extensions.conf on a production system? Will it affect the performance? it adds memory and increases load time, it also causes asterisk to walk a longer tree each time it has to do something in that context at least rather than not ... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme versus app_conference
On Sat, 2006-06-03 at 23:20 -0500, Erick Perez wrote: As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)? Thanks, app_conference doesnt require a timer unlike meetme app_conference claimed (I dont know if meetme has upgraded) that it only transcodes once per codec in question for everyone where meetme would transcode for each person. IE you have 3 callers, 1 on GSM 2 on speex. Any frames from the GSM caller get transcoded twice, one for each participant using speex. With app_conference it will transcode once and send the same frame to both callers - so its slightly more efficient in that aspect. meetme I believe has some additional functionality, such as the menu system. I dont know if app_conference has added in the DTMF detection stuff to add menus or not. I believe that there is a mysql/postgress addon to app_conference that sticks the info about the current users in a database in realtime that way you can see who is on, even comes with a web based example php program to pull this info and display it to callers. I dont know where this modification is offhand. For any given one situation one is probably better than the other, however becuase they work slightly differently you may have to use one over the other since they dont afaik support identical features. I have heard rumors but no facts that app_conference generally can support a higher caller load too. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom-Asterisk hints/presence
On Thu, 2006-06-01 at 21:16 -0700, Mike Fedyk wrote: The way asterisk works is it abstracts concepts from protocol details. sorta, it would be better if it actually abstracted everything so that applications (modules prefixed with app_ generally) dont have to know much, if anything, about a channel (which is the running implementation of a given protocol as channels use a given protocol). There would be a standard interface to tell the channel that some event occured instead of having some apps require channel specific modules to be loaded for symbols to resolve, but meh I digress... As for monitoring other extensions, that may be difficult becuase not every channel type has support for these types of events. If you have a SIP phone that supports BLF you might be able to do something like: exten = 123,hint,ZAP/1 although I havent tried this and do not know if it works at all. I have had issues with it working with other channel types though, but it works with SIP only clients just fine. I do know that there is a patch now to monitor via BLF calls that are parked. http://bugs.digium.com/view.php?id=5779 The only thing I am unsure of is how other channels will go into SubscribeContext as defined in sip.conf, but if that is the only limitation then it shouldnt be that hard. About line 5020 in chan_sip.c there is the part where it looks for the hints and sends the Subscription-state and that can be adapted (although I havent looked in enough detail to see how easy this would be) for other channel types. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Restricting amount of incoming calls
On Fri, 2006-06-02 at 14:56 -0500, Erick Perez wrote: If i get a 8XX number, my provider told me that they will send all the calls he gets. But due to bandwidth and asterisk capacitiy in this particular installation, the system is only able to handle 27 incoming calls. How in my dialplan do I regulate, sending a busy signal, when my system hits 27 incoming calls? thanks, look up setgroup/checkgroup and decide if you want to take the call based on that :) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Fri, 2006-06-02 at 22:49 +0200, Alejandro Vargas wrote: 2006/6/2, Leon Sun [EMAIL PROTECTED]: 10$/channel If you are connecting a device that uses g729 with another that don't support it... let's say it uses gsm. Then you will use 2 channels, one for encoding and one for decoding. Is it? one g729 license can encode, decode or both. The g729 consortium doesnt differentiate. However if you have 2 channels (ie call to A bridged to B) and asterisk is doing any translations between them then it will use a license for any translations that it does. Normally if A calls B via an asterisk box and both are g.729 then it just pushes bits and doesnt use a license. To create a contrived setup, lets say that A calls B both are g.729 and asterisk is playing some sound files in between, meaning its not just pushing bits but has to insert data, say from a gsm file which it must transcode, then it will use 2 licenses, one for each call leg. Although a meetme example would probably be better than what I just gave, but meh ... In other words the g729 consortium liceneses per channel, regardless of whether its encoding, decoding or both. A channel is typically an individual call leg. In your example, if you have 1 G.729 channel and one non-G.729 channel you should only need 1 license to handle that channel. The only time you would need 2 is if you have sound files in g.729 format that you are playing to both callers (or at least to the one that doesnt use g.729). Then you would need an additional G.729 license to decode that file. This probably just added to the confusion, I cant say ... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com/ We pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Fri, 2006-06-02 at 14:18 -0700, Lee Howard wrote: Andrew took issue with my initial sarcastic comment because this thread involves the G.729 codec - but remember that if someone does ultimately choose to obtain a license illegally that they're not cheating *Digium* - rather, they're cheating the patent holder (which is not Digium). technically they are doing both becuase the patent holder only wants like $0.25/license if bought in quantity. The remainder goes to digium (or whomever else is selling them). Currently afaik the only ones for sale are digium, but trxtel.com is working with another company to offer floating licenses for g.729 for asterisk (and otherp latforms) making the licenses more portable and even cheaper (estimated price now is about $4-5 per license). The community pool on the licenses is also a nice feature where your license server talks to a master server which can then let you share unused licenses with others and when you need more than you have you can borrow licenses from others. But meh. Estimated delivery time on that is about 2-3 months. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Fri, 2006-06-02 at 14:57 -0700, Lee Howard wrote: As an example: Company X sells PCs with pirated copies of Windows that, following proper and normal channels, they should have purchased from Distributor Y. Microsoft sues Company X and wins a court judgment against Company X. According to your interpretation Distributor Y was also *cheated* out of money. Yet, Distributor Y has no legal recourse to any monies from Microsoft or Company X. Why? Because Distributor Y has no exclusivity agreement with Microsoft guaranteeing it sole distribution rights to the product. I'm not trying to make a case to justify illegal activities. However, I don't think that a vendor has any claim to losses such as we're discussing. I was speaking not of a legal issue but of a moral one which is what this thread has been about for a while. People are saying that becuase digium sells ABE as a commercial product, er I mean becuase they allow GPL users to be their test bed for the software that people somehow are obligated to pay digium. In that context, and that was the specific context to what I was replying, and if you are of a mind to believe that beucase a company has given GPL code out means that you should buy things from them then you are cheating digium out of money by not buying their addons. If however you want to take a legal argument, which it appears that you want to change the context of the previous messages to be that, we can now at this time change it, rather than trying to change it after the fact to somehow prove a point. In your example they said that they should have bought from distributor Y if that statement were true then Y would have suffered loss and could go after X. If you did not mean to say they should have bought from Y but instead they might have bought from Y then you are correct that Y has no loss claim. But I cant make that determination because I can only go by what you did actually say instead of trying to make it up as I go along. In the case of the asterisk g.729 codec there are only 3 versions of it afaik. One is from digium (licensed and for sale), the Intel IPP stuff, which is being distributed in binary form with intel code that isnt paid for (they require money if you want to distribute) and with no license from the G729 consortium, and another version which afaik isnt publicly available. Its easier for people to take the IPP stuff than to modify the codec from digium (its really easy to disable the license stuff though) so lets focus on that. In that regard digium cant make an easy legal claim since there is no legal requirement that all modules come from digium. Infact to even attempt to ban commercial modules in asterisk would go against the GPL (as per the FSF on asterisk modules specifically) provided the rest of the gpl is upheld during their creation and distribution. Since that code doesnt come from digium, but instead intel only intel and the g729 consortium would have a claim (intel requires you to pay if you want to redistribute their IPP stuff). So in the original context digium was cheated out of sales (cheated isnt a legal term and not one I would use in a legal context anyway) however in the new context that you brought up they werent. Its all about the context that words are said in, changing the context after the fact doesnt change the validity of the original statements, it just looks argumentative. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] All non US 48 area codes?
On Fri, 2006-06-02 at 18:37 -0500, voiplist wrote: Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX- 3- Anything else that should be restricted if one was to restrict all calls to US 48 only The best way is to restric everything then allow what you know to be good. If you try to take the approach of allowing everything except what is denied then you are in effect saying you know everything bad and can accurately restrict it. This often proves to be a bad decision. There is a listing at astbill.com that I started and presumably they have maintained since I basically abandoned it. It has US as well as other countries in there. I know for a fact that its not complete on the global numbers, but believe it to be reasonably complete for NANPA numbers (although some wireless providers are listed as geographic, but that shouldnt affect this application). NANPA assigns the numbers, you can goto them for the information (they have lists of all assigned codes, which carrier they are assigned to and what geographic region they are from). http://www.nanpa.com/reports/reports_cocodes_assign.html That list is all the assignments from NANPA. It would be your best bet since they are much more authoritative on this than my list. I would be wary of any codes that have X for their rate center, as those may be premium since they arent assigned to a specific rate center. They may be something else as well, however once a NPA-NXX is assigned to a rate center it cannot change rate centers, so if it isnt listed odds are its some special premium service number. As for premium numbers, yes 900 is a NPA used for premium but there are local ones, while typically 976- that isnt always the case and some numbers, are assigned to lcoal carriers and have no rate center listed and can be a pay per call service (such as some in NJ that now verizon owns). If your numbers are in LIDB that should prevent most if not all billing from occuring on those numbers, and the call to those numbers will be rejected. Your carrier should be able to insert your numbers into LIDB for you. The carrier may not know LIDB by name (or at least the sales drone you talk to) but they should be familiar with call blocking for pay per call services. hope this answers your question :) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for very basic example
On Thu, 2006-06-01 at 11:18 +0200, Benjamin Stocker wrote:
At least you know to break this down into different parts, it still
amazes me how many people look at something as one big thing instead of
several smaller things that interrelate :)
you should have example config files that came with asterisk, if you
built from source you have to do 'make samples' to get them installed,
most binary packages will do this automagically.
I. Register my phone to my asterisk server, not directly to
provider.com
This has 2 parts, one set your phone to use your asterisk server.
Without any knowedge of your phone I cant say how to do this. The other
part is create an account within asterisk for that. In sip.conf you can
create sip users (examples at the end of the default file), in iax.conf
you can create iax2 users, and so on.
II. My asterisk server should ring my phone when somebody calls me
on mynumber@provider.com
You normally have to do 2 things to make your asterisk box register and
work with your provider. One is to add a register directive, ie
register = user:[EMAIL PROTECTED]/extension
the /extension is optional, if specified it will cause calls from your
provider to goto that extension, if omitted generally they goto 's'.
There are examples in at least sip.conf for this but probably iax.conf
as well. Again it depends on the protocol that your provider uses.
The other part is to create a account for your provider. This is
similar to what you would have to do with your phone. The context
declaration here will be used for inbound calls.
As for making it dial your phone, when a call comes in from your
provider. Lets say that the user account created for your provider had
context=incoming and the /extension on the register line was 123, you
could do in extensions.conf:
[incoming]
exten = 123,1,dial(SIP/25)
There are examples of this in extensions.conf.
III. Asterisk forwards my outgoing calls to provider.com
The context that you set your phone into controls what it can call. If
it has a entry like:
exten = _1NXXNXX,1,dial(SIP/myprovider/${EXTEN},90)
then anything matching that pattern (north american numbering pattern
and possibly other places too) will get sent via sip to your provider.
There are examples of this in the extensions.conf sample file as well.
A. When somebody calls me, he get's a user unavailable from
provider.com, but my asterisk server successfully registered at
provider.com:
(sip.conf)
register = user:pwd@sip.provider.com/user
does a sip show peers show that you are registered? Does the extension
at the end of the register line exist?
B. When I call a number, my asterisk server says: Failed to
authenticate on INVITE. But all login informations for provider.com
are correct.
Which leg is failing to auth? The leg from your phone to your asterisk
box or asterisk to your provider?
you only showed one entry in sip.conf, and if you think about it from
your asterisk box's perspective you have 2 people sending and receiving
calls. your phone and your provider. Think of them more or less as
equals and the rest might make sense.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058 US WA +1 360 207 0479
US NY +1 516 687 5200 FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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Re: [Asterisk-Users] INFO: TFOT book- n priorities and labels
On Wed, 2006-05-31 at 02:01 -0700, Michael Collins wrote: Regarding my earlier post about labels and the 'n' priority: The TFOT book covers the use of these. See the box on page 81 entitled Unnumbered Priorities. http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip And one of the authors of that book (Jim Van Meggelen) will be speaking at ClueCon (on asterisk topics I believe) in august if you want to talk to him in person :) for more info see http://www.cluecon.com/ -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternative to FWD
On Wed, 2006-05-31 at 19:31 -0600, Joseph wrote: Thanks for suggestion, but I'm not looking for software (spyware) type service. in that case how about www.dizzytel.com ? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recent debian packages?
On Tue, 2006-05-30 at 08:07 +0200, Attilla de Groot wrote: It may have been 2 years since I worked with Debian on production systems, but in my experience there are alot of unstable packages in unstable. So it's a bad advice to run unstable on production systems. the debian stable, testing, unstable, experimental branches mean different things on different systems. The arm series for example testing is generally really unstable and may not work, infact you may end up with package conflicts. However testing on x86 is generally pretty stable. This is admitted to by the package maintainers of the various platforms. On the x86 line stable is really stable, testing is fairly stable, unstable is questionable, depending on what exactly you have installed you may or may not experience package conflicts and experimental is most likely going to generate package conflicts. Also note that debian doesnt officially give out security updates for anything but stable. So if you dont use stable you wont get updates to any packages that arent in the stable tree from security.debian.org. This may or may not matter in your infrastructure (for example if the only network service is asterisk and you build from source you wouldnt get any updates anyway). With that said getting the source and building it is fairly painless on debian. Assuming that you have a sane build environment that is. tasksel may help you there if you arent quite sure what all is required. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 or asterisk
On Tue, 2006-05-30 at 16:03 -0400, Matt Roth wrote: mpg123 has the same problem with zombie processes as you were experiencing with MadPlay. For a scalable system, native MOH is the way to go. As per Kevin Fleming, it only introduces a slight memory overhead. mpg123 consumes CPU cycles to decompress the mp3s and in my experience, a large scale Asterisk system is much hungrier for CPU cycles than memory. To further support native vs mpg123 here is something from astertest.com - admittedly its old and may no longer be valid, but it should be useful for a reference ... 22/11/2004: Music On Hold Part 1. Today we reached 361 simultaneous SIP uLAW Music On Hold channels on that same 350$ machine. When using GSM we got up to 173. Out of curiosity we tried some optimized configs with anthm's music on hold native format patch and got to 395 channels, regardless of the codec used. We are currently investigating more ways of increasing that value. (As we think it should scale higher than app_playback). is that what you were talking about? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [ISSUE] Asterisk 1.2.8 not compiling.
On Wed, 2006-05-31 at 12:09 +1000, Peter J Dean wrote: ( cd asterisk; make clean ; make )This didn't compile ok, and outputs the following error. just a guess but did the headers change for libpri and not get installed to the same location as before, as a result you are using older headers? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hook into authentication
On Sun, 2006-05-28 at 23:41 -0400, Steve Totaro wrote: Henry J. Cobb wrote: to increase the security for remote extensions I would like to limit a sip-peer to a specific MAC address. Is it possible to hook into the authentication mechanism in asterisk and allow/deny incoming registrations? This would be only mildly useful on the same subnet and completely useless over the internet. -HJC I think it would work just fine over the internet using a bridged VPN. even on a local network this can be forged. If you cant control the device that sends this information it is user supplied data, even over a vpn (which uses a virtual interface not the physical one). It has the same value as any user supplied data - other than perhaps its additional data which makes guessing slightly harder. TLS might be a better way to go since it would require a certificate that you can control the issuance of, but that certificate can be stolen and the remote end point would need to support the same scheme that you use (fortunately there are standards that make this easier with some devices but most dont implement this). A vpn would provide security in that it would make it harder for someone to eavesdrop on the auth and attempt to derrive the password, however there is overhead associated with that. At least 1 IP packet per real packet (sometimes more) on the network side, and the crypto parts on the cpu side. For the server you would want to have a hardware based crypto card to deal with the VPN connections... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Now that Nufone is dead...
On Sat, 2006-05-27 at 12:48 -0300, Hermann Wecke wrote: Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? You may also consider Asterlink. I'm a new client there, their support is a little slow, sometimes irresponsive (you need to send several messages until they notice you), they also have a misconfigured mail server but other than these problems, so far so good. how is asterlinks mailserver not configured properly? I havent had any problems contacting asterlink people to get stuff resolved, how are you submitting queries? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone going to cluecon?
I was wondering if anyone was going to cluecon http://www.cluecon.com I would like to start by saying I am not affiliated with cluecon, or anyone speaking there. I just think this conference sounds good. For those that dont know its a telephony conference in chicago august 1-3. It will talk about various voip topics like sip, jabber, google talk, etc as well as various software products such as asterisk, bayonne, freeswitch, sofaswitch, sipx, etc. There will also be hardware vendors present to discuss things relating to physical interconnection. Speakers include one of the authors of asterisk the future of telephony Jim Van Meggelen, a presentation on asterisk modules by the guy who wrote the realtime voice changer asterisk module (i forget his name but he is on this list, or was so he can speak for himself :) Other speakers include: Kevin Lenzo - Co-Founder of Cepstral Text-To-Speech software. RJ Auburn - Voxeo Speech Recognition. Peter Nixon - Telephony Entrepreneur Nenad Corbic - Engineer, Sangoma Technologies Craig Southeren - Co-Author, Lead developer OpenH323/OPAL David Sugar - Lead Developer GNU Bayonne Derek Smithies - Developer OpenH323/OPAL/SOFA Anthony Minessale II - Author of FreeSWITCH, Major Asterisk Developer, CTO Asterlink ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
On 3/29/06, Matt [EMAIL PROTECTED] wrote: Hi, Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? IE... If your name is Joe Smith you can't have Mary Smith set as the caller-id name, unless mary smith is also on your account. ___ ruling by whom and in what jurisdiction? AFAIK there is no global governing body for telecom, but then I dont pay enough attention so there might be :) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Numbered Voicemails even with delete option!
On Mon, 2006-03-20 at 09:32 +, David Waugh wrote:
NOTE: This is my first shell script so I'm sure it can be improved!
noted, in that spirit see notes below ...
***
[EMAIL PROTECTED] INBOX]# more /etc/asterisk/voicemail-clean
cd /var/spool/asterisk/voicemail/default/1234/INBOX
this appears to be redundant since you specify the full path in the find
below ... It doesnt hurt anything though.
#Only move files that are not currently in use that are over 3 bytes
find /var/spool/asterisk/voicemail/default/1234/INBOX -mmin +1 -and
-size +3c -exec cp {}
/tmp \;
why cp em to /tmp? Seems a waste given what you do with the files
in /tmp later...
#replace contents of these files with 0 to save space
#find /tmp -name 'msg*.*' -and -type f -exec echo 0 {} \;
for i in /tmp/msg*.gsm
do
echo 0 $i
done
for i in /tmp/msg*.txt
do
echo 0 $i
done
if [ -f /tmp/msg\*.txt ]
then
rm -f msg\\*.txt
rm -f msg\\*.gsm
fi
after putting a 0\n in each file you then rm it without doing anything
else ... Why waste the time copying them earlier, then making the files
contain only 0\n just to rm em?
#delete any wav or WAV files
rm -f /tmp/*.wav
rm -f /tmp/*.WAV
#Delete any files that are not currently being used
find /var/spool/asterisk/voicemail/default/1234/INBOX -mmin +1 -and
-type f -and -size +3
c -exec rm -f {} \;
#Copy our changed files back to the directory to fool asterisk!
cp -f /tmp/msg*.* /var/spool/asterisk/voicemail/default/1234/INBOX/
rm -rf /tmp/msg*.*
But if those exited they were deleted above ...
#Cleanup
rm -f /var/spool/asterisk/voicemail/default/1234/INBOX/msg\*.gsm
rm -f /var/spool/asterisk/voicemail/default/1234/INBOX/msg\*.txt
If by some chance they were able to survive the previous copies,
deletes, then copied again, you make sure they dont survive any
further :)
how about this, would it do what you want (note I am basically using
what you started out with) It also makes em 0 bytes instead of 2 :)
And it works on more than one user at a time, although that may not be
desired.
The size +3c may need to be altered since it wont have 2 bytes, but it
shouldnt hurt anything to leave it, I left it becuase that is what you
started out doing, although for other reasons.
touch /tmp/vm.$$
for i in /var/spool/asterisk/voicemail/default/*; do
find $i/INBOX -mmin +1 -and -size +3c -and -name \*.wav \
-exec rm {} \;
find $i/INBOX -mmin +1 -and -size +3c -and -name \*.WAV \
-exec rm {} \;
find $i/INBOX -mmin +1 -and -size +3c \
-exec cp -f /tmp/vm.$$ {} \;
done
rm /tmp/vm.$$
This of course could still be optimized further, but to keep it simple I
decided to use what you originally did as a base...
app_voicemail can detect a gap in the sequencing between any rm and
creation of a replacement file, so I create a dummy file in /tmp then cp
that over the desired file to avoid that.
find is not that processor friendly so you will want to watch out if you
have a large number of users/voicemails.
I also dont see that big of a point in doing this, after how many years
and hundreds of thousands of voicemails that a user has listened to do
you finally reset that number? or do you want a life count forever?
Further the way app_voicemail works the larger that directory is the
more processing that is required to find the next available sequence
number ...
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058 US WA +1 360 207 0479
US NY +1 516 687 5200 FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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Re: [Asterisk-Users] [Fwd: Over 40 destinations for FREE!]
On Thu, 2006-03-02 at 17:51 +0100, Francesco Peeters (Asterisk) wrote: Just in my Inbox: From the makers of Voipbuster: http://www.internetcalls.com Over 40 FREE destinations, PLUS free VoipIn number AND Call Forwarding! Finerea has sipdiscount.com which also is offering the same deal. it appears they have peaked now and are mailing everyone off all their family of sites. I got one a while back for um something other than voipbuster I forget which of the 10 companies they operate (all basically the same deal). sipdiscount still makes you sign up with their stupid windows client but it freely gives you the sip settings so you dont have to guess if its sip.voipstunt.com or connserver.whatever or ... my guess is they are deprecating the other sites soon becuase they seem to really want to push internetcalls.com ... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast Ireland +44 02890 996 461 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Streaming Music On Hold
On Thu, 2006-02-23 at 08:09 +, Lee Archer wrote: I spent a days or two on this and in the end did Musiconhold.conf [livestream1] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@ /etc/asterisk/stream.playlist Then in stream.playlist I just put the links from Shoutcast I wanted to use That is useful to know, however that isnt the 'pls' play list files, they contain extra info over and above the url. however if one were to do something as simple as: grep ^File *.pls | cut -d= -f2- stream.playlist then they could use your information with pls files :) -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username as extension
On Thu, 2006-02-23 at 16:46 +0800, Nathan Alberti wrote: Is there a way to have extensions automatically created for registered sip users ? in sip.conf regcontext=sipregistrations that adds them to sipregistrations, you can make that anything you want however I am willing to bet there might be problems if its named something else (although that may not be the case). -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loops and Variables
On Sun, 2006-02-19 at 20:30 -0500, Doug Lytle wrote: I have the following in my dialplan, counts the number of loops and when it hits greater then 5, exit. It works, but errors initially with, syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or tolken; Input: +1. Could somebody tell me why? is count defined before it tries to do count + 1? if count is null you will see a parser error like that becuase it evaluates to count = + 1 -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loops and Variables
On Sun, 2006-02-19 at 21:05 -0500, Doug Lytle wrote: trixter aka Bret McDanel wrote: Could somebody tell me why? is count defined before it tries to do count + 1? No it isn't, thank you for the clue. I'll define it. since you have had a little time to play with this, was this the problem? -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] free tollfree termination
http://www.trxtel.com/index.php?page=Tollfree_Termination This is a free service, I am not selling anything with this service. I just thought that individuals that read this list may enjoy getting tollfree access free this way (yet another way) given that it lets you send your caller id and some of the other free providers dont let you do that. Starting a test service now, for individuals free north american tollfree termination. For carriers that do large quantities of minutes (a not really defined term as yet, more a negotiated value) we will share revenue with you for sending calls to us. If you set up IP PBX systems for customers, add a route in and make residuals off those customers. Run a ITSP? Get revenue for each minute that a customer dials a north american toll free. If anyone has any problems using the service I would appreciate hearing about it, the service will remain free even after the test period, however to get compensation requires an account so that it can be uniquely tracked. Granted tollfree traffic isnt usually the bulk of a provider, but at least now you can provide it free to your customers without losing on costs like bandwidth :) -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sipdiscount
On Fri, 2006-02-17 at 14:32 +0100, Alejandro Vargas wrote: 2006/2/17, adibar [EMAIL PROTECTED]: I would sugest, that you just register without balancing your account. Than use the supplied username/password and it will work. I doubt that the test/test works. Thanks. This worked. I already had a sipdiscoutn account without credit, but It never worked before (always needed to use test). they may have recently disabled the test account given that if everyone is using it abuse would be high. While a free account does little to stop abuse, it does add a very small hurdle to it, which can slow people down and potentially add for slightly better tracking of problem users. -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 trunking known problems?
I am curious if anyone has had problems trunking iax2 with 100+ concurrent calls. I am planning on testing this out tomorrow, however I wanted to know if anyone else has had a problem with this prior to my test so I know what to look for if anything is known and what resolutions have been found if there are any known problems. Specifically I am doing this on fbsd 6 with asterisk 1.2.4 using g.729 (bandwidth calculators suggest 0.8Mbps used with this setup, compared to the 2.3Mbps (calculated averaging 2.25Mbps so its close enough). That is a considerable savings, not to mention that there will be fewer sends/recvs so hopefully less irq time spent accessing the ethernet card :) I am just concerned with packet sizes in call quantities this large, and other factors. Anyone that has done this before I would appreciate hearing from you. -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 trunking known problems?
On Thu, 2006-02-16 at 13:38 +0200, Zoa wrote: A long time ago i tried to make one big iax2 trunk for one of my customers, i soon changed this to several small trunks. (bandwith doesnt rise all that much if you use 2 trunks instead of 1.) Asterisk didnt seem to like my big trunk very much (i don't remember how big it was, but probably over 100 calls). Its a very long time ago, maybe some of those issues are resolved by now. Zoa Do you recall the specific nature of the problems? -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 trunking known problems?
On Thu, 2006-02-16 at 14:04 +0200, Zoa wrote: I think, but am not sure, that with a lot of calls inside the trunk, some calls seemed to go suddenly go outside of the trunk in one or more directions, bursts of error messages appeared on the cli etc. i didnt investigate it a lot more, my problems went away with splitting them up in smaller trunks. I will watch for that both end points are 1.2.4 so it seems that this level of load testing is something that would be handy to have. How large were the smaller trunks? I am ultimtely looking at doing about 300-600 channels eventually, and this type of an issue would be nice to know of beforehand :) -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 trunking known problems?
On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote: The trunks were made to be maximum 60 simultaneous channels iirc. I doubt seriously you will be able to do 600 simultaneous on any system. (with or without trunking). (at least out of the box). Zoa At 100 with g.729 its running 95% idle, in theory this box should be able to handle it. We shall see :) I also dont run asterisk 'out of the box' but that is a different story. Thanks for your help so far, I will post back when I have more real info, unless someone else can post before then with their experiences on 100 trunked calls. -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 trunking known problems?
On Thu, 2006-02-16 at 14:54 +0200, Zoa wrote: When you have a lot of calls, try doing a show channels and iax2 trunk debug. (they are killers) Zoa not having trunks set up yet, I dont do the latter but I do the former all the time. Mostly becuase this is a new server and I wanted to make sure that its actually having calls go through and not just dropping them sitting idle. It only scrolls the screen but doesnt appear to cause any real cpu load that I noticed. Unless you were talking about something else ... -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 trunking known problems?
On Thu, 2006-02-16 at 14:58 +0200, yusuf wrote: also doing IAX2 trunking. What do yuo mean you dont run asterisk out of the box. Also want to know what is you bandwith usage for 100 calls and g729 I run a modified version of asterisk. There are a few things that I felt needed to be added, changed, deleted, etc. With SIP I am using about 2.25Mbps for 100 calls. Asteriskguru.com's bandwidth calculator said that would be 2.3Mbps so its really close to what I saw. There are a variety of reasons why I averaged lower, but still less than 3% difference is very close and acceptable. With iax2 trunking asteriskguru.com's bandwidth calculator said that it should drop to about 0.8Mbps or about 1/3 that of SIP for that codec. That is a considerable savings and doesnt count the savings on the ATM layer (about 500kbps because of the 8 byte trailer per packet sent) not does it count a few other things. All in all its enough to justify playing with it to see if it works with this load. I was just concerned becuase I recall someone saying they had problems with 20 calls trunked, although I didnt pay attention at the time and didnt catch the details. Zoa said that 60 per trunk works, or did for him, so that certainly counters what I remember being said not that long ago on this list. -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] show calls
On Thu, 2006-02-16 at 05:46 -0800, jonny hashem wrote: HI: what is command on console to know how many calls are sending in the same time? I will guess 'show channels' -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] audio cuts out
has anyone experienced a problem where RTP audio cuts out when doing 30-40 concurrent channels via sip? The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel - not even a timing source) The box has plenty of bandwidth, when a call to the same box is iax2 it works, but when its sip a call gets connected a few frames of audio are passed and then silence. When the box is completly idle sip does not experience this problem, it is only when there are a few concurrent calls. -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
