Re: [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)
2012/1/12 Ishfaq Malik i...@pack-net.co.uk Hi I'm using 1.8.7.0 with the RealTime architecture. If a call goes into application Queue and is abandoned by the caller, no entry is made in the CDR. Entries are made into the queue log. This cannot be correct behaviour, all calls should show in the CDR. Could anyone else try to reproduce this and if others get the same thing, I'll raise a bug on it. Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fraud advice
On 10/14/10 9:10 PM, Jeff LaCoursiere wrote: Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was taken advantage of over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN trunk, and of course there was no warning until the bill came today. Unfortunately the bill only covered the first few days of this fiasco, and was only $700. I am afraid the one that is on the way will be tens of thousands. ONE CALL on the bill that just arrived was $200 (80 minutes to Sierra Leone). I'm sure this started out as a single scan. It must have been posted, because I have at least ten IP addresses now that were placing calls via the same peer. They are from all over the world. So what is the accepted procedure? I'm in the US Virgin Islands, so do I go to the FBI? Police? Is their some telecom fraud body to report such things to? Does any one ever get any relief from such events? I'm basically sick to my stomach right now. j We were hit several times in our early days with PRS fraud that ended up costing us DEARLY. We contacted the FBI, but they were completely unhelpful. The origin of the caller was Egypt (using a network in Egypt that has long been a front for criminal activity, so the networking people on that end were less than useless), and the Egyptian cyber fraud division is two guys with a yahoo email address. The FBI contacted them, but they were neither equipped nor entirely willing to be of any real help in tracking down the perpetrator. It doesn't hurt to contact the FBI, though. They may already have an open investigation into the individual or group responsible and need the information for their case. But do not expect them to be able to do much. Eventually, some of our debt was quashed by the provider who had violated their own policies in charging us for unlisted premium rate services, but it changed the entire way we do business. Unfortunately, it's now MUCH more difficult to pay us money than it used to be, and that's turned a lot of customers off, but we've had no problems with PRS fraud since. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] billsec exceeds duration on some calls
On 8/20/10 1:24 PM, A J Stiles wrote: On Wednesday 11 Aug 2010, Tilghman Lesher wrote: On Wednesday 11 August 2010 03:59:28 A J Stiles wrote: I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon. With some calls, the value in the `billsec` field in the CDR is exceeding the value in the `duration` field. I'd love to know what circumstance caused that. I agree that this should not occur. I've done some more digging about. I was getting calls in the monitor folder where the outgoing and incoming halves were different lengths; so I temporarily disabled removing them after combining them into a single file, and let them build up for a few days. There doesn't seem to be any correlation between this phenomenon and billsec being duration, though. Can anyone else with a similar setup try running a query such as SELECT COUNT(*) FROM cdr WHERE calldate LIKE 2010-08-20% AND billsecduration ; and seeing if they have any calls like this? Any chance this has something to do with your system time? Are you running ntpd, or setting time at regular intervals via a central system clock and a cron job? Again... also just stabbing in the dark. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spam blacklist
Supposedly, the filters drop it in the transaction stage. But for some reason, every time I get dropped from the list, it's just after a spam email was sent out en masse, so I'm not sure what's up there. On 7/28/10 10:43 PM, jon pounder wrote: SIP wrote: what can you do ? simple discard spam don't bounce it. On 7/28/10 9:45 PM, Sam wrote: Just a note, the asterisk mailing list server continually gets blacklisted over at http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering mail to spamtraps. Perhaps something needs to be looked into... Regards, Sam Spammers sign up to the Asterisk mailing list and send spam once in a while. My spam filter rejects it, and bounces the emails back to the Asterisk list, which then drops me from the list because it got a single bounce. Bit of a pain in the left ventricle, really, but what can you do. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spam blacklist
On 7/28/10 9:45 PM, Sam wrote: Just a note, the asterisk mailing list server continually gets blacklisted over at http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering mail to spamtraps. Perhaps something needs to be looked into... Regards, Sam Spammers sign up to the Asterisk mailing list and send spam once in a while. My spam filter rejects it, and bounces the emails back to the Asterisk list, which then drops me from the list because it got a single bounce. Bit of a pain in the left ventricle, really, but what can you do. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need USA DIDs
On 6/23/10 7:20 AM, RSCL Mumbai wrote: Hi, Looking for some reliable and quality providers of USA DIDs. Any pointers ? Thx Sans We've had good luck with Vitelity and DIDForSale.com. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small PC to build and run Asterisk
Danny Nicholas wrote: Also cheaper to replace flash card than hard drive. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Monday, June 14, 2010 4:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Small PC to build and run Asterisk Why no flash? * Small pre-built PC (not buying board, case, all parts separately) * Low power consumption * No fan or very small fan * Hard drive (not flash memory) An ssd uses less power, so generates less warmth, hence less need for fan in the drive area. Also less noise.. I like this one, or its smaller brother: http://www.fit-pc.com/web/fit-pc2/fit-pc2i-specifications/ But a flash card needs replacing more often than a hard drive. It's just not designed for the same sort of lifecycle of writes that a hard drive is. Sure, the number is always increasing as they increase the capacity, but it WILL NOT LAST. Dependent on the type of filesystem access you need, SSD could be a great choice. But if you're heavy on logging and writing small data bits here and there (which isn't always something you can control if you don't write all the software), then a hard drive is just going to be the better choice to hold up for a long period of time. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small PC to build and run Asterisk
Randy R wrote: Hi, I'm looking to build an Asterisk box that can run at a remote location. Here are most of the specs of what I'm looking for: Physical hardware * Small pre-built PC (not buying board, case, all parts separately) * Low power consumption * No fan or very small fan * Hard drive (not flash memory) Capabilities/capacity * No GUI, no X * Register to multiple SIP servers * There will be no PSTN * No analog phones * Small number of SIP devices will register - maybe 10 max * Three simultaneous channels active * Skype for Asterisk needs to run on this - so this means x86, right? Recommendations wanted * What hardware * What distro * Which Asterisk version Comments and suggestions welcome. This is going to be discussed on VUC as well, so if you're comfortable with it, come on by: http://vuc.me Thanks in advance, /r We use the Acer AspireRevo AR1600-U910H in a lot of locations. It's enough to handle a few dozen remote office employees on a full asterisk install with transcoding. 160G hard drive, couple of gigs of ram (comes with 1), gigabit networking, and it's $200. We slap CentOS 5.X on there, and Asterisk 1.4.X (we don't do 1.6.X). It DOES have a fan, but it's a very VERY quiet fan. For pure fanless, you might try the Lenovo Q110, but it will run you a bit more (should come standard with 320G hd and 2G of ram, though). We love the Acer AspireRevos. I have one at home I use as a media centre, as well. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
Jeff LaCoursiere wrote: On Thu, 20 May 2010, Gordon Henderson wrote: On Thu, 20 May 2010, SIP wrote: Even IF you could get a keyboard with lights you could individually turn on and off (never seen one), http://www.artlebedev.com/everything/optimus/ Bit expensive though... Gordon Heh. A $2400 keyboard. That's crazy. Cool though. j Indeed. It is crazy cool. And crazy expensive. :) N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
Tzafrir Cohen wrote: On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote: 2010/5/18 Danny Nicholas da...@debsinc.com Dumb question – wouldn’t it be easier to monitor a web interface than a phone with 100 lights? Yes and no : operator already has a Flash Operator Panel on its screen. Information displayed by FOP is richer (you can see who is talking to who) but operator feels easier with dedicated buttons for both displaying activity and issuing transfers. I think 100 is the upper limit for both kinds of tools where at a glance, you can see all extensions : I think above a certain user count (120 ?), operator would prefer to specifically query its console to get current specific extensions phone activity. Just a thought: I have on my desktop a hardware device with some 100 or more buttons. No leds in them, sadly[1]. Remapping their labels is normally done using specialized hardware (sticky labels and the sort). Naturally there's the alternative of a touch screen. [1] A quick search found products such as http://blog.logitech.com/2009/10/15/new-logitech-gaming-keyboard-g110/ Even IF you could get a keyboard with lights you could individually turn on and off (never seen one), good luck getting a receptionist to use it. I can picture it now... you hand your receptionist a lighted keyboard and say 'make do,' and your receptionist brains you with said keyboard when your back is next turned. There's a big difference between a workable situation and a complete and utter kludge. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI == DeadAGI
On 5/2/2010 4:52 PM, Steve Edwards wrote: On Sat, 1 May 2010, SIP wrote: [snip] We run DeadAGI for a considerable number of calls since it has the ability to run post-hangup cleanup no matter which side hangs up (unlike AGI). [snip] When a channel hangs up, Asterisk sends a SIGHUP signal to the AGI. If the AGI did not establish a handler for the SIGHUP, the AGI exits. If the AGI established a handler, the AGI can choose to ignore the signal or execute appropriate code -- like clean up files, write a CDR to the database, etc. If the AGI is started when the channel is live, you should use agi() and catch signals appropriately. If the AGI is started when the channel is dead, you should use deadagi(). Right. That's the way it works in theory, with the nice separation of AGI on live channels and DeadAGI on dead channels. But with our scripts, we use DeadAGI because the channel will redial different gateways after a live connection is made if there's a problem, and we've been unable to figure out how to get that from AGI, since, once the channel is hung up, it won't let us redial again. I'm sure it's a matter of just some little collection of things we're doing wrong, but for the moment, DeadAGI works swimmingly, so we haven't delved too deeply. We've never run into one of the supposed problems with running DeadAGI on a live channel. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI == DeadAGI
On 4/30/2010 6:03 PM, Luki wrote: It is irrelevant who hangs up, you want to just use DeadAGI in the h extension I wish that would be the case, but at least on 1.4 I see: [Apr 30 14:59:38] -- Executing [...@master-route:1] DeadAGI(...) in new stack [Apr 30 14:59:38] WARNING[27845]: res_agi.c:2160 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI The good news is, we run tens of thousands of calls every day through this box and about half of them spit out this warning, but it never caused any problems for over a year. Thus this warning is probably safe to ignore. Luki Agreed. We run DeadAGI for a considerable number of calls since it has the ability to run post-hangup cleanup no matter which side hangs up (unlike AGI). We see this warning constantly, and ignore it... constantly. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted: free DID number and provider feedback
What country are you in? Makes somewhat of a difference. N. On 3/17/2010 8:49 PM, Mike wrote: Ok, I see there's alot out there of voip providers. Curious what to watch out for ? charges and fee's, etc ? If anyone has feedback as to a GOOD voip provider experience (one that gave FREE DID) Please share. Again, I am doing this to learn about asterisk, I'm currently testing it at home. thanks, On Wed, Mar 17, 2010 at 11:49 PM, Joe Grecojgr...@ns.sol.net wrote: Hi All, Anyone one info of where I can get a 'free' DID number ? I have setup my asterisk box (home) and want to learn more but I need a #. I highly suggest http://tinyurl.com/ya9vzsa ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted: free DID number and provider feedback
Well. For a free US DID, you could check IPKall (http://www.ipkall.com/) I've no idea about quality, since it's been almost five years since I've even LOOKED at them. But then generally work well with Asterisk. And they're free. N. On 3/17/2010 9:09 PM, Mike wrote: My bad, I'm in Los angeles california usa On Thu, Mar 18, 2010 at 1:06 AM, SIPs...@arcdiv.com wrote: What country are you in? Makes somewhat of a difference. N. On 3/17/2010 8:49 PM, Mike wrote: Ok, I see there's alot out there of voip providers. Curious what to watch out for ? charges and fee's, etc ? If anyone has feedback as to a GOOD voip provider experience (one that gave FREE DID) Please share. Again, I am doing this to learn about asterisk, I'm currently testing it at home. thanks, On Wed, Mar 17, 2010 at 11:49 PM, Joe Grecojgr...@ns.sol.netwrote: Hi All, Anyone one info of where I can get a 'free' DID number ? I have setup my asterisk box (home) and want to learn more but I need a #. I highly suggest http://tinyurl.com/ya9vzsa ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI and 1.6.1
Will Payne wrote: On 8 Mar 2010, at 22:08, Dave Poirier wrote: Top posting to remain consistent... I drop litter because everyone else does. ;) W Different entirely. People who switch to bottom posting on a top-posted thread make things MUCH harder to read by being needlessly pedantic. It's like those people who decide that, even though traffic is moving along at an average of 70mph, they're going to drive 55 in the fast lane to 'teach everyone the proper speed.' They're statistically MORE likely to cause accidents (or, in LA, get shot) than those travelling along with traffic at a speed above the posted speed limit. On some positions, it is not helpful to be unwavering. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI and 1.6.1
Will Payne wrote: it just seemed like a 'I know this is wrong, but...' comment :) Quoting entire emails is bad, m'kay. Quoting whole threads is worse. If you snip the quote down to the relevant portion, you can reply where you like, regardless of what's gone on beforehand. (Surely there's no such thing as 'needlessly' pedantic - all pedantry is necessary :) W Unless it's errant. Then you upset Churchill. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ideasip
David @ULC wrote: I use IdeaSip with IPKall. How may channels are open when we use IdeaSip ? Incoming IdeaSIP SIP channels are unlimited; however, I believe IPKall limits you to 94 channels via their DIDs. You would, of course, need the bandwidth to be able to handle 94 simultaneous channels. -- Neil Fusillo CEO Infinideas, inc. http://www.ideasip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXV3140 and Xlite video
Julian Lyndon-Smith wrote: Has anyone managed to get these two phones to make a video call to each other ? If so, care to share how the hell you managed ? the GXV is at the latest firmware, and xlite the latest download Asterisk 1.4 trunk TIA Julian Yes. Have done it often. Needed the firmware in the GVX that suppoerted H264 or H263.1 or whatever it was that Xlite 3 uses. Other than that, it was rather straight-forward. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Philipp Kempgen wrote: Leif Neland schrieb: Norbert Zawodsky skrev: The number +43-1-3207978 is my telephone number. I own it as long as I pay for it. And with extra digits behind it I can do whatever I like. I can create any extension - physical or virtual. I can attach a phone to extension 12, attach a virtual fax server for extension 12 to extension 99912 or could fire up my toaster if I call extension 911. I can invent any numbering scheme for my company. That's a fact! Again - At least here in Austria !! (can't speak for other countries) Invent all you want, nobody can call those fantasy-numbers anyway. Perhaps, a fraction of a percent, who are using ENUM. Leif, ever heard of direct inward dialing and PRI? http://en.wikipedia.org/wiki/Direct_inward_dialing http://en.wikipedia.org/wiki/Primary_rate_interface You can actually own a block of numbers like 01234567. You are free to map these DID numbers to extensions or do what ever you like. And it is guaranteed that nothing in the 01234567... range will ever be assigned to a different PSTN subscriber. Philipp Kempgen Exactly. And in such a case, this is exactly what the ENUM DNS is designed for -- handling those blocks. NOT for creating additional digits on top of one existing number. It may work in Austria, and may even be valid in Austria. But if that's the case, it's because Austrian dialing is a complete hack -- NOT because that's the way it's intended OR designed. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Benny Amorsen wrote: SIP s...@arcdiv.com writes: It may work in Austria, and may even be valid in Austria. But if that's the case, it's because Austrian dialing is a complete hack -- NOT because that's the way it's intended OR designed. Err no? It's perfectly sane, and it was intended and designed that way. You are providing no justification at all for your opinion that it is a hack. It is quite apparent where the hack is in this thread. /Benny Adding random digits to a PSTN and expecting to get the same person at a different extension you don't think that's a hack? I do. One should generally assume that my posts are my opinion, which really doesn't need any more justification than I think so. If I have to start putting legal disclaimers at the bottom of my posts stating everything in this post is considered the opinion of the poster and should not be presumed to be the opinion of anyone else just to make sure that's understood, I'll be happy to do so, but it will be both silly and pointless. An ENUM number is a mapping of an E164 number to a service. If, in Austria, you 'own' all numbers that are your PSTN number plus any string of random digits, that's great... but it doesn't work that way in most of the rest of the world (I'd wager ALL of the rest of the world, but that's based again on supposition without accompanying hard data and spreadsheets and pie charts), and if the incumbent telecoms in the world thought people would use that methodology to cheat them out of money for additonal DIDs, they'd clamp down on it in a hurry. But regardless of how they do things in Austria, the fact remains that the original poster was asking a question about how to configure ENUM so that his phone extensions reached his correct targets. The suggestion by the registrar was only to register the main number. Suggestions by myself and some others were to register the main number and any other numbers. It was never stated that this was a limited subset country in which dialing codes are relaxed and non-standard, and so the information provided was, perhaps, not acceptable for the use case given the data provided. For making assumptions based on limited information, I apologise. But for thinking that such a dialing system is a distortion of the ENUM concept? I don't apologise one bit. And if you choose to go above 15 digits, you're violating not just ENUM, but the E in ENUM, since an E164 number is limited to 15 digits. N. The opinions written herein are the sole opinions of the poster of this email and should not be construed to be the opinions of anyone else mentioned here, even though there wasn't anyone else mentioned here to whose these opinions could be ascribed. This email and all its contents are considered private and only shared by the consent of the original poster. If this email was delivered to any unintended recipients, please delete the email and all records from any email servers it passed through. Also be sure to take the necessary alcohol, recreational drugs, or lobotomies required to remove all knowledge and memory of this email's contents. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Raimund Sacherer wrote: Adding random digits to a PSTN and expecting to get the same person at a different extension you don't think that's a hack? I do. One should Sorry, please do not call a whole country using a hack when their solution is legitimate. Austrian PSTN https://supportforums.cisco.com/docs/DOC-1343 Excerpt: Dialplan The Austrian dialplan is a variable length numbering plan, which consists of area codes and subscribers numbers. Area Codes are from 1 to 4 digits in length, subscriber numbers vary even more. Some Numbers (mostly used by companies) allow the use of Direct Inward Dial Extension. If such a number is used, it is up to the company to decide on the length of extensions, meaning that any length can be used.If too long extensions are used, the numbers might not be reachable from some sources. This is especially relevant with regard to number presentation inside of isdn, as stated under Voice Interfaces and signalling. A copy of the Austrian dialplan is maintained by the austrian regulatory council at http://www.rtr.at/en/tk/nationaleRufnummern. The emphasis on variable length numbering, if you consider that austria is a small country which is physically not able to harbor more than 16 million peoples in a sane way, the numbering plan is more then sufficient, austria is not like germany with then 88 million inhabitants which needed a reconfiguration of their numbering plan some years ago. my 2cents Raimund It may be an effective hack, but it's a hack. I'm not saying the people of Austria are hacking their phone system. That's an entirely different definition of the word 'hack.' And the application here is in terms of ENUM... which is an E164 mapping system. When you start playing fast and loose with the E164 numbering scheme (going above 15 digits, for example), I don't care how cool or useful it is, it's a violation of a standard. Standards are there for a reason... so people can create order from chaos. Adding in extra chaos just to satisfy one particular subset of people is still a bad idea. I don't care if the Austrian telecom advertises that everyone can make his own 16-digit number for the price of a can of cheez-whiz and a can of soup, the simple fact is ENUM is designed to map E164 numbers to services and a 16-digit number is NOT a valid E164 number. Doesn't matter if the entire country of Austria has one, it's still not valid as per the description of E164. You can say I'm being insensitive to Austrians or whatever you'd like, but that's not the case. I honestly think it's a neat idea what they've done with their numbering plan. But it's still a hack. And you can't violate a standard and then ask why your newly-devised rules don't work in situations applied to the standard. E164 - max 15 digits ENUM - E164 mapping as a UNIQUE identifier for services. 16-digit Austrian number != E164. Therefore, attempting to ascertain why a 16-digit number doesn't work well with ENUM should be a bit of a no-brainer. And with 11-15-digit numbers, you're still playing fast and loose with the concept of 'unique' in the whole unique identifier bit. When 20 numbers essentially map to the same thing, it's no longer unique. It's only unique-ish. Quote more from Austrian regulations. Please. It doesn't make their solution any less of a hack. It just makes it a widely-accepted and intentional hack. Again... it's a neat hack. It's a cool hack. But it's still a hack. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
John Novack wrote: Raimund Sacherer wrote: Adding random digits to a PSTN and expecting to get the same person at a different extension you don't think that's a hack? I do. One should Sorry, please do not call a whole country using a hack when their solution is legitimate. Austrian PSTN https://supportforums.cisco.com/docs/DOC-1343 Excerpt: Dialplan The Austrian dialplan is a variable length numbering plan, which consists of area codes and subscribers numbers. Area Codes are from 1 to 4 digits in length, subscriber numbers vary even more. Some Numbers (mostly used by companies) allow the use of Direct Inward Dial Extension. If such a number is used, it is up to the company to decide on the length of extensions, meaning that any length can be used.If too long extensions are used, the numbers might not be reachable from some sources. This is especially relevant with regard to number presentation inside of isdn, as stated under Voice Interfaces and signalling. A copy of the Austrian dialplan is maintained by the austrian regulatory council at http://www.rtr.at/en/tk/nationaleRufnummern. The emphasis on variable length numbering, if you consider that austria is a small country which is physically not able to harbor more than 16 million peoples in a sane way, the numbering plan is more then sufficient, austria is not like germany with then 88 million inhabitants which needed a reconfiguration of their numbering plan some years ago. my 2cents Raimund For those who have no background in telecom history, this may seem strange, but in fact in bygone years this was not JUST Austria that had this scheme. In the Electro-mechanical days what some of us know as Direct Inward Dialing ( not the DID term often misused in modern times ) was handled this way in open numbering plans. It is unfortunate, but all too common, that in a great many fields,very smart educated people are ignorant of the history of their field, and are doomed to re-invent the wheel, or proceed down a blind alley. By the time telephone operators began to be replaced by mechanical switches, open numbering plans became impossible to design for. Once software switches came about and massive modifications to numbering plans became as easy as coding new exceptions and pushing them out to the millions of switches on the network, numbering plans had, largely, been codified to make for logical and understandable patterns. In North America, with a closed numbering plan, all numbers are of a fixed length, 10 digits. Technically the one is NOT part of the number. In earlier days, no one was needed even for toll calling to distant cities and area codes. Some, fewer each year, are able to dial within their NPA with 7 digits. with the NANP turned over to the inmates ( the state PSC's ) some locales require 11 digits for all calls, others 10 for local, 11 for toll, and others 10 digits for all calls. The closed number plan is somewhat easier to parse, with its fixed length, and no timeout or send/end digit is needed. the open plan can be more efficient in use of numbers. Different locale dialing pattens do make that more of a challenge, however. How all of this works in regard to ENUM is for others to decide, but if it cannot handle an open numbering plan with variable length numbers it needs fixing. John Novack ENUM has no issues with variable length numbering plans in its design. However, you have to stop calling it ENUM if it gets above 15 digits, since it's no longer a valid E164 numbering scheme as per the design there. ENUM is even applicable to local-only dialing plans, in which you'd run your own server, point your phones to that server as their primary ENUM server, and off you go. It's a very flexible idea as far as mapping E164 numbers to services go. However, once you get into the realm of registering with the IANA approved servers and trying to place your own ENUM DNS server into the mix, you can't start mixing local-only and public numbering schemes or things break. Austria is somewhat of a special case in which their numbering schemes are such that they allow the ad-hoc creation of additional virtual DIDs by simply tacking on digits to a valid DID. It's an open numbering plan, but from what I gather, it's a variation of the traditional open numbering plan in that each DID owner or designate gets to create his own additions instead of the telco approving all variable-length DIDs. This doesn't break any ENUM rules (unless the number exceeds 15 digits), but it does create a scenario in which it may become difficult to apply traditional ENUM tools to the scenario at hand with an attempt to get the results you're after. For instance, if user X owns the number +4311234567, and he decides he wants to create a slew of virtual DIDs after that (+4311234567[01-99]), it doesn't violate the ENUM standard because all those
Re: [asterisk-users] Please some enlightment on ENUM !!
Norbert Zawodsky wrote: But then you create phonenumbers in enum, which doesn't exist as pstn-numbers. Not the idea behind enum. On the other hand, if you owned 10 or 100 pstn-numbers in series, you could get the last one or two digits delegated to your dns-server. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Why do I create numbers in enum which doesn't exist as pstn ? A simple example: My pstn number is +43-1-1234567. Everybody around the world can call me using this number. Lets say, I have 3 extensions: 0=reception, 10=secretary, 20=boss. If someone calls ENUMLOOKUP(+4311234567) he will get a uri sip:0...@ip.of.my.asterisk ENUMLOOKUP(+43112345670) he will get a uri sip:0...@ip.of.my.asterisk ENUMLOOKUP(+431123456710) he will get a uri sip:s...@ip.of.my.asterisk or sip:1...@ip.of.my.asterisk (which ever you prefer) ENUMLOOKUP(+431123456720) he will get a uri sip:b...@ip.of.my.asterisk or sip:2...@ip.of.my.asterisk All this numbers exist because they connect to different persons. Why shouldn't that be the idea behind enum? Norbert ENUM is, quite literally, E164 Number Mapping (that's what it stands for). If you're mapping numbers which are invalid E164 numbers (i.e. in your scenario in which you're taking an E164 number and attaching digits to it), you're violating the ENUM idea for the sake of convenience. You're also making the somewhat unfounded assumption that there will never be an actual number issued (to someone else) with those extra digits. Right NOW, there may be a convention that says that you can only have 10 digits in your country's phone numbers, but that could conceivably change at some future date, and then you'd be mapping numbers that belong to someone else to your own services. The only VALID way to assign ENUM numbers is to assign numbers you actually own. Not numbers you own with additional digits. Not numbers you own with extentions tacked on. Not numbers that are similar to ones you own. But ONLY ones you own. In this case, you own +4321234567, and only THAT number should be allowed to be registered as an ENUM number. Unless you, for instance, also own +4321234568 and +4321234569 or some such... at which time you would certainly be able to register those numbers and point them to your PBX. What you're suggesting, though, violates the ENUM standard... and should not be allowed. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Norbert Zawodsky wrote: Leif Neland schrieb: - Original Message - *From:* Norbert Zawodsky mailto:norb...@zawodsky.at *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Monday, November 23, 2009 3:15 PM *Subject:* [asterisk-users] Please some enlightment on ENUM !! Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service) Assume my number was +4311234567 dig 7.6.5.4.3.2.1.1.3.4.e164.org and dig 7.6.5.4.3.2.1.1.3.4.e164.arpa both return NAPTR records. Now for the less clearer points: Your'e supposed to register your number without any extension. If I have some extensions here, how can the calling party get the correct sip uri to the requested extension? Do I have to run my own DNS server in that case? If for example if someone wants to call extension 10, is the ENUMLOOKUP(431123456710) request forwarded to my local DNS server by the e164.arpa server? Or how does that work? If everybody supported enum, it might be usefull to register extension 10 in enum, otherwise: Your extension 10 must have its own phonenumber, to be able to dial it directly. Just as with ordinary pabx. Eg: 123 555 is the reception 123 555 0010 is extension 10 Just some ideas: Is there free (as in not connected to a voisp) numbers, which can be registered in enum? Then you could use those numbers for extensions. But they would only be callable by enum. If the calling of extensions is only to be used by knowledgeable friends you could have them add your own enum-domain to their setup. Leif Hi Leif! No, I cannot believe that this was the right way. It would mean that I would have to register ( pay !!) for every single extension. BTW the How-To, the registration agent I'm using provides on his website, states, that if you're operating a PBX, you should only register your main number (=without any extensions). I *assume* that if I do an ENUMLOOKUP() of a number which includes some extension at the end, the DNS request is somehow delegated to that sub-server which is authorative over this sub-domain. This leads me to the next *assumption* that the right way would be to run an own DNS server which returns the sip-uri's for my extensions. Can someone confirm this? Norbert Yes... you would have to register (and possibly pay for, dependent on the ENUM registrar) each individual number. The idea behind ENUM is that it's an E164 number that is already yours that maps to whatever you want it to map to (email, SIP, etc). The key point here is that you already own the E164 number. If you do, then you could register them all at e164.org for free. If you don't own the individual numbers, you shouldn't be allowed to register them as your own. That sort of breaks the ENUM concept of a number you take with you as a personal identifier. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] solution for NAT issues?
Does the phone have some sort of NAT Keepalive setting? Often, the only way to keep that port open on the user's NAT gateway is to have the NATted client send the occasional data out through the port. N. Ron wrote: i have also tried setting qualify='yes' but cpu usage spiked to 100%. Ron wrote: Hi All, I been having issues on my users behind NAT, even if i hard set a specific port on the phone, there are some network that NAT's it out to a different port, in turn, some time later the phone could not be reached by the server. i think because on the server, e.g. the user is still registered on port 49923 but when the request is sent to that port the NAT router does not forward port 49923 to port of the IP phone, maybe nat mapping has expired or something. I have tried STUN, still sometimes the phones just cannot be reached. is there any other software to manage binding of ports on specific users so that the routers always keeps the port mapped to port of the ip phone . TIA Regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Eh... if VoIP fraud weren't so rampant, and I didn't constantly see mailings to the Asterisk list about How do I secure my system from the people who've been costing me tons of money lately, I would say that having a lax stance on security in exchange for additional usability might be a good thing. But as is, that's simply not the case. The 'usability' you get from this is really only questionably essential in its ability to save time, but the security one would get from a change could save some people actual money -- not just time. As someone who used to design systems and networks, I would vote for security over nebulous desire to keep the status quo. True, you can't keep stupid people from doing stupid things, but given a choice between protecting the ignorant from a bad situation or catering to those who want to avoid an extra step or two on installation, I'd side with protecting the ignorant every time. There's always a trade-off between usability and security, and I'm of the opinion that security is the more important of the two when dealing with systems connected to the Internet. Call me a cynic. :) N. Danny Nicholas wrote: Gentlemens clubs usually don't have any. While LH probably has a valid point, jumping on Til isn't the way to bring it home. You can't protect the stupid or lazy from themselves. If you can't do this right, pay someone else to. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard Sent: Thursday, November 12, 2009 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] allowguest defaults to yes for SIP Tilghman Lesher wrote: The issue in question was suspended, while the reporter makes the case on the Asterisk-dev mailing list, which is not this list. The opinions there amongst contributors (meritocracy, not democracy) are that keeping the sample configuration as it is now is probably the way to go. Sigh... of course. It's a gentlemen's club and only members have a say. If you want to create a new issue and attach your patch there, I'll look at it. I sent a patch. I pointed you at a case. That should have been FAR more than enough for my attempt at contribution to be acceptable. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 870
Remco Barendse wrote: On Fri, 30 Oct 2009, hbk wrote: Hi, I have played with the 820 for some weeks, mostly love it excellent speech quality. Even got the mini browser running showing my favorite webcam, this could be put to real use too:) Issues so far: Some embarrassing crashes while speaking, was able to speak but all freezed. Still a little fresh firmware I guess. Error 404 after showing webcam picture, but it works! Have to use *1 to start recording, record soft button does not seem to work with *. Still I recommend it, best IP phone I have tried! Not sure 870 is worth the extra money, not tested that yet. How is the build quality of the 870? The mortality rate on power supplies, diplays and the number or broken receiver hook swicthes on the lot of Snom 360's i bought 3 years ago is outright embarrassing. That's odd. We've had Snom 190s, 320s, and 360s running day in day out for years with not a single issue. Maybe we got all the good ones from your batch. If that's the case, I thank you for 'taking one for the team' as it were. ;) N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon
Sounds like it wasn't a very interesting track. ;) N. Danny Nicholas wrote: Is THAT a summary :)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R Sent: Wednesday, October 21, 2009 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Astricon On Wed, Oct 21, 2009 at 7:46 PM, Barry L. Kline blkl...@attglobal.net wrote: Randy R wrote: I missed the first part of this, but has anyone said: not all the presentations were recorded. Hi Randy. Yes, that was mentioned. Actually, three of the four tracks were videotaped IIRC. Barry And I was in the one that wasn't. So I guess I'll have to summarize... except I was a sleep one of the days :) /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls hang up after 20 seconds
Kevin P. Fleming wrote: SIP wrote: In an ideal world, when Asterisk sent an ACK, whatever server/client it was connected to would respond accordingly. It is, however, not an ideal world, so this doesn't always happen. This is not correct; there are no responses to SIP ACK messages. In addition. ACK messages are *required* for proper SIP operation; lack of an ACK to a response from Asterisk absolutely requires that Asterisk assume that either the response was never delivered to the requester, or that that requester has stopped responding. In either case, the SIP dialog/transaction in question must be terminated, because it is no longer in a determinate state. If the SIP network does not route ACK responses properly, it is broken. And yet, again, many clients send no ACKs at all. Asterisk assumes they're not connected, and disconnects them. Even after the conversation is going nicely. ACK is required for INVITE requests (ONLY) that have route header fields. Otherwise, you rely somewhat heavily on loose routing of the ACK messages, which can result in any manner of fun loops dependent on the proxies in the mix and what sort of routes they may be tacking on. ACK was intended as a reliable method of determining whether or not a conversation has been well and truly established. But in reality, it is one of the less reliable methods. Add in PRACK to the mix (which increments the CSeq), and you can even get some fun race conditions which cause a search for matching ACK/CSeq pairs to fail on some servers (I'm not saying Asterisk does this, I'm just saying it does happen -- I've no idea how Asterisk handles PRACKs). Those servers may not proxy the ACK back to the final destination, assuming that the ACK is no longer an hop by hop but and end to end ACK dependent on how the CSeq matches up. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls hang up after 20 seconds
Alex Balashov wrote: SIP wrote: What is your citation for this qualification? RFC 3261 does not seem to me to say that, as in 13.1: Because of the protracted amount of time it can take to receive final responses to INVITE, the reliability mechanisms for INVITE transactions differ from those of other requests (like OPTIONS). Once it receives a final response, the UAC needs to send an ACK for every final response it receives. Or 13.2.2.4 (2xx Responses): The UAC core MUST generate an ACK request for each 2xx received from the transaction layer. I think, perhaps, I am misremembering the difference between sec. 17 ACKs and sec. 13 ACKs. I'm pretty sure one has somewhat less stringent requirements when sending an ACK from a TU (which a client would be, and Asterisk is, since it's not a transactionless server). I vaguely remember the language being softer in its requirements (which is where the initial confusion on the Implementors list arose when we were interpreting it). N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls hang up after 20 seconds
Gianni Fioretta wrote: Hello. I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason. The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and one of them answer, the call ends itself after 20 seconds from the answer. I've tried many configuration in sip.conf, but no one solved the problem. Log from /var/log/asterisk/messages: [Oct 8 15:49:05] WARNING[10659] chan_sip.c: Hanging up call e3f0204b-b35811de-bba58889-6d53c...@83.211.2.220 - no reply to our critical packet. and from CLI: [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 59e62874-b35911de-9a598915-c5a6b...@195.62.226.16 for seqno 101 (Critical Response) [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1972 retrans_pkt: Hanging up call 59e62874-b35911de-9a598915-c5a6b...@195.62.226.16 - no reply to our critical packet. == Spawn extension (incoming, 03411885583, 4) exited non-zero on 'SIP/03411885583-081e0d78' (peer 101 was not connected at this time, but Asterisk also hags up with all the peers connected) Any idea? Thanks in advance. This is a very weird Asteriskism that we see from time to time. Some SIP servers don't route ACK packets properly (or there will be an ACK loop). The nature of ACK packets is tenuous at best in the SIP world. Many clients don't even send them. Asterisk relies heavily on ACK packets to determine if a call is currently connected. If it doesn't receive one, it hangs up the call, even if the rest of the packets have been routed properly and the call is working fine. There's no configuration to turn this off, but there is a way to remove the check in the code. I can't recall the appropriate line to comment out, though. Perhaps someone else knows? In an ideal world, when Asterisk sent an ACK, whatever server/client it was connected to would respond accordingly. It is, however, not an ideal world, so this doesn't always happen. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
David Wathen wrote: Hi, My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. Thanks, David Wathen Depends on what level of firewall you're looking for. For a full firewall on either a dedicated system or one of your own, I cannot strongly enough recommend Astaro Linux firewall. Better throughput than a pix, worlds easier to operate and configure, and comparable in price. Very SIP/VoIP friendly. Loads of optional modules (we use its mail filter module to filter spam/viruses for several hundred thousand user mailboxes, for instance) to limit the cost to what you need. Also has a built in SIP Proxy, although I've never used it. Excellent platform. Of course, at home, I just use a little Linksys WRT box. It's hardly a corporate-grade firewall, but it's quite SIP-friendly. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VUC: RE: Friday 11th: Aswath Rao: Trapezoidal VoIP is Evil on VoIP Users Conference at Noon EDT
Kristian Kielhofner wrote: On Thu, Sep 10, 2009 at 2:24 PM, Dean Collins d...@cognation.net wrote: It might just be me but what is trapezoidal voip? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). I don't know if I'll be able to make the call but my guess is he's referring to the SIP trapezoid: http://www.iptel.org/sip_trapezoid The SIP trapezoid is a concept/teaching tool usually used in situations involving a proxy (or multiple proxies) illustrating the SIP concept of distinct separation between signaling and media. Granted this isn't unique to SIP... ISDN, SS7, H.323, MGCP, and I'm sure others also make this distinction. IAX is the only protocol I know of that doesn't (which is where most of it's NAT advantages come from). He's probably going to talk about the advantages and disadvantages of the trapezoid although from the title I'm guessing he's going to focus on the disadvantages ;). Then again I could be completely wrong. The SIP trapezoid is real but this speculation is purely my own. See... I would say the 'trapezoid' is one of the great strengths of SIP. Forcing RTP along the same path as SIP means you can't rely on all those incredibly powerful advantages that routers have in pushing packets along the best routes they have. You have to rely on the SIP proxy to do routing better than a router... except with additional mandatory hops in between to hit remote proxies. Just a bad idea overall. Let the proxies authenticate, since that's what they're best at. And let the routers route, since that's what they're best at. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?
An Asterisk MeetMe conference sounds like the ideal sort of scenario for you, allowing people to join in or drop off during a session as they please. N. li...@mgreg.com wrote: Hi All, As is obvious by my joining the list, I'm interested in learning more about Asterisk. I have downloaded the PDF manual (for version 1.4) and am beginning to go through it. What I'm looking for in the short-term, however, is a more concise reference for common Asterisk configurations and setups. I currently have a non-profit client to which I am donating work. They are looking to allow callers to listen in to public speaking sessions. They currently have a single phone line with call waiting and are using an archaic one-person switch to then allow folks to call-chain via 3-way calling. What they want is basically a switchboard that allows multiple people (5 to 10) to call in at a time of their choosing and begin listening to the in-progress session. My first question would be: Is Asterisk the proper tool for this job (or is there something else you'd recommend)? A follow-up question would be: What kind of cost is involved in a small setup of this nature? Your input is much appreciated. Best, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to deal with PayPal frauds?
When you start taking credit card payments (assuming you will), be careful with small payment amounts. You'll become a fraud haven. A lot of CC thieves or people who've just bought a CC number will use a small amount charge to check and see if the card is any good. Check out some of the MaxMind stuff for fraud prevention. They will do a lot of the IP geolocation checks and such for you for an exceptionally SMALL fee per transaction (fraction of a cent). It is absolutely worth it. N. Zeeshan Zakaria wrote: Thanks Matt, and everybody else, very useful information. I guess I'll have to sit again and spend time coding delays, small amount payments for new accounts and paypal=signup email match. -- Zeeshan A Zakaria On Mon, Aug 31, 2009 at 12:07 AM, Matt Riddell li...@venturevoip.com mailto:li...@venturevoip.com wrote: On 31/08/09 2:45 PM, Zeeshan Zakaria wrote: Those who are more experienced in this business, please advise how to avoid this type of fraud, and which service to use in place of PayPal, because PayPal doesn't seem the right payment solution for a prepaid VoIP service. Also now that they have all the payments put on hold and asking for a resolution, their resolution center is good only for shipped merchendise, not for online services. How would I prove to them that the buyer who is asking his money back has already utilized my service by making lot of international calls, which I now have to pay for to the carrier. I've used CDR for that and don't automatically accept payments. When we receive a payment we compare: 1. IP Address of user (whois normally gives approximate location) 2. Paypal account holder email (should match sign up email) 3. Countries for emails and ip address should match. 4. Initial payment should be $1-$2 (i.e. noone is going to sign up for a service and in order to test it put down $500 via paypal) If any of the above look suspect I ask the paypal account holder to email me and start looking at email headers to see how sus it looks. If it's a large amount then they have to have already been doing business with us successfully with small amounts - most scammers can't be bothered doing this. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPKall and FWD
A quick look at the system shows you're not logged in, which is why you're getting that message. N. David @ULC wrote: Oh my god.. Today its saying there is NOONE to take your call.I am using IdeaSIP What could be the reasons ? It was working perfectly till saturday . On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: IdeaSIP worked perfect for me. On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPKall and FWD
Means your username is not registered on the IdeaSIP system (your client/phone is not logged into IdeaSIP). N. David @ULC wrote: you're not logged in means ? On Mon, Aug 24, 2009 at 11:39 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: Oh my god.. Today its saying there is NOONE to take your call.I am using IdeaSIP What could be the reasons ? It was working perfectly till saturday . On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: IdeaSIP worked perfect for me. On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPKall and FWD
IdeaSIP, GizmoProject, IPTel, maybe OnSIP (don't quote me on that one, I'm not sure, but someone around has surely used it), etc, etc. There are a lot of alternatives about. Disclaimer: IdeaSIP is my personal unruly child (hence top billing on the list of alternatives). N. David @ULC wrote: We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Accessing to ekiga.net through Asterisk
Daniel, I'm a little confused as to what I'm seeing here. You're bounding through two RFC1918 address networks -- 10.1.0.X and 192.168.2.X. Is this some sort of dual NAT scenario? Perhaps if you can explain a little more about your network setup. N. Daniel Bareiro wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 SIP wrote: Daniel, Hi SIP. Check your stunaddr setting. Is it misspelled, or do they really use stun.exiga.net instead of stun.ekiga.net ? Thanks to indicate that error to me. I doing the test again. I don't believe that this solves what I commented before about 192.168.1.2 direction, but, just in case, I copy the output of debugging when trying to communicate to ekiga.net. The problem continues persisting after the correction. alderamin*CLI --- SIP read from 10.1.0.65:5060 --- INVITE sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 INVITE Contact: sip:2...@10.1.0.65 Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 933572867 1938524932 IN IP4 10.1.0.65 s=- c=IN IP4 10.1.0.65 t=0 0 m=audio 8000 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 - - --- (13 headers 12 lines) --- Sending to 10.1.0.65 : 5060 (NAT) Using INVITE request as basis request - kafgeaflkmsd...@defiant.freesoftware.org --- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKrqslryke;received=10.1.0.65;rport=5060 From: Hector sip:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10;tag=as0a3a462b Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=497d879d Content-Length: 0 Scheduling destruction of SIP dialog 'kafgeaflkmsd...@defiant.freesoftware.org' in 32000 ms (Method: INVITE) Found user '201' alderamin*CLI --- SIP read from 10.1.0.65:5060 --- ACK sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10;tag=as0a3a462b From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 ACK User-Agent: Twinkle/1.2 Content-Length: 0 - - --- (9 headers 0 lines) --- alderamin*CLI --- SIP read from 10.1.0.65:5060 --- INVITE sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKxpraybjr Max-Forwards: 70 Proxy-Authorization: Digest username=201,realm=asterisk,nonce=497d879d,uri=sip:8...@10.1.0.10,response=9cb53107d4d15b7a2e7df8599e851b80,algorithm=MD5 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE Contact: sip:2...@10.1.0.65 Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 933572867 1938524932 IN IP4 10.1.0.65 s=- c=IN IP4 10.1.0.65 t=0 0 m=audio 8000 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 - - --- (14 headers 12 lines) --- Sending to 10.1.0.65 : 5060 (NAT) Using INVITE request as basis request - kafgeaflkmsd...@defiant.freesoftware.org Found user '201' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.65:8000 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe (gsm|ulaw| alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.1.0.65:8000 Looking for 8500 in from-internal (domain 10.1.0.10) list_route: hop: sip:2...@10.1.0.65 --- Transmitting (no NAT) to 10.1.0.65:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060 From: Hector sip:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE User-Agent: Asterisk PBX Allow
Re: [asterisk-users] Platform decision ...
Steve Totaro wrote: On Tue, Aug 18, 2009 at 1:52 PM, Mauro Sergio Ferreira Brasil mauro.bra...@tqi.com.br mailto:mauro.bra...@tqi.com.br wrote: Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ;): The main points listed on Asterisk's CONS that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT traversal; * Asterisk doesn't use SpanDSP; * Use of no longer maintained Berkeley DB1 engine as its internal database; * Asterisk doesn't allow CSRC entries in RTP; * Asterisk doesn't have an universal jitterbuffer for use with any channel type; * Asterisk doesn't use POSIX realtime extensions (having dependency with Zaptel timing); We were considering Asterisk as the chosen platform, but after reading this I got a little worried. The comparison considers 1.4 old version of Asterisk. So, can someone give me an update on what have changed for this items considering new 1.6 version ? Maybe someone can point me a site with an updated comparison. As long as I could see by now SpanDSP is present on new version of Asterisk, so this item isn't a difference any more. Right ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:mauro.bra...@tqi.com.br mailto:@tqi.com.br http://tqi.com.br : www.tqi.com.br http://www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 Don't forget to add FreeSwitch to your comparison chart too. Then you'd have to add the con: cryptic, difficult to find, and wholly incomplete documentation. Don't get me wrong. FreeSwitch is a very nice back-end product. But as far as ease of putting it into deployment goes, it's a nightmare from its complete dearth of anything related to coherent docs. It still feels very nuts and bolts. Like being handed a Porsche Boxter engine, frame, and a wrench and being told to sort of 'figure out' how it all goes together. And even when you do, it will function screamingly well. But it won't have doors, windows, AC, or creature comforts that we've all come to expect. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Platform decision ...
Moises Silva wrote: Hi there, I though to chime in here just to share my opinion for what is worth. As a developer who enjoys playing with telephony in general I try to remain as objective as possible when talking about one or the other, and I felt that N from arcdiv was a bit unfair with FreeSWITCH docs. Then you'd have to add the con: cryptic, difficult to find, and wholly incomplete documentation. Don't get me wrong. FreeSwitch is a very nice back-end product. It's hard to not get you wrong when, in my opinion, you start by writing as facts what is barely your particular poor user experience with it. Others, including me, have found what they need in FreeSWITCH wiki just as I have found what I need about Asterisk docs in voip-info. Sorry, Moises, but I've been to the FreeSWITCH Wiki. It's sparse at best. You may have found what you were looking for based on your possibly very simple needs, but that doesn't make it very complete documentation. far as ease of putting it into deployment goes, it's a nightmare from its complete dearth of anything related to coherent docs. It still feels very nuts and bolts. Like being handed a Porsche Boxter engine, frame, and a wrench and being told to sort of 'figure out' how it all goes together. And even when you do, it will function screamingly well. But it won't have doors, windows, AC, or creature comforts that we've all come to expect. You mean comforts which you have come to expect. Again, my needs have been so far fulfilled for conferencing and SIP/PSTN gateway uses. Pointing to particular missing applications instead of making your own analogy would be useful, otherwise you are not really being of much help, and just introducing FUD. I'm not introducing FUD by stating my opinion about the lack of documentation, Moises. You're sounding incredibly defensive. Why? Many users are confused because they try to do things the same way they are used to with Asterisk and some concepts just don't fit or are differently applied. From what I've seen the users that get annoyed the most are those who keep trying to do things in the Asterisk-way and get overwhelmed by the configuration differences, instead of learning the FreeSWITCH-way to accomplish the same goals. Users just get impatient because they're already familiar with something and this new engine is not managed as the old one. The recent announcement of FreePBX running over FreeSWITCH (http://www.freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future) should help to close the gap in user configuration and ease of management. Of course, there is some truth in your statements. FreeSWITCH needs to catch up with documentation, but I would defy anyone to say they've come to hang around on IRC and did not get their question answered. IRC answers from people hanging around is not documentation. It's what open-source developers like to think of, sometimes, as a 'complete' solution, but it doesn't even come close. My comment was about documentation. Which is lacking. This is not FUD. This is not me saying Don't touch FreeSWITCH. This is me saying that, if you're looking for a product with good docs to make an easy transition from traditional PBX tech to new, or even an easy transition from Asterisk to something else, you will not find them with FS. I'm sure that's changing as time goes on, but it's not there yet, and the focus doesn't seem to be on ensuring it gets there. The focus seems to be on the band-aid of asking questions on fora and IRC to try and get an answer. That may work for some things, but for overall deployment, it's lacking. I'm sorry you're offended by my opinions, but in your words, 'I defy you' to show me some comprehensive FreeSWITCH docs. Heck, even SER has more comprehensive documentation, and that's saying a LOT. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Accessing to ekiga.net through Asterisk
Daniel,
Check your stunaddr setting. Is it misspelled, or do they really use
stun.exiga.net instead of stun.ekiga.net ?
N.
Daniel Bareiro wrote:
Hi all!
I'm trying to connect to ekiga.net through a client connected to my
Asterisk server. For it I am being based on this [1] document. Next I
put the configurations that I am using.
/etc/asterisk/sip.conf:
; Outgoing to ekiga.net
[ekiga]
type=friend
username=MyUser
secret=MyPass
host=ekiga.net
canreinvite=no
qualify=300
nat = yes
stunaddr = stun.exiga.net
insecure=port,invite ; required for incoming ekiga.net calls
/etc/asterisk/extensions.conf:
[from-internal]
...
exten = _8.,1,Dial(SIP/ekiga/${EXTEN:1},20,r))
I tried a echo test, dialing in my case to 8500, but in spite of seeing
traffic towards Internet, nothing is heard. Setting 'sip set debug', I get
the following thing:
--- SIP read from 10.1.0.65:5060 ---
INVITE sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: sip:8...@10.1.0.10
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 183 INVITE
Contact: sip:2...@10.1.0.65
Content-Type: application/sdp
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.2
Content-Length: 247
v=0
o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
s=-
c=IN IP4 10.1.0.65
t=0 0
m=audio 8000 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
-
--- (13 headers 12 lines) ---
Sending to 10.1.0.65 : 5060 (NAT)
Using INVITE request as basis request -
mrsyiysrdkwm...@defiant.freesoftware.org
--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKjxcxrrks;received=10.1.0.65;rport=5060
From: Hector sip:2...@10.1.0.10;tag=uucwz
To: sip:8...@10.1.0.10;tag=as095989a3
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 183 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=76b2dfe8
Content-Length: 0
Scheduling destruction of SIP dialog
'mrsyiysrdkwm...@defiant.freesoftware.org' in 32000 ms (Method: INVITE)
Found user '201'
alderamin*CLI
--- SIP read from 10.1.0.65:5060 ---
ACK sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: sip:8...@10.1.0.10;tag=as095989a3
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 183 ACK
User-Agent: Twinkle/1.2
Content-Length: 0
-
--- (9 headers 0 lines) ---
alderamin*CLI
--- SIP read from 10.1.0.65:5060 ---
INVITE sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKoilauqhp
Max-Forwards: 70
Proxy-Authorization: Digest
username=201,realm=asterisk,nonce=76b2dfe8,uri=sip:8...@10.1.0.10,response=d49c0fdf11c9977fcd1fce6a50f445fe,algorithm=MD5
To: sip:8...@10.1.0.10
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 184 INVITE
Contact: sip:2...@10.1.0.65
Content-Type: application/sdp
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.2
Content-Length: 247
v=0
o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
s=-
c=IN IP4 10.1.0.65
t=0 0
m=audio 8000 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
-
--- (14 headers 12 lines) ---
Sending to 10.1.0.65 : 5060 (NAT)
Using INVITE request as basis request -
mrsyiysrdkwm...@defiant.freesoftware.org
Found user '201'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 10.1.0.65:8000
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.0.65:8000
Looking for 8500 in from-internal (domain 10.1.0.10)
list_route: hop: sip:2...@10.1.0.65
--- Transmitting (no NAT) to 10.1.0.65:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
From: Hector sip:2...@10.1.0.10;tag=uucwz
To: sip:8...@10.1.0.10
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq
Re: [asterisk-users] Time of Day Routing
Tony Mountifield wrote: In article 05d03313-994b-4892-b045-f61332ddb...@geekinter.net, Steve Howes st...@geekinter.net wrote: On 14 Aug 2009, at 09:17, Neeraj Chand wrote: Asterisk version 1.4 From: Neeraj Chand Sent: Friday, 14 August 2009 8:17 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] Time of Day Routing Hi David, With this: ifTime(00:00-12:00|*|*|*) Whatever time you specify at the end, I believe asterisk continues to evaluate this condition as true for 2 more minutes. So in this case, it will be valid for 00:00-12:02, even though you’ve specified 12:00 Cheers! Neeraj Post a few hours ago.. Actually, that's 12:02, because times before 1.6.2 are only accurate down to the 2-minute interval. So 12:01 is treated the same as 12:00. Starting with 1.6.2, times are accurate down to the minute. Hmm, I would still consider it a bug, whether on 1 or 2 minute resolution. The example condition should start being true at 00:00 exactly, and stop being true at 12:00 exactly. So at 12:00:01 it should NOT match: if (now = start_time now end_time) This then is independent of the resolution, provided the end time is an exact multiple of that resolution. After all, if a shop shuts at 5pm prompt, and you get there at 10 seconds after 5pm, it is shut, not open until 5:00:59.99 or whenever. Cheers Tony We're talking precision here, though. With a 2-minute precision, you have to understand that there IS no 12:00:01 as far as Asterisk is concerned. There is simply 12:00 and 12:02. At exactly 12:00, it evaluates true, just as has been put in the if statement. It checks again at 12:02 and it evaluates false. That's not a bug. That's just a lack of precision in checking. It can't check ALL the time without devoting cycles to checking, which takes cycles away from other things. Think cron on a unix system. Nothing happens in 30-second increments. Things happen in 1 minute increments at the smallest because that's the maximum precision that's built into the program. You could WRITE a cron that checks every 5 seconds, but it's not a bug in cron that it only checks every 1 minute. That's simply the way it works. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
I'm not sure there IS an issue, per se. There are lower bitrate codecs that will work fine for voice communications in both directions. But if you're trying to force a low-end codec to the upstream, that just means the downstream on the remote end is going to be stuck with a low-end codec. And if he's trying to force a low-end codec on his upstream, then you're still going to get a low-end codec on your downstream. If you're relying on the Asterisk box somewhere in the middle to transcode these two streams into a higher bitrate codec, you're still not going to get any higher quality than you send. Your option is using a low bitrate codec for both directions and not bothering to try and min/max your streams. Excellent quality low bitrate codecs exist in the form of g729, iLBC, and Speex (in that order of prevalence). Just deal with the fact that, even if you min/maxed the streams, you still wouldn't be able to get any more streams than will fit in your upstream pipe, so let that be your guide for the technology involved. You may end up with some extra bandwidth on the downstream side, but trying to fill it up with something just to fill it up with something won't get you anywhere. N. Elliot Murdock wrote: Hello Everyone! Thank you for all the information. I am wondering how the Asterisk community has been working on solutions to deal with the asymmetric quality of ADSL. Voip is becoming popular and a bottleneck does exists on the ADSL upload side. Elliot On Sun, Aug 2, 2009 at 3:17 PM, Kevin P. Flemingkpflem...@digium.com wrote: Tim Panton wrote: The protocol expects the 2 ends to agree a single symmetrical codec as part of the connection setup, but it doesn't define what actually happens if the codec specified in the first (full frame) voice packet isn't what was agreed. Asterisk only supports symmetric codec configuration on its internal channels, so in Asterisk's IAX2 implementation, if a frame is received from the other endpoint that is not in the 'expected' format a warning is issued and the outbound direction is automatically switched to the same format. The same is done for any protocol using RTP in Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on OpenWRT
I've had similar results to you. Packet loss even when not transcoding. Overall poor performance across the board. We considered it a failed experiment. N. Zoa wrote: I have played with DD-WRT on linksys wrt54g version 5 last week (2 different ones, they are the model with less memory so i needed to use the micro version). I tried to use it as a repeater. (might have something to do with it) So far i read reports on great succes everywhere, my experience are not as good, the machines become highly unstable and i experienced heavy packetloss at random times. Encryption didn't work at all. Maybe other versions (with more memory or faster CPU's) are better, but my results were a disaster and i would not consider running Asterisk on top of that. Joachim David Cook wrote: On Mon, 27 Jul 2009, Jeff LaCoursiere wrote: 1) The latest 8.09 kamikaze no longer supports the Broadcom radios, so ... Because of closed-source drivers the Broadcom chips only work on the 2.4 series kernels. OpenWRT does make a 2.4 kernel version _and_ a 2.6 kernel version. Use the 2.4 and the radios work fine. 2) I suppose this should have been clear to me from the start, but without an external (or hacked internal) storage of some kind, running asterisk on Make sure you have the right version number within the Linksys model. They changed drastically the RAM/Flash in the units (downward) as the production ran on. There are some charts online to go by. But the skinny is use a WRT54GS v4 or lower. V1.1 2 were the good ones with double the RAM. 3) OpenWRT seems to be less stable and not as mature as dd-wrt, which I I guess this is someone subjective and OpenWRT is somewhat in flux with 2 products under the same brand right now. White Russian was the previous release which is still available. Used predominantly NVRAM configs and had a smaller audience of platforms that it would support. It did however have a great GUI with lots of features. Kamikaze is the new version which has moved to more traditional config files and has an objective to be more platform agnostic. As a long-time White Russian user I admit the GUI has a long way to go before it can be considered a replacement for the White Russian version. I myself have never encountered stability problems with either version. Not sure how much DD-WRT has improved. A few years back OpenWRT was the clear winner (in my mind - no flames please) and I haven't re-evaluated the competition lately. -dbc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best practices for running asterisk as SIP B2BUA
Alex Balashov wrote: BTW, if you need a generic, scalable, high-volume B2BUA, it is not a best practice to use Asterisk for that purpose. Indeed. But you can grow some good SMB B2BUA systems out of it. Freeswitch would be a grand alternative... if it had documentation. Anywhere. Ever. That DIDN'T involve scouring the source files (which is the slack OS coder's idea of 'good' documentation). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: confirm f1ab6c493110edited]
Dunc wrote: Doug Lytle wrote: Your membership in the mailing list asterisk-users has been disabled due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug I just did yes, Don't know why :) Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It's because two spam emails advertising canadian pharmaceuticals came in in rapid succession. Any rational filter would have bounced them to the sender (the sender being the Asterisk List Server), and will therefore be subject to the 'too many bounces' rules on the list. Which, I assume, means more than 1. Or possibly only 1. Honestly, those rules are a little undocumented. I get those messages often and find myself re-enabling my list membership on a regular basis due to spam that hits the list, but that my server likes to refuse. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and FW settings
Michael wrote: On Tue, 14 Apr 2009 20:47:29 you wrote: Hi michael, you should open both tcp,udp 5060,5061 too and as you mentioned between 1-2. AFAIK 5061 TCP is for TLS SIP which isn't used much yet? Is TCP the default for 5060, with UDP as fallback, or is this provider dependent? Michael That's provider dependent. MOST SIP is done with UDP. Some people use TCP to get past firewalls or try and alleviate NAT issues, but it's non-standard and falls into the category of 'complete hack' where SIP is concerned. TCP is allowed via the RFC, I believe (I vaguely remember a transport=tcp setting somewhere in a header field), but whether or not it's supported by the provider or software varies widely. Microsoft uses only TCP in their communicator product, I think. Some clients will let you choose TCP or UDP. But for the most part, when dealing with a default asterisk install and your own phones/softphones, you shouldn't need to worry about TCP. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Security
If that someone is between you and the other endpoint (like between you and the switch, or using port-mirroring on a router somewhere), then yes. The conversations can be recorded. In the US, the ability to be able to do this is required by law. You've little to worry about random hackers coming in off the Internet for this sort of thing. It's usually something to do with having physical access to the network in which you or the other end is connected. There's ARP poisoning and the like which could make this possible in a local network environment on either side, but for the most part, you'll know who's on your local net, and they likely have physical access to your phones as well. A listening device would be easier to plant in the mic pickup of your phone if they REALLY wanted to listen in on your calls. There are all sorts of levels one can to to find out what you're doing, and preventing against them can involve a great deal of creativity. That said, the answer is yes. You could use a VPN tunnel from one end to the other, and many people do just that to help ensure the privacy of their connections (both data and voice). N. Tom wrote: Since we are talking about security, if I am using * to talk to a cisco gateway via SIP, is there some sort of encryption you can use? Like a vpn tunnel? Can someone capture packets and re-assemble to make out a conversation? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: Saturday, April 04, 2009 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Security Lets not be that paranoid. If you have these ports open to the internet then from time to time someone will check if your default unsecured context can dial out to PSTN... with sip.conf you can add allowguest=no With IAX2 there's no allowguest but I believe you have to have a guest username in iax.conf with no password to access the unsecured context. Martin On Sat, Apr 4, 2009 at 3:42 PM, Todd Reese trees...@gmail.com wrote: Hi All, Coming in to day, the logs on the asterisk server showed several entries such as: [Apr 4 15:25:16] NOTICE[9280]: chan_sip.c:14627 handle_request_invite: Call from '' to extension '9810380487965419' rejected because extension not found. This has gotten me to thinking about security of this box. 1. Currently the box sits behind a firewall with iax and sip ports pointing to it for the ip phones that are offsite. There isn't any other access through the firewall to this box. 2. All devices have an extension assigned to them in sip.conf and extensions.conf. i.e. supra ATA, Grandstream GXP-2000 3. The box is fed via Les.net and Voicepluse. All other feeds are shutoff when not active. I'm looking for ideas to tighten up on the security so that this won't happen again. TIA, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.238 / Virus Database: 270.11.41/2040 - Release Date: 04/04/09 16:53:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. N. Dean Collins wrote: Does IPKALL still exist? I am after a free SIP trunk – who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
Daniel Nowacki wrote: SIP wrote: IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. I get an ugly 404 when trying to sign up or log in... That is probably abandonware... :( No. It's just poorly-checked web management. http://phone.ipkall.com/ Is the signup link. The /ipphone stuff appears to be an old document tree that no longer exists. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?
randulo wrote: This brings up a side issue. Banks on the Internet have had to provide a sort of insurance that allows the customer to be protected if someone hacks in to his or her account. ITSP will need to think carefully about having a similar policy that protects people from an attack to the provider, no? What do those of you who sell these services thing about liability? Has anyone come up with a statement on this? /r The customer IS protected because it's excellent marketing for the bank or credit card provider. If someone steals my card number and racks up a bunch of charges, I'm often not liable for those charges (dependent, of course, on bank policy). However, the seller who was duped into selling those items because the bank approved the charges on the card? They're simply out of luck. They're charged any relevant charge-back fees AND are out any fees for services or product losses they may have incurred. The bank still gets its money. In the end, SOMEone has to pay. As an end-point ITSP, I can assure you, it would be us who's assessed the requisite charges. If someone uses a fraudulent card, we're required to pay. If someone uses a three letter password on his account, and it's hacked into and uses to rack up charges, we have to pay. In the purely virtual sense, as we're often selling to people we've never met via the Internet, it becomes difficult to say with any certainty if the person who logged into the account and used up the account's money is a hacker or just the account holder who doesn't want to own up to the charges. It puts us in a difficult position. Obviously, in some cases, this becomes more obvious. If the account holder is in the UK and the calls come in from China or Nigeria or Turkey or some such, it would be more likely to be suspect and if the account holder challenged the charges, we might be more liable to work with him or her. However, for the most part, we require a certain 'strength' of password to be used, and we rely on safeguards and monitors on the site itself to try and avoid brute force hacks. With no evidence for a brute force attempt or some other security failure on our side, we're somewhat at the mercy of logic to assume that calls from a customer's premises using a customer's account actually came from the customer, and I think we might be hard pressed to simply ignore said charges. If the security failure is clearly ours, though, I don't think it would be at all reasonable to expect the customer to accept responsibility. I'd be especially wary of a company that blamed the customer for its own security failings. -- Neil Fusillo CEO Infinideas, inc. http://www.ideasip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?
randulo wrote: On Thu, Mar 26, 2009 at 1:32 PM, SIP s...@arcdiv.com wrote: As an end-point ITSP, I can assure you, it would be us who's assessed the requisite charges. If someone uses a fraudulent card, we're required to pay. If someone uses a three letter password on his account, and it's hacked into and uses to rack up charges, we have to pay. Neil, It hadn't occurred to me when writing it, but obviously there are situations that don't match the banking paradigm. For example, suppose I run my own asterisk, I have a contract with a company like yours and you have my banking info with an authorization to top up. If the fraud is someone on the banking end (hacked my card details for example) that's covered by the bank. But if they brute force hacked my asterisk install because the extension, the username and the secret are all '2005' and then make $100k worth of calls, people like lawyers and judges won't easily see that it's the asterisk install that's responsible, not your company or even the bank. I wonder what steps can be taken legally right now to make responsibilities clearer to the legal world? I once had a guy break in to my house and call his girlfriend in Mexico about 50 times in two weeks. When I called Pacific Bell, the operator placed a call to the number, the woman (stupidly!) admitted, yes I know Luis, he calls me all the time and even though the operator heard this, PB still refused to exempt those charges and go after the guy. I closed my PB account and opened a new one under a variation of my name. /r Indeed, the old method of this sort of fraud involved a lineman's handset or a phone modified with alligator clips to attach to the NID outside the home of someone not in town, thereby being able to call long distance on someone else's bill. I've heard of NO cases in which the phone company accepted liability for those charges, even if they forgot to lock the NID itself. For all intents and purposes, it's a telco-installed back door into your system with poor overall security. The problem with getting the legal system to understand whose responsibility this is is a difficult one. Politics and an overall lack of good, unbiased information has always affected legislation and, as such, jurisprudence. Politicians neither know nor tend to care about the finer points of technology and how it may be used. They rely on advisors to tell them the bullet-point version of any issue before they make a snap decision on whether it's expedient to back it legislatively. These advisors are either lobbyists, PACs, or advised by such, and all of them have an agenda. I can assure you that the agenda of the home or home business with Asterisk is not heard. Ever. This leaves a judge to make a decision should it come to court, and it could go either way, but it would be a messy and expensive battle, and the decision of the judge would be tempered by what's written into the law, which right now is hardly kept up to date for modern technologies. In a situation like ours, we'd be dealing with legal systems in a variety of countries, which would make things even more complex. I think step one in this sort of fight is, and has always been, having a true political voice that can be heard above the din of established special-interest groups. The VON Coalition was an idea like this, but it's an incredibly exclusive membership -- designed for companies making hundreds of millions if not billions a year in revenue. With minimum annual dues of $10,000 or more, it's quite reasonable as a semi-democratic organisation for business making $500,000,000 a year. For smaller companies, it's laughable. And so, the voices heard are the ones which were heard before -- the ATTs, the British Telecoms, the Comcasts, and the Verizons of the world. It becomes just another avenue to get the same political point across. A second opinion that's guaranteed to be the same as the first, as it were. And so, in answer to your question, I don't think there ARE necessarily steps that can be taken right now to ensure that there's a rational approach to the resolution of such an issue of fraud. Barring some sort of major legal precedent, it's going to be anyone's guess how the verdict comes out in the end. -- Neil Fusillo CEO Infinideas, inc. http://www.ideasip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?
randulo wrote: On Thu, Mar 26, 2009 at 2:38 PM, SIP s...@arcdiv.com wrote: And so, in answer to your question, I don't think there ARE necessarily steps that can be taken right now to ensure that there's a rational approach to the resolution of such an issue of fraud. Barring some sort of major legal precedent, it's going to be anyone's guess how the verdict comes out in the end. Hence the need for all of us, everywhere to step up measures to prevent as much as possible, the unlawful use of a system. Maybe some kind of (optional modular) monitor or engine could be built for the asterisk platform to at least send alerts when it deduces suspicious activity? r There are generally two approaches to this. Neither is necessarily 'correct,' but one is considerably less unwise. The first approach is the current approach: build software with little thought to how it will be secured, opting for all the work of securing the product once it's been implemented to come down to a requirement for the deployer to both know and, more importantly, understand good security practices. This has a value for enthusiasts because many of them will be running the service just in a home network or test environment, and it lets them get things up and running without worrying about all the little issues that might get in the way of a quickly-deployed system. It's essentially like choosing 'install everything' on a linux install and opting to have no firewall. It's wonderfully easy to deploy and there are no weird rules getting in the way of using the system immediately. It's also a really REALLY (I can't stress how strongly enough) bad idea if you're building a product that is deployed by more than just enthusiasts and will ever be in any remote way tied to someone's finances (including, but not limited to, telephone access charges, bandwidth fees, etc). The second approach is to build the product to be as secure as it can possibly be right out of the box, and require those deploying it to essentially remove levels of security in order to get things working in a particular environment. This also requires a certain knowledge of security practices, and it relies on those deploying the product to understand that the errors they may be seeing on deployment are likely to do with security feature X or Y. This takes time and a lot of work, because every component of the system has to be hardened and tested to ensure a seamless security model throughout without worries about incompatibilities in the basic security model between modules of a complex system. It also makes the system harder to deploy out of the box because it requires tailoring for the specific environment not just to handle a different user base, but also simply to work. I think there's a lot of push back on this sort of model for something like Asterisk because people feel that security should be this nebulous thing that exists 'somewhere else.' But in reality, security starts with the software itself and works outward. Just as you can't build a stable house on an unstable foundation, any weak link in the security chain is an invitation to disrupt the entire system with an exploit. And the weak link in MANY systems when it comes to security is the knowledge of the person deploying it. I believe a certain level of high grade security should certainly be built into Asterisk, and that it should have an overall security model, as well as documentation discussing the security of the system and the parameters that accompany it. Not only would this alleviate the concerns of many people deploying, but it would be excellent marketing. Have you seen the number of cars that advertise their side-impact air bags, safety rating, and other such features? Nothing will keep a person from killing himself in a car if he chooses not to wear a seatbelt and drive unsafely in heavy traffic. But if he's in a car without seatbelts? Or with a horrible crash test rating? Chances are he may end up getting hurt anyway. Even if he makes sure he drives carefully. -- Neil Fusillo CEO Infinideas, inc. http://www.ideasip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?
randulo wrote: On Thu, Mar 26, 2009 at 4:19 PM, SIP s...@arcdiv.com wrote: The first approach is the current approach: build software with little thought to how it will be secured, opting for all the work of securing What about SIP itself? Does it provide enough crypto to be solid? Or is that handled only by the layer above it? /r ___ SIP CAN be reasonably secure, but it suffers from some inherent issues in the protocol for which things like TLS and the like were developed. It's still comparatively new, and it's a draft that I think needs some work. But it also suffers from an increasing amount of competition from upstarts that are trying to muddy the field somewhat (IAX, Jingle, etc.) and position themselves as the 'new' and 'better' way to address communication. This detracts from a unified methodology -- even if only somewhat. SIP is, for all intents and purposes, as secure as vanilla SMTP email. In fact, SIP was designed to closely resemble a combination of SMTP and HTTP to make it easy to implement and process. However, like both SMTP and HTTP, I think what SIP needs is a solid roll out of a secure layer over and above the MD5 hashes commonly used to pass passwords -- but that isn't really necessary to secure the protocol from password-sniffing ne'er-do-wells who are out to steal your accounts. SIP was written in such a way that the hashes it sends for passwords could, with only a trivial rewrite of the server code, be SHA1 instead of MD5 -- which would increase security to the level that, currently, it would be far more trouble than it's worth to even bother to attempt to crack. For keeping people out of your paid accounts, this would make SIP quite secure. The only issue most people have with SIP at the moment is that, if you're sniffing the network, you can read the SIP messages themselves, even if you can't crack the passwords, so even with SRTP or some other form of RTP encryption to protect the voice, your basic privacy is still at risk. But to protect money? I think SIP is perfectly fine even without TLS. It just needs a change in commonly-used password hashing to alleviate the concerns people have with the breakability of MD5. -- Neil Fusillo CEO Infinideas, inc. http://www.ideasip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?
Dave Platt wrote: SIP was written in such a way that the hashes it sends for passwords could, with only a trivial rewrite of the server code, be SHA1 instead of MD5 -- which would increase security to the level that, currently, it would be far more trouble than it's worth to even bother to attempt to crack. I strongly doubt that the known weaknesses in the MD5 hash are the weak point in SIP account security. Weak passwords are almost certainly much more of a problem. Performing a dictionary attack is going to be a lot faster than attempting a brute-force mathematical attack against MD5... and switching from MD5 to SHA-1 provides no significant defense against dictionary attacks. The only good way to keep passwords secure against dictionary attacks, is to make sure that the passwords aren't guessable by that means... no common words, no names, no simple permutations or birthdates or anything like that. Use a decent random-number generator and number-to-character conversion algorithm to generate SIP passwords that are sufficiently long and very dtr8fbwf_==...@\.-+!n$ and you'll be well defended. I'm referring to the weak link in the SIP protocol. Not in Asterisk's SIP accounts. The question was whether or not SIP itself was secure. -- Neil Fusillo CEO Infinideas, inc. http://www.ideasip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
David Ruggles wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200da...@safedatausa.com I believe SNOM 300s do PoE (might have to check that, though) and are around $100. We've little experience with them, but we use an office full of Snom 320s, and we're nothing but pleased with them. Good speaker, good handset, lots of excellent options. And reasonably priced. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX based war dialer
Not to burst your bubble, Jon, as I agree with a majority of what you said... but using an argument about the evolution of email to support an argument about how telcos should have better tracking and accountability is somewhat weird. We get 3 million email messages a day through our servers. 98% of those emails are spam -- often difficult if not impossible to trace because of the numerous methods of hiding one's identity including but not limited to spoofing domains and IPs, and using compromised machines. I hope to god the telcos don't 'fix' the phone world the way email 'fixed' the communications world. N. Jon Pounder wrote: Tim Nelson wrote: The fact that this would be even being discussed on this list is an embarrassment to the asterisk community. I am constantly being pestered by cold callers with fake caller ids, probe calls such as this, etc. I think for once CRTC/FCC need to step up to the plate and take some useful measures : - make knowingly presenting forged caller id a federal crime (its fraud and harassment already) - block caller id spoofing at the telco boundaries (we all do this now for ip addresses, so why not caller id ?) - ban offenders from having telecommunications service of any sort nationally once convicted. If the telcos can't adapt to providing service and accountablity this way and actually serving the customers who pay them, telecommunications with just evolve without them. Much the way the post office is being left behind since they can not compete with the speed of fax and email for documents or couriers for packages. Another war dialer with IAX capabilities: http://www.softwink.com/iwar/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Steve Edwards asterisk@sedwards.com wrote: This may be of interest -- as a tool we can use to test our systems and as a weapon that may be used against us :) http://warvox.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building a phone
Tzafrir Cohen wrote: Hi folks A common wisdom here is that one should use a proper hardware phone rather that an extra software on the user's PC. Why is that such a big issue? Marketability for one. People worldwide understand the telephone paradigm. You have a handset and a box with numbers. You pick it up and dial, talk through the handset, and listen in the other end. It's simple. It's an elegant design. And everyone from 1 year olds to my 97 year old grandfather can use it. Software phones? Not so much. In fact, not even close. The additional complexity of running software on a machine ALONE would keep my grandfather and that 1 year old from using it. Headsets? Seriously? Since when have those been user-friendly OR comfortably. In essence, adherence to a software phone paradigm breaks a century of design advancement in telephone ergonomics, psychology, and reliance, and replaces it with something that's clearly just a kludgy add-on to a product which was never originally designed for the task. One thing that bothers me with the current crop of hardware SIP phones is that they are hopelessly properitary. So what would it take to build a fully-adaptable phone? Here are some of my thoughts. This is not anything I plan to do soon (if at all), but I really find it strange that there aren't such phones already. == Small Quantities: When you look at such systems it becomes aparant that you can get much nicer prices if you buy large quanities. But this is something that will be a problem. Not only for prototying. The fact that you're limited to a strict hardware setting is very limiting. No mixing and matching like in a standard PC. I'm not exactly sure how to overcome that. This is one of the biggest reasons all the hardware phones are proprietary -- they're each written for different basic hardware. == Platforms: There are many embedded platforms nowadays. I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. My target phone should be able to handle at least two concurrent Speex calls. Preferrebly 6 speex calls and above. OTOH, I can't afford a monster CoreDuo. I need a quiet system with no fan. Thus the target CPU may be higher end VIA or Atom. Not sure about Geode. There are also some interesting ARM-based boards around. I'm completely unfamiliar with them but I suspect that they may prove to be cheaper. == SIP Software: Not really sure here. There must be something close to usable already, I guess. == Micro Browser: Hell no! The device should have an LCD display, and the content of that display should be programmable. Programming it using a HTML renderred is a bad design decision. The device should be a good phone. It should not attempt to be a web browser, as it will be a lousy one. == Handset: I suppose that an obvious starting point for a handset is skype phones such as USB handsets from yealink. Far from an optimal design, but a driver already exists. == Ease of Use: A phone must be usable. The target device must be something my mom can use. However that does not mean it must be easy to program. It must be programmable and hackable. But I can live with a complicated user interface for that. If such phones become successful and useful, better interfaces will eventually be written. Just a note here -- a complicated user interface, though you personally may be able to live with it, will pretty much ensure that the phones never become successful enough for a better one to be written. UI design is about 10% code and 90% psychology (and so FEW people who call themselves UI 'programmers' understand that). Just having a UI that can get you from point A to point B without typing in commands is NOT a UI worth making, as it will never be a UI worth using. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
Michael wrote: This has absolutely nothing to do with the fact that something is opensource. The fact that the source is open has nothing todo with its pricetag. Sometimes opensource products are more expensive then closed source products. If you want support/maintenance/dedicated_features/you-name-it you'll have to pay for it. But you only pay for what you want/need, and not because some egghead decided what he wants to put together as a sales-package. Opensource is about the freedom to check and to change, security, quality. If you doubt it, check with SLES/RHES/ABE/... There even seems to be companies that do _only_ support on open products, like typo3, openoffice, And make a living out of it. Big companies, especially those with major computing systems use paid software because they want a vendor they can hold responsible for it. As for OSS and FOSS, it is majorly used by the sort of businesses and individuals who call me (and other IT pros) up and talk the talk, but they don't have a 2 dimes to rub together. This problem is only going to get worse as the so-called 'recession' bites... fellow I.T. professionals - get used to your clients trying to weasel free service out of you. Everything I am hearing from fellow I.T. people is that there is no shortage of 'work' but a lot of clients are resisting paying. ___ No... there's no shortage of work that needs doing. But there's a definite shortage of money to pay those to do it -- hence the massive, worldwide layoffs. Your little corner may not be affected, but to discount basic economics because you don't see it? Well... that's incredibly short-sighted and provincial. Expect a bigger push to FOSS simply because fewer companies can afford what they used to be able to afford. They can't get loans. The people who buy their services and wares have all but vanished, so they have no influx of capital. This is not some 'media-created' concept. There's some incredibly good OSS and FOSS out there (Asterisk is a case in point). People who sneer at companies that use it, saying they're somehow lesser than companies that don't are, I usually find, those who are making a living overcharging for their products. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed
Grygoriy Dobrovolskyy wrote: 2009/2/13 Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote: I've been involved with getting better data for running Asterisk on the Amazon EC2 cloud computing system. Here are some calculations I've made on costs based on current published prices on Amazon's system. Feel free to tell me that I'm wrong with these calculations - but be specific if you find any problems, as I suspect others may glom onto these figures as gospel and I'd hate to have the wrong data in there. http://www.loligo.com/asterisk/misc/amazon-ec2.xls The net of my calculations is that a small instance of 20 users in a standard office environment would cost about $75 per month, which when compared to running a server in-house works out to be (raw cost, not including admin time and not discounting out-of-office bandwidth) only $38.56 more. Very interesting. For 20$ or slightly more you can rent a Xen or OpenVZ virtual host which will probably do as well. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com mailto:jabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir http://iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users And in France it is possible to have a dedicated server with 100 mbit /160 gb hdd 1.6 Ghz for 19€ and unlimited bandwith, and it is real unlimited. Seriously? Where? Sign me up! N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Amazon Flexible Payment System - micropayments finally cracked?
James Moore wrote: Notice that one of the prohibited items is: # Phone Services - includes 800 or 900 phone services and audio text services, prepaid phone cards, and prepaid phone services. https://payments.amazon.com/sdui/sdui/about?acceptableuse Google Checkout started with these limitations, but they were eventually eradicated. It could be that we'll see these restrictions eventually done away with on Amazon as well. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
Ira wrote: At 09:30 AM 1/27/2009, you wrote: People are always going to ask stupid questions. For me it's not so much the stupid questions as the expectations that we're here to solve their problems according to their needs. If that continues to happen and the noise level gets high enough those that have the most to offer will leave and all will be lost. Maybe there needs to be a beginner list and posting on this becomes invite only from people who participate on that list. Ira And which kind soul is going to post on the beginner list to help beginners, but still be annoyed to the point that he'd leave the non-beginner list because of all the beginner questions? And who does the inviting? Suddenly, I see poor John Todd having wy too much to do. ;) N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Root Password not taking
Steve Edwards wrote: On Thu, 22 Jan 2009, Wilton Helm wrote: If some of your directories like /home and /user have separate mount points, they don't have to get wiped out in the process. If there is any reason to suspect a hack, re-installation is the only way. I would replace the suspect drive and do a fresh install on a fresh drive. If you can bring it up to current patch level before exposing it to the 'net, all the better. Having the suspect drive available to crib configuration details from will come in handy. Just mount it read-only on a non-executable mount point. After a hack, no executable or configuration file can be trusted and all data is suspect so even if /home and /us[e]r are not clobbered, they cannot be trusted. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ Have to agree with Steve there. While a majority of hacks are just script kiddies using the vulnerability du jour, some are quite expertly done. I'd a friend in college who hacked into the university's main servers and spent a lot of time replacing system binaries with his own that he'd tailored to have the same byte count and same overall properties (with hidden extra switches here and there) so they wouldn't be readily noticed. This was WAAAY back in the day before things like tripwire and the like, but a careful hacker can become next to undetectable. The only SURE solution is to wipe the drive and start fresh, making sure to patch any holes through which the hacker might have come while you're doing a new install. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] evaluate SIP response codes in dialplan
Take a look (if it still exists) at the Asterisk B2BUA project. It has a patch that adds direct access to SIP response codes. It takes a little modification of the patch file to use in some of the newer asterisks (and to strip out the one codec option that's somewhat irrelevant), but it's a good starting spot. N. Klaus Darilion wrote: Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken, error, circuit busy (e.g. 503) 486 is mapped to DIALSTATUS=BUSY but both 503 and 404 is mapped to DIALSTATUS=CONGESTION As when Asterisk forwards the response with SIP to the caller the same response code is used, I suspect this information must be stored somewhere inside the channel variable. So, are there any means to access it? thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What are the various models of DID providers
. ITSPs/VoIP providers retailing VoIP services (be it wholesale origination trunking, or full-featured end-user oriented services like hosted PBX, or whatever) are customers of carriers, not carriers themselves. This key fact is often obscured by the marketing language of VoIP providers, which are NOT carriers (although most carriers certainly provide VoIP services like DID origination too). Some claim to be carriers in some sense of the term; this is false, they are not carriers as per the definition I have outlined. Some seem to imply ownership of numbers; they do not own them, they buy them from carriers. Number portability also confuses this discussion because people often talk about porting numbers into and out of VoIP providers as such. It doesn't actually work like that. Only carriers port numbers amongst themselves. You have to be a carrier to participate in the portability regime. When a VoIP provider ports in a number from a customer of some other VoIP provider, this process is accomplished through backoffice channels to their respective underlying carriers. For example, when a customer leaves provider A for provider B, provider B has its underlying carrier (or one of them) port the number from provider A's underlying carrier on behalf of the customer. Porting, like PSTN trunking itself, is a derivative process. (Of course, there do exist some regulatory guidelines for protecting customers to a certain extent from the fact that their VoIP provider doesn't really own numbers, and also serve to convey to the end-customer a rudimentary ownership of their numbers. Specifically, end-customers have the right to have their number ported to a different provider and in theory, compliance from the underlying provider and carrier is mandatory. In theory. It doesn't always work that way in practise.) Wholesale DID providers are resellers of carrier services and the general purpose they serve in that value chain is very similar to that of other types of VARs, distributors, and other middle-men. The essence of their rationale in the market has to do with the same sorts of economies of scale as wholesale in other industries; it is not, traditionally, economical for carriers to sell small amounts of DIDs, push small amounts of traffic, provide technical support and interoperability with relatively low-end customer premise equipment, or market to and acquire those types of customers. Carriers want large commitments and traffic volumes from organisations that know what they're doing in this space, so if you've got a small business Asterisk PBX going and need 20 numbers, you go to companies that specialise in that sort of thing and not the carriers themselves. The carriers aren't interested in trying to work with your Asterisk, deal with such beans in revenue terms, or market to you. That's the general picture, anyway. Some of this is changing, and some carriers are approaching smaller users increasingly for direct VoIP trunking. And of course, customers with very large volumes of traffic can go to the carrier directly and often do, if the business case for it is right. The VoIP wholesale DID providers traditionally interfaced with the carriers via hard TDM links such as ISDN PRIs or, less commonly, SS7, and often order very large links (i.e. channelised DS3s worth of PRIs and up). The DID provider's equipment would then spit out VoIP on the other side to you, and they would provide a variety of value-added backoffice tools and business processes to take care of provisioning (i.e. ordering and decommissioning numbers) and billing matters. So, the VoIP providers made the capital investment in the sorts of equipment, circuits, and contracts required to do that on your behalf and just sold you the VoIP trunking and numbers that ultimately result. They also take care of billing and other headaches you'd also face dealing with carriers via an intra-industrial channel. This is changing now as more and more carriers are offering SIP trunking to their wholesale customers, which means that VoIP providers themselves can now pick up the traffic over the Internet or via a dedicated private IP link without having to deal with all that TDM stuff. This lowers the barriers to entry and capital requirements to become a VoIP service provider and has a positive impact on pricing, although it does have the problem of attracting a lot of fly-by-night operators who think they need little more than to throw up an Asterisk box and some rudimentary PC hardware to sell DIDs. This makes it harder to tell the more bricks and mortar operations from something that is a purely virtual and possibly haphazard resale play. Matter of opinion, I suppose. Of course, not all the business models are this simple; sometimes there are more complicated, multiple levels of resale involved
Re: [asterisk-users] What are the various models of DID providers
Alex Balashov wrote: SIP wrote: What's interesting is the number of caveats and mixes even in the CLEC and ILEC world. I work with a CLEC that is also an ILEC (in certain areas), since they encompass various areas in Georgia (and own the state's largest contiguous network, passing through old rural ILEC lines (now purchased and updated)). They maintain CLEC status in some areas because they're not the incumbent there, but it helps them continue their network across lines owned by the incumbent with various peering agreements and the like. One of the interesting things we ran across was a discussion with them about UNEs. They provide strictly data lines throughout the state, and their CLEC status allows them the purchase of UNE DS1s and DS3s at exceptional rates to provide data to small installations in counties and municipalities. I don't know that the price of UNE DS1s and DS3s is really all that exceptional. Sure, it seems impressive that you can get a T1 in LATA 438 for some odd $44, but once you factor in the costs of interconnection, CO colocation, EELs and interoffice mileage if not colocated in the CO to which the circuit is being generated, private SONET for backhaul, etc. Not to mention in that in urban areas the ILEC commonly suspends UNE pricing discipline on the grounds that the wire center is impaired - i.e. there is enough competition in the CO. That requires you to revert to wholesale / special access and pay a lot more. The interconnection, CO colocation, private SONET, etc, are already in place in something like 60 municipalities and 4 Atlanta metro areas. They're using the UNEs to cut costs. Honestly, you could ask me some complex questions about their network, but I don't know it all that well... However, upon reading the current governmental regulations (the somewhat more recent E911 provisions), it states specifically that a UNE MUST have, to each logical circuit, an assigned DID and the ability to pass voice traffic to the local E911 call center. The problem being, of course, that these were for data and not voice. However, the law is very clear (in that murky way in which laws are), and to avoid possible hassle down the road from an unfriendly ILEC or an upset ATT who wanted to press the issue, it was decided that DIDs would be purchased and assigned to those UNE circuits as they were deployed. I'm not sure I follow. Voice trunks need routing to E911 tandems, but what do data circuits have to do with this? Nothing. This is part of a law governing who can get UNEs. I don't have it handy, but I'll look it up on Thursday (when I get back to the office... have it in email there but not here for some reason). This is where we came in, and where the middle-man model still works to some degree. They could simply buy great swaths of DIDs for themselves at ridiculously low rates (being a LEC), but the caveat there is that the DIDs have to be USED, or they're reassigned. Depends on the area; NANPA and pooling blocks aren't necessarily cheap. The numbers they quoted us were reasonable. Something like $500 for 2000 DIDs. Or possibly $200. Again... fuzzy on the exact numbers there, but I remember it was quite good. We stepped in to provide DIDs (which we purchase elsewhere) to their UNE circuits and maintain them (even with no use), as well as maintaining the information for E911 dispatch on each of the circuits (assuming, for the sake of argument, that someone were to convert the data line into voice). Thus, they can get the rates they want on the UNEs they deserve, and not worry about the hassles of actually dealing with the technology and contracts on the voice side that is simply not part of their core business model. Why would they have to deal with this when someone buying directly from ATT off the special access tariff doesn't? (i.e. independent ISPs) Again. Thursday I'll have that info. Now this is, to be certain, an odd and unusual case. I doubt we could find too many customers if that were our ONLY sort of business. But it does illustrate your point that there is still, for now, a logical place for the middle men companies in some situations. Agreed, although I'm still very confused as to why you need DIDs for data UNEs. Is this some bizarre feature of their ICA or something? It has to do with a recent modification of the telecom laws concerning who's allowed to have access to the UNEs and who isn't, and it stipulates that, in order to have access to them, you're now required to be able to provide E911 service over them (as the law seems to just outright assume that you'll be using them for voice). The law itself doesn't seem to take into account that there's even a possibility that someone might use a UNE for ONLY data (like many of the more recent modifications to the telecom act, it appears to have been hastily
Re: [asterisk-users] Bring India together
Look, ma... spam! We dun never seen that 'n before. N. Sunkara RaviPrakash wrote: Hi, Imagine a billion Indians together. Already 3 million Indians have chosen Indyarocks.com to bring India together. I am already part of it and dont be surprised if you find most of your other friends too :). Also you can send Unlimited Free SMS to your friends in India from anywhere in the world. Click here to get together http://www.indyarocks.com/register_step1.php?invitor=MjEyMjkyMA==emailencryp=YXN0ZXJpc2stdXNlcnNAbGlzdHMuZGlnaXVtLmNvbQ==. -Sunkara RaviPrakash Please note: This message was sent to you by a user at Indyarocks.com. Click here http://indyarocks.com/static/unsubscribe.php?eml=YXN0ZXJpc2stdXNlcnNAbGlzdHMuZGlnaXVtLmNvbQ==uid=MjEyMjkyMA== in case you do not wish to receive any invite from this user. Click here http://indyarocks.com/static/unsubscribe.php?eml=YXN0ZXJpc2stdXNlcnNAbGlzdHMuZGlnaXVtLmNvbQ== if you do not wish to get any invitations from any Indyarocks user. If you have any queries please contact us at priv...@indyarocks.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK
*From:* asterisk-users-boun...@lists.digium.com on behalf of Philipp
Kempgen
*Sent:* Thu 18/12/2008 4:17 PM
*To:* Asterisk Users
*Subject:* Re: [asterisk-users] Dial timeout with SIP - how to set
timeout for INVITE ACK
Julien Chavanton schrieb:
I have a concern with Dial command, I want to enable a secondary
route with a remote partner, if the first route fails then we use the
second one :
Solution1: it will try both (there will be 2 simultanious actives
calls ringing) this is not clean when calling an endusers
exten = _X.,1,Dial(SIP/${ext...@remote-sip1,5
SIP/${ext...@remote-sip1,5 mailto:SIP/$%7bexten...@remote-sip1,5 )
exten = _X.,1,Dial(SIP/${ext...@remote-sip2,5
SIP/${ext...@remote-sip2,5 mailto:SIP/$%7bexten...@remote-sip2,5 )
You can't have the same priority (1) more than once per
extension (_X.).
Solution2: it will wait until 5 seconds of timeout (on answer) and
then try the second alternative n
exten = _X.,1,Dial(SIP/${ext...@remote-sip1,5
SIP/${ext...@remote-sip1,5 mailto:SIP/$%7bexten...@remote-sip1,5 )
exten = _X.,n,Dial(SIP/${ext...@remote-sip2,5
SIP/${ext...@remote-sip2,5 mailto:SIP/$%7bexten...@remote-sip2,5 )
the problem is we can not select what timeout represents, timeout on
ACK from INVITE would be perfect I think (1 second for example),
timeout for answer ? this is to hard to predict, some mobile phone can
ring for 30 seconds, etc.
So why not use 30 and let Asterisk take care of the SIP details/
timeouts?
And just to be sure: Don't put those mailto things in
extensions.conf. :-)
Philipp Kempgen
Julien Chavanton wrote:
So why not use 30 and let Asterisk take care of the SIP details/
timeouts?
Asterisk will wait the until it receive answer or timeout
I need to timeout a SIP call on SIP INVITE ACK, in ISDN for exmaple
this is translated to PROCEEDING
Meaning I have received the call, now I will look what to do with it
The result with the suggested timeout is not good enought, you may
wait for the whole timeout even if the other side as not sent
anything, this will be the case for all your calls, depending on the
timeout this would be killing the traffic.
It sounds as though you want the result of the SIP INVITE (looking for,
say, a provisional 1XX response) and want the timeout to be set for
whether or not you receive the provisional response in time? i.e. You
want to know if the remote address/proxy is up and running before you
bother trying to wait on it for very long. Is this right? Or am I
missing the point of the question?
N.
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Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK
It's a valid concern, but be prepared for people to tell you that this
should be done with the qualify parameter to determine if a host is up
and running. Not the most ideal way to handle it, I'll agree. But the
SIP proxy functionality of Asterisk is limited (as it's not intended to
be a SIP proxy).
We use a modified source to enable the return of SIP response codes into
our AGI scripts that help us do more intelligent logic in this regard
(mostly for LCR functionality). It could be modified, I suppose, to
look for the return of a provisional response indicating that the remote
proxy is up and running and responding to SIP requests. Not sure how
you'd then initiate a timeout, though.
N.
Julien Chavanton wrote:
You want to know if the remote address/proxy is up and running before you
bother trying to wait on it for very long. Is this right? , yes this
would be a good start ?
- But the IP could be up and the SIP service down, we need a signaling
timeout, I beleive a good way in term of responsability would be :
If I do not receive a response to the SIP INVITE in timeout duration
then I would cancel the call and try with another route.
- With AGI can we control and react to the signaling events, I guess not ?
Thank you
*From:* asterisk-users-boun...@lists.digium.com on behalf of SIP
*Sent:* Thu 18/12/2008 6:13 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Dial timeout with SIP - how to set
timeout for INVITE ACK
*From:* asterisk-users-boun...@lists.digium.com on behalf of Philipp
Kempgen
*Sent:* Thu 18/12/2008 4:17 PM
*To:* Asterisk Users
*Subject:* Re: [asterisk-users] Dial timeout with SIP - how to set
timeout for INVITE ACK
Julien Chavanton schrieb:
I have a concern with Dial command, I want to enable a secondary
route with a remote partner, if the first route fails then we use the
second one :
Solution1: it will try both (there will be 2 simultanious actives
calls ringing) this is not clean when calling an endusers
exten = _X.,1,Dial(SIP/${ext...@remote-sip1,5
SIP/${ext...@remote-sip1,5 mailto:SIP/$%7bexten...@remote-sip1,5 )
exten = _X.,1,Dial(SIP/${ext...@remote-sip2,5
SIP/${ext...@remote-sip2,5 mailto:SIP/$%7bexten...@remote-sip2,5 )
You can't have the same priority (1) more than once per
extension (_X.).
Solution2: it will wait until 5 seconds of timeout (on answer) and
then try the second alternative n
exten = _X.,1,Dial(SIP/${ext...@remote-sip1,5
SIP/${ext...@remote-sip1,5 mailto:SIP/$%7bexten...@remote-sip1,5 )
exten = _X.,n,Dial(SIP/${ext...@remote-sip2,5
SIP/${ext...@remote-sip2,5 mailto:SIP/$%7bexten...@remote-sip2,5 )
the problem is we can not select what timeout represents, timeout on
ACK from INVITE would be perfect I think (1 second for example),
timeout for answer ? this is to hard to predict, some mobile phone can
ring for 30 seconds, etc.
So why not use 30 and let Asterisk take care of the SIP details/
timeouts?
And just to be sure: Don't put those mailto things in
extensions.conf. :-)
Philipp Kempgen
Julien Chavanton wrote:
So why not use 30 and let Asterisk take care of the SIP details/
timeouts?
Asterisk will wait the until it receive answer or timeout
I need to timeout a SIP call on SIP INVITE ACK, in ISDN for exmaple
this is translated to PROCEEDING
Meaning I have received the call, now I will look what to do with it
The result with the suggested timeout is not good enought, you may
wait for the whole timeout even if the other side as not sent
anything, this will be the case for all your calls, depending on the
timeout this would be killing the traffic.
It sounds as though you want the result of the SIP INVITE (looking for,
say, a provisional 1XX response) and want the timeout to be set for
whether or not you receive the provisional response in time? i.e. You
want to know if the remote address/proxy is up and running before you
bother trying to wait on it for very long. Is this right? Or am I
missing the point of the question?
N.
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Re: [asterisk-users] top posting again [was: Re: CDR Design]
Steve Edwards wrote: On Wed, 17 Dec 2008, Danny Nicholas wrote: OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M @#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHAT MSN IS FOR. Spoken like a true top-poster... Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Top posting. Bottom posting. Honestly, if you can't use an effing scrollbar, please tell me so I can take you out back and beat you to death with something heavy. The .5 seconds it takes to scroll from one end of a message to another is no excuse for spending 2 minutes writing a tirade about how you don't like to spend that extra .5 seconds. I swear. You people need to get up, walk away from the computer, go outside and realise that this level of egocentrism is incredibly unhealthy. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Jeff LaCoursiere wrote: On Sun, 14 Dec 2008, Tzafrir Cohen wrote: Right. So for those of us who want to do simple things and avoid complicated stuff such as telephony in shoddy continent of North America, could you please provide data for your country? So far we have AU, IL and NZ. Not that I am trying to put down the project, but I am struggling to understand how this will be useful to anyone. What will you actually *do* with this information once it is compiled? j Step 1: Compile a list of country codes broken down into landline/mobile to the best of anyone's random guesswork. Step 2: ??? Step 3: Profit!!! N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
Michael wrote: Yes, but with an A-Z carrier, this can become risky when landline calls are charged very differently to cellular calls, as is the case in NZ, Australia and many other countries, unless someone is just a 'virtual' provider and letting their up line do the invoices. Some of our providers have rates that don't change much (they've built in tolerance levels to them, so that if there's a fluctuation of 5c in one direction or another, it won't much matter. Some of our providers pass us a new A-Z rate deck every WEEK. Including rate changes and prefix changes. Countries go from 5 prefixes to cover mobile, to 25, and then to 18, and then to 7, and then to 130... changing on a weekly basis (and sometimes daily in a few countries we deal with). You'd need to get more than just the Asterisk community into this. You'd need an overall organisation of underlying carriers worldwide which could update their destinations whenever there's a change. As a project, that's not only daunting technologically, but massively difficult politically. A lot of those UCs aren't going to WANT to join your coalition of information. After all, what's in it for them? Add to that that the information it gives YOU is not going to be applicable on a grand scale. While the actual carrier who maintains prefixes 56-110 may change their structure on a weekly basis, it's possible the contracts they have with providers you'd be using have differing information available to the provider. Which means that just because something in the landscape changes, the rates may not change to you (or might change to YOU, but not to someone who uses a different provider that uses the same UC). I'm not sure I can see the value of a community-driven effort to keep track of things which, by nature, are not applicable to everyone in the community, as we all have our own contracts with our own providers and our own set of rates based on our own conditions of traffic. Perhaps you can explain better the value of the proposition in more detail. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using DECT phones as SIP phones?
Fred Posner wrote: On Dec 5, 2008, at 11:31 AM, Michael Graves wrote: On Fri, 05 Dec 2008 10:59:54 -0500, Neil Fusillo wrote: Michael, Was there something particularly special you had to do to get your M3 to work? I'm now on my second one from E4 Technologies (from whom I'm still waiting for a return service call), and both of them, after following the instructions in the manual, have the same problem. Neither one will pick up an IP from DHCP. If I set a static IP with the handset (which pairs to the base just fine), I can then ping them when they boot, but the web configuration service is unavailable (nothing running on port 80, as if it never gets through the boot process). Hitting the up volume button on the phone gives me info of an ff:ff:ff:ff:ff:ff mac accress, no IP (with DHCP) or the static I've put in, and a Boot Status of 'Failed' Was there a trick to getting these running? I've done a factory reset. I've tried different cables. Different switches. Now different handsets and base stations. Nothing. Same message every time. Did you run into anything similar, or am I just lucky? I had no such troubles. Just plugged in and went. What firmware release does it have? Mine were v1.22 just the other day when I checked, but I have it set to D/L the current beta automatically once a day. Michael -- Mine automatically updated as soon as it got an IP address. (default setting) That seems to be the core problem. Neither of mine ever seemed to get IP addresses. And even setting it as a static IP makes it pingable, but nothing else in its networking stack seems to run and it can't update. Two duds in a row, though... I was thinking it might be user error (as I'm sure the supplier is thinking). However, I don't APPEAR to be doing anything wrong as far as I can tell (and network/voip hardware is hardly new to me). N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday, Asterisk is 9 years old!
randulo wrote: Hi, December 5th, 1999 was the initial release of Asterisk by Mark Spencer. We'll be celebrating this by gathering as usual at 12 Noon Eastern (9AM Pacific, 10 MST, 11 Central, 5PM UK and Western EU) for the VoIP Users Conference. You can get all the dial in information at http://VoipUsersConference.org including info on a SipAddHeader() kludge to avoid DTMF problems. IRC is Freenode.net #voip-users-conference join this even if you can't call in. Call via SIP: [EMAIL PROTECTED] (thanks to OnSip.com) Call via PSTN (724) 444-7444 DTMF 22622# 1# or try this: [EMAIL PROTECTED] (thanks to IdeaSIP.com) or to just look up talkshoe server IP: ts.x2z.eu (thanks top me for the DNS record) We start about 15 minutes to the hour with an informal chat. Join us anytime, but especially, grab a virtual beer and join us Friday the 5th. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users December 5th, 1879 is also the date when the first automatic telephone switch was patented. A good day for telecom all-round. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
Doug wrote: At 18:56 12/1/2008, Tilghman Lesher wrote: On Monday 01 December 2008 06:21:33 pm Doug wrote: We tell our customers that they are not allowed to download copyrighted material. So your customers are only allowed to download public domain material? That kind of restricts the amount of information available on the Internet. Nitpick: just about everything, including this email, is copyrighted by somebody. Forbidding the download of copyrighted works is not only a draconian policy, but may actually violate several copyright laws (you're interfering with a copyright owner's right to distribute his/her/their works, and courts are generally not very sympathetic with your position). Oops! Didn't mean to start a fire here. I meant to say illegal copyrighted material. Also, if they are using up hundreds of Internet connections, we can see that. It essentially causes a Denial of Service situation for other users on that leg of our wireless network. The system supposedly has rate limiting, but seems to get overloaded when someone goes completely nuts with BitTorrent. We are working on ways to limit the number of simultaneous connections. When we get a copyright infringment notice from our upstream provider, we are compelled to reprimand the user. I don't think we have sent a customer to the shower even if they had several notices. Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Es tut mir leid für das Durcheinander meine Brüder! This is the classic logical fallacy that people seem to perpetuate when reporting news about P2P activity. ISPs oversubscribe. It's a common practice, and reasonably valid. But when you oversubscribe, you use a model based on 'projected' use of the available circuits and bandwidth. If you have a user who pays for a circuit that you've advertised as an X Mb line, and he uses X Mb ALL the time, he's using what he's paying for. If you then proceed to tell him that he can't do that, you're either wrong or you're not being up front enough with your pricing and marketing materials. You can't then proceed to blame the customer for use you did not anticipate. Imagine a farmer who sells tomatoes. He's promised you a bushel, but he gets a harvest of only so many. You walk up to the counter just after he's sold all of his tomatoes to someone and he tells you Sorry. There are no more tomatoes because that customer before you just 'stole' them all from you. He's abusing his privileges by buying up my whole crop. Now whose fault is it that you don't get the tomatoes you want? Is it the customer's fault for buying all the tomatoes the farmer sold him? Or is it the farmer's fault for selling them? The same works with the ISP vs P2P argument. If the ISPs were up-front about saying that they do not intend for you to actually USE the bandwidth you think you're paying for, I would say they had a leg upon which to stand. However, hiding this information from the customer and then blaming the customer when he does what he believes is well within his rights... it may play well in the media, but it's bad for the whole system and is incredibly divisive. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
Doug wrote: At 04:03 12/2/2008, Benny Amorsen wrote: Doug [EMAIL PROTECTED] writes: Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Yes, just like the airlines need to somehow control those users who keep showing up to the flight they booked, every single time! It's impossible to do overbooking with customers like that, so we need to find ways of punishing them. What happens if everyone who owns a car drives it at the same time? Owns a telephone and uses it at the same time? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If everyone who owns a car drives it at the same time, there's lots of traffic. You know who gets blamed? The right people -- the people to create the infrastructure. Drivers aren't blamed for driving their cars when they want to as long as they do it legally as prescribed by the very open and easy to find laws. If everyone who owns a telephone uses it at the same time, it's just like the Internet issues. Telephone companies also practice oversubscription. But it's clear to everyone that it's the phone company that doesn't have the capacity for it... people don't blame the customers for using their phone. They pay for it. They should be able to use it when they want. But if everyone uses the Internet access they pay for? Suddenly, they're violating a user agreement (usually not a specified one in the case of many ISPs) or a usage policy and it's all that crazy P2P to blame. They're stealing bandwidth from other users. Which is absolute poppycock. That's a marketing spin on poor infrastructure planning. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
Doug wrote: At 07:00 12/2/2008, SIP wrote: Doug wrote: At 18:56 12/1/2008, Tilghman Lesher wrote: On Monday 01 December 2008 06:21:33 pm Doug wrote: We tell our customers that they are not allowed to download copyrighted material. So your customers are only allowed to download public domain material? That kind of restricts the amount of information available on the Internet. Nitpick: just about everything, including this email, is copyrighted by somebody. Forbidding the download of copyrighted works is not only a draconian policy, but may actually violate several copyright laws (you're interfering with a copyright owner's right to distribute his/her/their works, and courts are generally not very sympathetic with your position). Oops! Didn't mean to start a fire here. I meant to say illegal copyrighted material. Also, if they are using up hundreds of Internet connections, we can see that. It essentially causes a Denial of Service situation for other users on that leg of our wireless network. The system supposedly has rate limiting, but seems to get overloaded when someone goes completely nuts with BitTorrent. We are working on ways to limit the number of simultaneous connections. When we get a copyright infringment notice from our upstream provider, we are compelled to reprimand the user. I don't think we have sent a customer to the shower even if they had several notices. Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. Es tut mir leid für das Durcheinander meine Brüder! This is the classic logical fallacy that people seem to perpetuate when reporting news about P2P activity. ISPs oversubscribe. It's a common practice, and reasonably valid. But when you oversubscribe, you use a model based on 'projected' use of the available circuits and bandwidth. If you have a user who pays for a circuit that you've advertised as an X Mb line, and he uses X Mb ALL the time, he's using what he's paying for. If you then proceed to tell him that he can't do that, you're either wrong or you're not being up front enough with your pricing and marketing materials. You can't then proceed to blame the customer for use you did not anticipate. Imagine a farmer who sells tomatoes. He's promised you a bushel, but he gets a harvest of only so many. You walk up to the counter just after he's sold all of his tomatoes to someone and he tells you Sorry. There are no more tomatoes because that customer before you just 'stole' them all from you. He's abusing his privileges by buying up my whole crop. Now whose fault is it that you don't get the tomatoes you want? Is it the customer's fault for buying all the tomatoes the farmer sold him? Or is it the farmer's fault for selling them? The same works with the ISP vs P2P argument. If the ISPs were up-front about saying that they do not intend for you to actually USE the bandwidth you think you're paying for, I would say they had a leg upon which to stand. However, hiding this information from the customer and then blaming the customer when he does what he believes is well within his rights... it may play well in the media, but it's bad for the whole system and is incredibly divisive. Yep. In our contract we say things like shared, best efforts, etc. If you want a dedicated pipe with guaranteed bandwidth, you gotta pay a hefty price. Then I applaud you for doing something most ISPs do not do -- being a LITTLE more up-front about the realistic limitations of the service. ISPs tend to promise the world to grab users, knowing full well they can't deliver. And when the users try and use what they've been promised, they're blamed for bringing down the network. And what's worse, this clear spin line is propagated throughout even LARGE media organisations as an accepted fact. P2P Steals Bandwidth. That's reported as a simple and plain fact when, in reality, you can't steal what you've been allotted by your ISP. If the ISP said we only have the capacity for X users to use their service ALL the time, so users who want to pay basic usage and use little can pay this small sum, or users who want to get unlimited but very throttled and pay this larger sum, it would go a long way toward fostering trust all-round without relying on misinformation and vilifying the users who are using what they think they're paying for. Of course, it would be a marketing nightmare, as the other ISPs would say, But we have UNLIMITED access at much higher speeds -- clearly lying about their capacities for the sake of bamboozling non-tech-savvy customers, and then relying on media organisations to propagate their disingenuous epithets against the P2P crowd. N
Re: [asterisk-users] Any other free toll free SIP providers out there?
Tom Browning wrote:
FWD (Free World Dialup) allows any SIP call to US toll free numbers
via [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
This works WITHOUT the need to be registered at FWD so in my dialplan
I have something like:
exten = _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r
http://fwd.pulver.com/*$%7BEXTEN:1%7D,60,r)
exten = _8.,2,Hangup
And I just dial 8-1-8xxyyy and presto ... calls go through just
fine 99% of the time.
I'm wondering if there are any other providers out there that allow
calls to toll free numbers without the need of being registered? I'd
like to have a backup or two.
Tom
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IdeaSIP doesn't require registration for free Toll-Free.
[EMAIL PROTECTED]
N.
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Re: [asterisk-users] Load balancing Asterisk.
Unless the LB is SIP-aware, and can maintain a SIP session, I don't see how it would work. As the SIP command stream sends discrete commands, without some sort of basic level of session awareness, there's no guarantee over a reasonable-length call that the INVITE and BYE would even get sent to the same Asterisk box. If there are on-hold messages or transfers occurring, you add even more possibility of error into the mix. Now... you could do some sort of VERY long session timeout, but overall, that's a hack that's going to drop your concurrent connection count faster than using a smaller box would. I don't know of any functioning, SIP-aware load balancers at the moment. Doesn't mean they don't exist. I just can't think of any off the top of my head. N. Nitzan Kon wrote: Alex, I realize and agree that hardware load balancers are actually software based. I'm less concerned about that and more about the general specs: Foundry ServerIron XL: rated for 1,000,000 concurrent connections Linux box where OpenSIPS is sitting: rated for ...??? Not to mention a simple rule on a load balancer would be much, much easier to implement. All I need is IP-based load balancing so installing and maintaining OpenSIPS is an overkill. Again, I appreciate the feedback but I am not asking nor looking for a software solution. My question is simple: Will a HARDWARE load balancer work? any reason why it WON'T work? Thanks! --- On Thu, 11/20/08, Alex Balashov [EMAIL PROTECTED] wrote: What do you mean by hardware options? There are no ASIC-assisted SIP load balancers out there. :-) The embedded hardware-based options are load balancers built just like PCs - often on top of a UNIX kernel - that run a software application-aware load balancing suite. Your best bet is a proxy for the round-robin part, and Linux-HA for the high availability of the proxy, as Grygoriy suggested. Nitzan Kon wrote: --- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. Thanks for the info. :) I want to stay away from software solutions however. Are there any hardware solutions? would a plain load balancer work? If we can't get it working with a LB we'll look at OpenSIPS, but I'd like to explore hardware options first. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Alex Balashov wrote: I was about to say, I'm sure F5 can do it... but... price was over 6 figures Why??! It's spending money on these types of things when they are unnecessary that is the undoing of every struggling VoIP provider I watch, in the misguided belief that only will half a million dollars get you enterprise strength. That was the conventional wisdom about Linux ten years ago too. Who's saying that now? Ditto. F5 has ALWAYS been overpriced. Incidentally, anyone who wants to know, F5 is a unix-based box, just like the others. Last we used the F5s, they were all running a slightly modified BSDI. And only slightly modified in packaging. As for the current F5 SIP load balancer, we tried it a few years back and it was a dismal failure. It wanted to do cookie-based SIP load balancing and only worked with certain SIP proxies. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing SIP calls dropped after 30 seconds.
Kurt Knudsen wrote: Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to any number and (4/5 times) it will drop the call after 30 seconds. I noticed that the little timer that pops up on the LCD on the phone is missing when a call will be dropped. This timer appears when the phone is answered, so I have about 30 seconds to talk to them before the call is just dropped. Known Causes: It's a NAT issue, I know that much, I just don't know how to fix it. SIP debugging shows that it attempts to retransmit packets to my phone and since it can't, it drops it after 30 seconds. Log snippet: -- Executing [EMAIL PROTECTED]:19] Dial(SIP/203-b7a2b558, SIP/bw_outbound/+18005551212|300|) in new stack Audio is at public IP port 11968 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 216.82.224.202:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] IP Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 07 Nov 2008 19:06:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 151.196.61.115 s=session c=IN IP4 public IP t=0 0 m=audio 11968 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Called bw_outbound/+18885551212 FreePBX*CLI --- SIP read from 216.82.224.202:5060 --- SIP/2.0 100 Giving a try Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 - --- (8 headers 0 lines) --- FreePBX*CLI --- SIP read from 216.82.224.202:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 Record-Route: sip:216.82.224.202;lr;ftag=as3ed791f3 From: 8881231234 sip:[EMAIL PROTECTED] IP;tag=as3ed791f3 To: sip:[EMAIL PROTECTED];tag=VPST50603522629853 Call-ID: [EMAIL PROTECTED] IP CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Content-Type: application/sdp Content-Length: 184 v=0 o=- 1226084867 1226084868 IN IP4 209.244.42.253 s=- c=IN IP4 209.244.42.253 t=0 0 m=audio 64706 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 - --- (10 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.244.42.253:64706 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.244.42.253:64706 -- SIP/bw_outbound-08bf43d0 is making progress passing it to SIP/203-b7a2b558 Audio is at public IP port 16244 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (NAT) to 172.16.2.203:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.2.203;branch=z9hG4bKed293df65EAFD78F;received=172.16.2.203 From: Me sip:[EMAIL PROTECTED] IP;tag=28354B-27A53F00 To: sip:[EMAIL PROTECTED] IP;user=phone;tag=as600b952c Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] IP Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 public IP s=session c=IN IP4 public IP t=0 0 m=audio 16244 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (10 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.244.42.253:64706 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video
Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]
Greg Woods wrote: On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. Socialism has already completely failed. What should we do, go back to a barter economy? :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users WHAT?? The telco has raised the cost of my call to Italy from 1 hen per minute to 3! I'll be out of chickens in WEEKS if this keeps up. Next thing you know, the cable co will start demanding 1.5 cows per month. Of course, they don't care that I can't give them .5 cows without wasting a WHOLE one. It's terrible! I only wish there were something we could use for payment instead of commodities -- maybe some sort of note that took the place of a physical commodity! Hey...that sounds like a terrific idea! N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
randulo wrote: On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves [EMAIL PROTECTED] wrote: In any case, the wideband bridge for this weeks VUC call supports only G.722. But we do plan to make a recording of both conference version available, AFAIK? r But will it be a high-def recording? N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange ring tone: Long-Short-Short
Joseph wrote: I'm using Linksys SPA3102 adapter and have a strange ring tone: Long-Short-Short or Long-Long-Short-Short Does anybody know which setting adjust this ring tone on SPA3102 Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab Interestingly, I get that, too... but only SOMEtimes. I swear, the number of weird issues I've had with the Linksys ATAs is staggering -- occasionally losing all their stored configs, sometimes refusing to set an IP either via DHCP or manually, weird rings, etc. This has happened on at least a dozen of them, too. It's a wonder I keep buying the things, but unfortunately, they have the reputation as being the 'best' out there. Kind of sad. It's almost certainly going to be somewhere under the regional tab in one of the distinctive ring areas. But since mine are default, and I get weird patterns only sometimes, I'm hesitant to tell you what values are proper there. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER + Asterisk
That's not actually true. SER is very much alive and well and under constant development. How do I KNOW it's constant development (other than the chatter on the mailing list)? Because things keep changing in CVS, but there never seems to be a 'release' version. Just a release candidate. ;) Seriously, though... this seems to be a popular misconception. I hear it a lot. Where did you come across the information that SER is no longer developed? N. Alex Balashov wrote: No, the issue isn't my value or preference. The issue is that SER is no longer maintained or developed and has not been for several years. Tobias Wolf wrote: Alex Balashov schrieb: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Well, i am not getting the correct meaning of 'defunct', but from the last part of your suggestion i guess you value Kamailio/OpenSIPS more than SER. Are there some hard reasion for this. I am in the process of deciding which SIP server i want to use with Asterisk and just made a go at SER. Compilation was a little rough but it was manageable. I threw away every module which funtionality i didn't wanted at after it just worked. I was able to register SIP phones at the server and configure an outgoing rule so that every call that could not be handled by the SIP server would go to Asterisk. But i confess, that i didn't looked at the other two projects ... Maybe they are so much better. Can you please write one or two aspects that makes me understand better why this two projects are the better choice ? Thank you very much ... Tobias On Fri, October 17, 2008 9:36 pm, Joseph wrote: I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently Im using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call Asterisk. Can anybody provide a link to practical example. I'm comfortable with Asterisk but I just install SER and can not find appropriate example to follow on www.iptel.org web-page. There are a lot explanations but not enough practical examples to follow. -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER + Asterisk
Alex Balashov wrote: SIP wrote: Seriously, though... this seems to be a popular misconception. I hear it a lot. Where did you come across the information that SER is no longer developed? That seems to be a consequence of looking at the releases. Anyway, I spoke too soon in saying that there's absolutely nothing going on with the project whatsoever in terms of development. Yes... I'll agree the releases are a bit... odd. SER 0.9.6 (or possibly 0.9.7 -- I'm never sure) was the last actual 'release' labeled stable. However, SER 2.0 rc1 has been available for over a year now, and hasn't been granted that 'stable' label, even though I gather it's no more unstable than 0.9.6/7. All the while, work on SER 2.1 is commencing long before there's been a release of SER 2.0. It's incredibly difficult to follow. But this is where the OpenSER (now OpenSIPS) and SER projects differed in their ideology most often -- that of releases and documentation. SER was always a bit sparse on both, preferring to make up for it by way of solid innovations in the core code. Unfortunately, it's a bit like the tale of Seymour Cray. Here was a man who was convinced that if you built a supercomputer, people would buy it because it's the fastest thing out there, and building peripherals and/or software for it as part of the business plan was a waste of time and money. This ideological difference is why he left Control Data. This is why he was encouraged out of Cray Research. And this is why his final company, Cray Computer Corp failed -- that sort of missed idea that people will buy technology simply for the sake of having better technology. I see a lot of parellels there with OpenSIPS and SER. OpenSIPS is a stable plaform that has dozens of modules and documentation galore on how to mesh the system with this, that, and the other. SER has rock-solid, incredibly innovative core code, but prefers to leave the writing of modules and documentation as an exercise for the user, thereby making it perhaps overly difficult for anyone to implement or integrate. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How Secure Is Asterisk
It's not 100% secure. Like any dual-key encryption, it's subject to the classic man-in-the-middle attack. This is why the Windows Zfone app has the addition of a visual key you can read and coordinate with the recipient to determine if a MITM attack is occurring. But only if you know what you're doing. There is no such thing as an unbreakable system. YOU might not be able to attack it with your limited resources, but someone determined enough to do so, will find a way. N. Jonn R Taylor wrote: What about zfone project??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam Tam Sent: Tuesday, October 21, 2008 12:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How Secure Is Asterisk There are no 100% solution but we can only do our best. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of broadband Voice Sent: Tuesday, October 21, 2008 4:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How Secure Is Asterisk lol On Mon, Oct 20, 2008 at 3:34 PM, Sam Tam [EMAIL PROTECTED] wrote: VPN IP phone? Then firewall up the asterisk to disable any outside access and place the vpn server with the asterisk in a locked cabinet . Sure that will stop someone trying to physically listen to their call. Or they can always use the good old landline or mobile phone and let the government listen to them too/ Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Anness Sent: Tuesday, October 21, 2008 3:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How Secure Is Asterisk I am sure this has been discussed prior, however, I am sitting here and being asked this very question by my superiors. They are loving what I have done with our two Asterisk servers here; however, they keep asking me if it is secure or not. Of course, as with anything, I suspect that on a secure network they can be reasonably safe. However, realistically if I am using the asterisk server to make internal calls and discussion very private matters, how possible is it for someone to listen to calls? How good is the encryption if any over an IAX trunk? Steve Anness ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No reply to our critical packet
This message is usually caused by Asterisk not receiving an ACK after about 30 seconds of attempts. There are countless misconfigured UAs and proxies out there that don't handle ACK well, so it would be nice to be able to turn this 'feature' off. What's annoying is that the explanation has always been If we can't get an ACK, we can't send any RTP data. This is patently false, as the RTP will often work fine even if ACK handling is misconfigured (we see it all the time). But alas. As far as I can tell, there's no way to disable this check. I suppose I could code around it, but not being the world's most proficient C coder, I'm always afraid I'll break something else. ;) N. Andrew Joakimsen wrote: I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 2 (Critical Response) [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. It seems to me that the problem is the way Asterisk is handling this critical packet -- of course it can not be sent to 192.168.1.54, the phone is at that IP behind a NAT and the Asterisk server is not. I can make any other phone call from this same phone as long as it is not voicemail and I can be on the line for hours with no problem. I am really at a loss here. I have searched a bit and come up with nothing other than blaming the UA. I know the Polycoms dont have the best NAT support but besides this it works problem-free. It's odd I can make a call anywhere else even for hours and not have any issues at all but 30 seconds into a voicemail call it just drops app5*CLI sip show peer 17865221569 app5*CLI * Name : 17865221569 Secret : Set MD5Secret: Not set Context : blended-lcr Subscr.Cont. : sla_stations Language : en AMA flags: Unknown Transfer mode: closed CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 17865221569 VM Extension : 14193016245 LastMsgsSent : 0/0 Call limit : 2 Dynamic : Yes Callerid : CENSORED MaxCallBR: 256 kbps Expire : 63 Insecure : no Nat : Always ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : Yes Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 74.CENSORED.213 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 17865221569 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : OK (130 ms) Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Reg. Contact : sip:[EMAIL PROTECTED] app5*CLI core show version Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on 2008-07-09 01:41:43 UTC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: text/plain
Philipp Kempgen wrote: Andrew Kohlsmith (lists) schrieb: On October 5, 2008 12:22:37 pm Philipp Kempgen wrote: ---cut--- http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html http://lists.digium.com/pipermail/asterisk-users/2008-October/219541.html ---cut--- That quoted text is not very eye-friendly. Konqueror also renders these as huge one-line messages; I am blaming the mailing list archival software for this, as it is not being sufficiently suspicious of the data it's processing; it should be either stripping the pre tags or otherwise forcing them to be web-friendly, IMO. Sure. It could choose to transform HTML to plain text itself. But let's compare it to VoIP: If the other party offers codecs A and B and I want A I would negociate A instead of trancoding B to A myself. How can I know in advance or by automated means that what the other party sends using codec A is hardly useful. Philipp Kempgen This all depends on whether or not you take a descriptive or prescriptive approach to things. Perhaps the codec you want isn't actually offered. It's more an Aa, as opposed to just A, and your old client simply can't tell the difference. One might say that, with close to 300 million hotmail users as of February this year, if your email client doesn't decode Hotmail emails in a readable fashion, your email client is faulty. If you're a programmer and can't see fit to code around what may indeed be a bug, but is still the second most-used email service in the world, you're either a) lazy, or b) too stubborn to be allowed to complain. The truth is there are plenty of email clients that CAN decode Hotmail messages, and if you choose one that can't, you can't blame anyone but yourself. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]
Eric ManxPower Wieling wrote: Olivier wrote: I don't have any spare zaptel enabled system I could try this on, but I was not aware of this CHANNEL variable. Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables Maybe, I will add a line in www.voip-info.org http://www.voip-info.org to keep others (me?) from searching again. You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt There's lots of cool information there, and all of it is up to date for your version of Asterisk, unlike voip-info.org. I often wonder why nobody seems to read the docs that are included with Asterisk. Web and/or context-searchable documentation will ALWAYS win out over a somewhat loose collection of text files. That's basic UI psychology 101. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
A machine with more than one default gateway is a VERY special case (used for load-balancing or possibly failover). Most systems will not allow it. I mean... logically, it's odd. Default means when not applied to any other special rule, I choose this one.Not this two. Not this three. This one. It used to be called the Gateway of Last Resort. Last being final and not penultimate. With that being said, if you somehow manage to get by the internal consistency checks and more than one interface (and by interface, I also mean alias, as those are 'virtual interfaces') matches the default gateway, your machine is misconfigured and internet traffic will not properly flow. I know you're just the messenger here, and it's not your fault. But the message is wrong. Ekiga has tried to solve a problem (that of determining a 'best path' for SIP to allow data flow in a NAT or filtered scenario) using poorly thought-out logic. While there may be any number of SIP proxies out there (SER is one of them, and I know that's what the Ekiga service uses) that might be able to handle a mistake on the client side with ease and grace, there's no guarantee that they all will, and assuming they will simply because your test environment allows it is lazy. The RFCs are there for a reason. All SIP forking is UAS territory. Not UAC territory. N. Brian J. Murrell wrote: On Fri, 2008-09-26 at 08:43 +0100, Grey Man wrote: It's not particularly difficult to determine the best IP address for a piece of client software to use. Oh? Check the local machines default gateway, apply the subnet mask and then compare it against all the local IP's. Yeah? And if more than one matches? Then what? Have you read the whole thread here? b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
Brian J. Murrell wrote: On Fri, 2008-09-26 at 10:16 -0400, SIP wrote: The RFCs are there for a reason. All SIP forking is UAS territory. Not UAC territory. From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras asks: I repeat, Ekiga is doing something perfectly legal. The real question is why does Asterisk think it is the same request when the from tag is different ? b. Oh yes. It's perfectly legal. It's also a) NOT SIP forking, b) Lazy, and c) Poorly designed. Sending multiple requests and hoping and praying that the recipient will ignore two of them (it will NOT in many cases -- specifically set out by the RFC -- see MESSAGE) because the tag is different doesn't make it any less poorly designed just because it's not specifically written that it can't be done. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
My thoughts are that to do parallel requests from every IP address on the machine is extremely weird behaviour. How would any server know which to respond to? SIP forking is supposed to send requests to multiple different destinations (or fork mid-stream to send to different destinations). Sending from multiple different points of origin doesn't make any sense at all in either a logical or rational fashion. What's it supposed to accomplish? N. Brian J. Murrell wrote: So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't work. I am told by the ekiga devs in http://bugzilla.gnome.org/show_bug.cgi?id=553595 and http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is that Asterisk does not support SIP forking. The issue is that I have multiple addresses on my workstation: 2: eth0: BROADCAST,MULTICAST,UP,LOWER_UP mtu 1500 qdisc pfifo_fast qlen 1000 link/ether xx:xx:xx:xx:xx:xx brd ff:ff:ff:ff:ff:ff inet 10.75.22.1/24 brd 10.75.22.255 scope global eth0 inet 10.75.22.101/24 brd 10.75.22.255 scope global secondary eth0:1 So when ekiga (3.0) tries to place a call through Asterisk it in fact does parallel requests from all addresses. This is what appears to confuse Asterisk. Please see the above tickets for more details. Thots? b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
That strikes me as being careless and unreliable. Call me a purist, but I'm of the opinion that you should KNOW which interface to use based on which interface is registered and choose ONE interface based on the rules you've established during registration. What happens if you want to ensure that data goes across a VPN (in order to encrypt your VoIP communications) instead of the public internet? Or if you want to ensure a particular route based on why you created your multiple interfaces in the first place? That takes all the logic out of the equation and just says, Here's a bunch of packets. Figure out what to do with them. I'll be waiting for your response. There's a reason routing rules exist and mature services allow you to control the interface from which it originates. N. Brian J. Murrell wrote: On Thu, 2008-09-25 at 14:56 -0400, SIP wrote: Sending from multiple different points of origin doesn't make any sense at all in either a logical or rational fashion. What's it supposed to accomplish? It seems to be a shot-gun approach to making a SIP connection. The assumption being I suppose that one or more of the IP aliases will fail for whatever reason (policy routing, filtering, etc.), so just try them all, and use the first one to make a completion and drop the others. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
Alex Balashov wrote: You need to define what you mean by SIP forking. There are many things people mean by that. They are usually one of: 1) Call branching (proxies do this). 2) Parallel but distinct call legs managed by a UAC (this is what Asterisk does when you Dial(SIP/exten1SIP/exten2SIP/exten3,...)). Exactly. These are all endpoint or middlepoint things. SIP forking is never an original starting point thing. That's just WEIRD. You fork to hit multiple endpoints simultaneously. Not one endpoint from multiple starting points. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming MoH on 1.4
Olivier wrote: Hi, A somehow related question, is broadcasting streaming music as music on hold, submitted to any licencing fee ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Unless you wrote and performed the recording of the music yourself (or commissioned it for your music on hold), then yes. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle
It's common sense. Using all iLBC, I can't seem to get 100 simutaneous calls on my AMD 486 dx2/66. I don't get it! ;) N. Eric ManxPower Wieling wrote: Where did you hear this? Shaun Wingrin wrote: I have heard it said that, Asterisk falls over at 100 simultaneous calls. Is this true? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Append String to CIDNAME
Hi Sean,
Worked like a charm, thanks so much for the help!
On Sat, Sep 13, 2008 at 6:48 AM, Sean Bright [EMAIL PROTECTED] wrote:
Sip Support wrote:
exten = s,1,Set(CALLERID(name)=${CALLERIDNAME} AppropriateTag)
Try:
exten = s,1,Set(CALLERID(name)=${CALLERID(name)} AppropriateTag)
--
Sean Bright
[EMAIL PROTECTED]
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Re: [asterisk-users] Which internet phone protocol best to choose
Really? I thought both IAX and SIP are, at 3 characters apiece, equally short. However, if you get into IAX2, then yes... SIP is definitely a shorter answer. N. Alex Balashov wrote: The short answer is SIP. Stefan Gofferje wrote: http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients Terve, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
