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| 00:06.05 | bigdawggi | ZX81, is your article posted anywhere? |
| 00:06.15 | ZX81 | :) |
| 00:06.20 | ZX81 | sec I'll look it up for you |
| 00:06.23 | bigdawggi | lol, thanks man |
| 00:07.11 | ZX81 | might be this: |
| 00:07.11 | ZX81 | http://www.venturevoip.com/news.php?rssid=1647 |
| 00:07.54 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 00:08.55 | bigdawggi | thanks ZX81, I should definitely be able to get some points out of that! Thanks for writing it! |
| 00:09.15 | ZX81 | :) no probs |
| 00:12.10 | *** join/#asterisk RoyK (n=roy@ip-27-3-149-91.dialup.ice.no) |
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| 00:15.17 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 00:36.30 | Katty | hai |
| 00:37.09 | khronos | There any pointers out there for what should be charged on an anual basis for phone systems for offices? |
| 00:37.26 | khronos | Say 1 to 10 users this price, 10 to 20 another up to 100 or so? |
| 00:37.34 | khronos | Any basic guide lines that I can use? |
| 00:38.09 | vodak | khronos: it depends... you can set things up on a connection basis... per minute basis. things like that |
| 00:38.39 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 00:39.57 | khronos | I mean like for something like a company that has so many phones. |
| 00:40.17 | khronos | I know that all pbx venders do this, Cisco, etc based on how many phones you have in your installation. |
| 00:41.34 | vodak | rates are dependent on where you plan to terminate.. how many minutes per month your going to comitt to ang things lik that. |
| 00:42.10 | vodak | I recommand that you get detailed stats on current calling patterns and start throwing out lines with VoIP providers to see what they offer. |
| 00:43.57 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:43.57 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 00:59.47 | khronos | Not sure if I was clear on what I was asking. |
| 00:59.47 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 01:00.00 | khronos | I'm not talking about this from a voip situation. |
| 01:00.18 | *** join/#asterisk xenonex (n=xenonex@88.204.197.225) |
| 01:00.25 | khronos | I'm talking about from a physical management situation where I've got an office with x number of phones and a phone server. |
| 01:00.37 | khronos | or two. |
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| 01:11.48 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 01:11.58 | Alpha_AI | Hello |
| 01:12.30 | Alpha_AI | hey guys ive built an application on windows that requires asterix. is there a good asterix windows build out there? |
| 01:12.40 | russellb | what's asterix? |
| 01:12.51 | Alpha_AI | asterisk |
| 01:12.53 | russellb | oh |
| 01:13.06 | russellb | asterisk 1.6 should compile in cygwin ... |
| 01:13.17 | russellb | but it seems silly |
| 01:13.21 | Alpha_AI | why? |
| 01:13.35 | russellb | i wouldn't trust a production phone system in cygwin ... |
| 01:13.42 | russellb | feel free to try, though. |
| 01:13.51 | Alpha_AI | well im only using it for voip calls. |
| 01:13.52 | russellb | you can also run it in vmware inside of windows, i guess |
| 01:13.55 | russellb | still |
| 01:14.02 | JT | why? because it's crazy |
| 01:14.21 | Alpha_AI | well i dont have another spare pc and my app is not built for linux |
| 01:14.54 | JT | how important is it that the voip works? |
| 01:15.31 | Alpha_AI | really important |
| 01:16.03 | Alpha_AI | majority of my incoming calls will be based on it |
| 01:16.56 | Alpha_AI | which one would be better for a newbie? asterixnow or trixbox? |
| 01:17.19 | russellb | so ... why did you build this on windows knowing that asterisk doesn't run on windows? |
| 01:17.41 | JT | asterix is a comic. |
| 01:17.55 | JT | the best bet is to read the book a bit |
| 01:17.55 | Alpha_AI | pronounced the same though |
| 01:17.59 | JT | ~thebook |
| 01:18.00 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:18.01 | russellb | NO |
| 01:18.06 | russellb | it is _NOT_ pronounced the same |
| 01:18.08 | JT | certainly not spelt the same |
| 01:18.17 | Alpha_AI | who cares seriously? |
| 01:18.20 | russellb | i do :-p |
| 01:18.21 | JT | we do |
| 01:18.25 | Alpha_AI | blah |
| 01:18.30 | russellb | but i work with asterisk all day every day. |
| 01:18.36 | russellb | so i'm picky. |
| 01:18.43 | Alpha_AI | blah |
| 01:18.46 | russellb | blah? |
| 01:18.49 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:18.54 | Alpha_AI | were talking about the same thing anyways |
| 01:18.57 | JT | if you can't be bothered to spell it correctly, what else will you take liberties with? |
| 01:18.58 | russellb | uh huh. |
| 01:19.13 | russellb | so what does your app do? |
| 01:19.17 | russellb | manager interface? |
| 01:19.29 | russellb | as in, does it use the asterisk manager interface |
| 01:19.58 | Alpha_AI | well the app is supposed to compile wavefiles to play through asterisk |
| 01:20.13 | Alpha_AI | well create wavefiles on the run and play them through asterix |
| 01:20.24 | Alpha_AI | asterisk |
| 01:20.29 | russellb | i see ... |
| 01:20.45 | russellb | well if you don't have another box, your best bet is to try asterisk 1.6 in cygwin |
| 01:20.47 | russellb | it works, in theory |
| 01:20.52 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 01:20.57 | Alpha_AI | do you have a link for that? |
| 01:21.16 | JT | best bet is virtualisation |
| 01:21.21 | JT | cygwin is a major hack |
| 01:21.34 | russellb | a link for cygwin? |
| 01:21.37 | *** join/#asterisk SomethingISODD (n=wno@S010600a0d1757bfb.cg.shawcable.net) |
| 01:21.49 | SomethingISODD | hello all does anyone here know about freeradius and has a second to help me out? |
| 01:21.52 | russellb | and that's true ... running linux in vmware may perform even better ... |
| 01:22.12 | JT | or xen or kvm on a processor with virtualisation :) |
| 01:22.52 | Alpha_AI | gotcha i have the asteriskwin32 PBX installed already |
| 01:23.01 | russellb | that stuff is ancient, though |
| 01:23.17 | russellb | but that's just asterisk in cygwin |
| 01:23.37 | russellb | it's basically a few years of development behind what you'd get with 1.6 |
| 01:23.47 | russellb | if it works for you, then fine |
| 01:23.56 | russellb | but you won't get any support with it ... |
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| 01:27.06 | tzafrir | shouldn't recent 1.6 build on cygwin? |
| 01:27.39 | russellb | yes |
| 01:29.58 | codefreeze | Oh, Realtime experts: I've been diving into realtime. I think all is loaded OK, but "sip show peers" only shows the few in sip.conf, not all the goodies in the realtime db... what more do I do? |
| 01:30.32 | eric2 | with realtime, it does not hold all the peers in memory |
| 01:30.39 | eric2 | they only stick around for a while |
| 01:30.47 | eric2 | they're looked up when required |
| 01:31.16 | eric2 | and as a bonus, you won't get the pulsing dial tone when you have a message either |
| 01:32.13 | codefreeze | I didn't even see the normal burst of poke_peer_s activity when I started up... |
| 01:32.59 | file | codefreeze: they aren't loaded until they have to be... so until a call comes in, until you do a sip show peer <name> load, and even then they disappear afterwards unless you enable caching |
| 01:33.14 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:34.32 | codefreeze | Hmmmmm... then something's not right. I'll review the setup. |
| 01:34.51 | file | what is happening that you are considering it not right? |
| 01:35.14 | SomethingISODD | if anyone here knows freeradius could you please take a look and see if you can help me figure out what i am doing wrong http://www.pastebin.ca/940543 |
| 01:36.14 | codefreeze | file: picked one of the db'd peers, and did a "sip show peer :q |
| 01:36.25 | file | codefreeze: with load at the end? |
| 01:36.53 | codefreeze | er, did a "sip show peer <name>", and got "Peer <name> not found." |
| 01:37.01 | file | sip show peer <name> load |
| 01:37.51 | codefreeze | oh! ahhhhh.... ok. |
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| 01:39.38 | codefreeze | wooo, the lifetime must be less than like 10 seconds |
| 01:40.19 | codefreeze | so, there's no peer-poking of realtime peers? At all? |
| 01:40.26 | file | unless cached. |
| 01:40.46 | file | if caching is not enabled the peer is destroyed after usage |
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| 02:11.15 | jameswf-home | so who has grabbed the new druid |
| 02:13.31 | C4away | what the heck is wrong with this statement in a php-agi script: |
| 02:13.33 | C4away | $in = fopen("php://stdin","r"); |
| 02:13.47 | C4away | it is claiming it as an unexpected T_VARIABLE |
| 02:15.41 | lvl- | forgot the ; on the previous line? ;) |
| 02:15.50 | C4away | damn, you are good |
| 02:15.59 | jameswf-home | yeah sort of need context |
| 02:16.03 | C4away | lol |
| 02:16.14 | C4away | still new at this agi stuff |
| 02:16.31 | C4away | the little things are what trip you up it seems |
| 02:16.40 | lvl- | hehe |
| 02:17.00 | jameswf-home | rule of thumb for php is the error is usualy 1 line before the actual error |
| 02:17.18 | C4away | although with any ; language 99% of the time it is a forgotten ; somewhere in the code |
| 02:17.27 | C4away | and yes, that makes sense |
| 02:20.47 | high-rez | Anyone willing to recommend 802.11 SIP devices ? |
| 02:21.12 | ReDNeQ | how many you planning |
| 02:21.24 | high-rez | .. me? |
| 02:21.30 | ReDNeQ | yes |
| 02:21.35 | high-rez | Just one, for my wife. :) |
| 02:21.44 | ReDNeQ | any other wireless dev's |
| 02:22.17 | high-rez | laptops on the net same network (two), not much usage from them though |
| 02:22.49 | ReDNeQ | 108? N G ? |
| 02:22.53 | high-rez | G |
| 02:23.00 | ReDNeQ | brand? |
| 02:23.18 | high-rez | Buffalo (dd-wrt firmware) |
| 02:23.23 | ReDNeQ | ahh |
| 02:23.43 | ReDNeQ | but for small home office, wifi works fine |
| 02:23.53 | ReDNeQ | i mean its the novelty at least what it sounds like |
| 02:24.04 | jameswf-home | no wifi phones are crap reguardless of size |
| 02:24.13 | high-rez | right now she's using some crappy 5.8ghz wireless phone plugged into a sipura ata |
| 02:24.37 | high-rez | but that thing bit the dust and i'd like to just get rid of the middle man all together. |
| 02:24.40 | jameswf-home | we have been through a dozen wifi phones all bad |
| 02:25.08 | high-rez | james: Suckage :( In what ways have the been bad? |
| 02:25.36 | jameswf-home | unreliable poor sound |
| 02:26.01 | jameswf-home | all look neat but thats as far as good things go |
| 02:26.18 | jameswf-home | aastra with dect is a god send |
| 02:26.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:27.26 | high-rez | yeah i tink that might be overkill for what I want to do ;) |
| 02:28.18 | *** join/#asterisk Patrickz_ (n=patrickz@61-90-163-149.static.asianet.co.th) |
| 02:28.19 | ReDNeQ | high-rez: go with one of the Cheap'os |
| 02:28.23 | ReDNeQ | what you got to loose |
| 02:28.29 | ReDNeQ | its for novelty, play, learn |
| 02:28.42 | Patrickz_ | hello all |
| 02:29.00 | ReDNeQ | nothing and i mean nothing in this field is or will be good/god/perfect till more commercial interests start using/deploying it |
| 02:33.04 | *** join/#asterisk chendy (n=chendy@202.122.97.200) |
| 02:33.40 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 02:35.23 | bigdawggi | we use the hitachi 5000 Wireless IP phone, it's been good for us, just a *bit* $$$$ |
| 02:40.19 | jameswf-home | drmessano: ping |
| 02:43.35 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 02:45.57 | fujin | Anyone here using multiple queues with individual queue weightings? |
| 02:46.43 | fujin | I'm seeing some odd behaviour, namely; when two queues with the same weighting have calls waiting in them, they aren't presented in the 'longest waiting' order |
| 02:46.49 | fujin | they're presented randomly, it seems |
| 02:47.12 | fujin | like for example I have a queue called residential and a queue called business, with a person waiting in the residential queue for 1~min if a person joins the business queue, the business one is answered first |
| 02:47.16 | fujin | for some odd reason |
| 02:47.43 | ectospasm | and they're the same priority? |
| 02:47.47 | fujin | yep |
| 02:47.51 | ectospasm | are you sure? |
| 02:47.52 | fujin | the wrong one gets answered first |
| 02:47.54 | fujin | absolutely |
| 02:48.09 | fujin | I'm doing some rigorous testing to ensure that it's going to work perfectly in production |
| 02:48.12 | fujin | so this is a show stopper |
| 02:48.31 | ectospasm | so you're saying if the queue situation was reversed, a business caller in the same queue for 1min gets answered after a new residential caller? |
| 02:49.11 | ectospasm | well, my instinct would be to combine all equivalent priority queues, such that all queues have different priority levels |
| 02:49.27 | fujin | not applicable |
| 02:49.31 | fujin | you can't report based on a blob queue. |
| 02:50.03 | ectospasm | are these different DIDs? |
| 02:50.10 | fujin | yes |
| 02:50.34 | ectospasm | well, then report on which DID the call came in on, and separate them with your analysis software |
| 02:50.41 | ectospasm | that's my guess, anyway |
| 02:50.55 | ectospasm | I haven't dealt with queues since 1.0 |
| 02:53.21 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:54.52 | fujin | ectospasm: bad news, for the business/residential queue, I do caller-id checking out of a database to pump them into the right queues |
| 02:54.57 | outtolunc | bets it is probably as simple as B comes before R <G> |
| 02:54.57 | fujin | which effectively makes the called-DID moot |
| 02:55.03 | fujin | Yep. |
| 02:55.06 | fujin | I'm thinking the same thing |
| 02:56.16 | ectospasm | ...which would mean different priorities |
| 02:56.32 | fujin | no, it'd mean different queues |
| 02:56.42 | fujin | they're the same weighting, just in different queues for reporting |
| 02:56.44 | *** join/#asterisk _ShrikE-LT (n=_ShrikE-@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:57.05 | ectospasm | I dunno... |
| 02:58.02 | fujin | asterisk needs to be able to identify that calls in queues with the same weighting should be delivered based on holdtime, not just randomly |
| 02:59.56 | outtolunc | part of the issue is that to be able to determine a weight across queues, it last to look at all queues, all agents, then loop again to see if that queue and that agent are still available, if not, start over.. (at least that was how it was years ago) |
| 03:00.16 | outtolunc | i used to refer to try_calling as 'hunter-killer method' |
| 03:01.03 | fujin | so is the fact that asterisk just a,b,c,d.. delivers calls to the same queue weighting a known issue, or? |
| 03:01.55 | outtolunc | i have no clue what is/isn't *current* for queues, let alone *known* issues |
| 03:02.10 | outtolunc | (meaning i left using asterisk queues years ago) |
| 03:02.24 | outtolunc | between that and the deadlocks.. <G> |
| 03:02.28 | outtolunc | BUT.. |
| 03:02.41 | outtolunc | i've heard fairly good reports lately |
| 03:04.46 | outtolunc | .. /* traverse all defined queues which have calls waiting and contain this member |
| 03:04.46 | outtolunc | <PROTECTED> |
| 03:06.24 | outtolunc | note i'm just reading this for the first time in years... but it if is comparing weights, shouldn't the weights be weighted? (1++) and biggest when done wins? |
| 03:09.36 | *** join/#asterisk ph0ne (n=ph0ne@206-248-164-27.dsl.teksavvy.com) |
| 03:10.21 | *** join/#asterisk ccvp (n=axe@user-24-214-126-81.knology.net) |
| 03:11.46 | *** join/#asterisk PepOSX (n=angeldav@190.72.148.27) |
| 03:13.43 | outtolunc | hmm |
| 03:14.32 | tengulre | I Write a Asterisk Manager with delphi , If you like pls download here http://www.megacomm.cn/download/AstManager.zip |
| 03:15.09 | outtolunc | cool, thanks |
| 03:15.15 | tengulre | It can write custom IVR script with this tool, very easy! |
| 03:15.22 | tengulre | monitor Status. |
| 03:15.34 | outtolunc | no source code? |
| 03:15.55 | tengulre | but first , you must modify manager.conf in your asterisk paltform. |
| 03:16.25 | tengulre | outtolunc: sorry! many days later ,I will open it! |
| 03:16.38 | outtolunc | sweet, look forward to it |
| 03:16.55 | jameswf-home | open sores? eww |
| 03:17.49 | tengulre | If you have problem, pls email me: tengulre#126.com s/#/@, and subject : AstManger.... |
| 03:17.52 | *** join/#asterisk ahbritto (n=guest@adsl-68-125-197-181.dsl.pltn13.pacbell.net) |
| 03:18.03 | tengulre | msn: tengulre#hotmail.com s/#/@ |
| 03:18.29 | tengulre | I will provide service in 24x7 ! :) |
| 03:22.17 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 03:24.59 | znoG_ | anyone know of a way to find out the SIP client's IP address from an AGI script? |
| 03:25.04 | znoG_ | or from Asterisk itself |
| 03:25.12 | znoG_ | i could pass it as a variable to AGI |
| 03:25.35 | bigdawggi | asterisk -rx "sip show peers" | grep "<peername here>" |
| 03:26.49 | znoG_ | eek, thats pretty ugly... anything better? :) |
| 03:27.20 | bigdawggi | or something to that effect with AGI, using the "exec" |
| 03:27.23 | bigdawggi | lol, true |
| 03:27.24 | bigdawggi | :) |
| 03:32.10 | Katty | ello |
| 03:35.28 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 03:35.31 | *** join/#asterisk sergey (n=sergey@sergey.iks.ru) |
| 03:44.58 | Katty | so quiet. |
| 03:45.06 | Katty | jbot: mew? |
| 03:45.06 | jbot | A MIME mail reader for Emacs/XEmacs. URL: http://www.mew.org/ |
| 03:45.13 | Katty | jbot: incorrect! |
| 03:46.14 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 03:46.28 | ph0ne | Hi Katty how are you this evening? |
| 03:47.21 | ph0ne | I'm just lurking around trying to learn something |
| 03:48.04 | Katty | im good |
| 03:48.19 | Katty | plotting naptime. thinking about 911 ideas. |
| 03:48.36 | Katty | currently stuck on the how do you get a phone call into meetme. |
| 03:48.39 | Katty | when the server initates it |
| 03:48.41 | ph0ne | wow thats nice |
| 03:49.35 | Katty | it would have to be a .call file |
| 03:49.50 | Katty | i can't think of any other way |
| 03:49.59 | ph0ne | sorry that I can't help |
| 03:50.09 | Katty | i don't need it (= |
| 03:50.21 | Katty | just a matter of wrapping my brain around the notion |
| 03:50.43 | ph0ne | puzzle |
| 03:50.50 | Katty | i like a good puzzle |
| 03:51.18 | ph0ne | ya I love problems |
| 03:51.25 | Katty | i hate problems. |
| 03:52.20 | ph0ne | what type of OS do you run? |
| 03:53.35 | Katty | debian |
| 03:53.38 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:53.38 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:54.04 | ph0ne | I used that many years ago |
| 03:54.11 | Katty | hi russellb (= |
| 03:54.33 | russellb | Katty: greetings |
| 03:56.10 | Katty | jbot: flash |
| 03:56.17 | Katty | jbot: you are so useless sometimes |
| 03:56.17 | jbot | Katty: I think you lost me on that one |
| 03:56.19 | GiantPickle | Hello all. I have a weird issue with one of my extensions. When I do an 'sip show peers' my extention 106 shows up as host address equal to that of my router... not the actual ip that the phone has been configured to pull. Can anyone beging to tell me where I need to dig to figure out what might be going on there? |
| 03:56.42 | Nasra | any outhere that speak spanish running Asterisk? |
| 03:57.35 | jameswf-home | jbot: flash jbot |
| 03:57.36 | jbot | ACTION starts a firmware update on jbot, then pulls the plug halfway through |
| 03:57.51 | jameswf-home | Bricked!! |
| 03:57.54 | Nasra | pm me please...thanks |
| 03:58.52 | Katty | is ${BRIDGEPEER} a real variable? |
| 04:01.43 | fujin | yes |
| 04:01.46 | fujin | it certainly is |
| 04:02.03 | Katty | what sort of... |
| 04:02.04 | Katty | response |
| 04:02.07 | fujin | russellb: I don't suppose you're familiar with app_queue at all? |
| 04:02.07 | Katty | would you expect to see? |
| 04:02.14 | russellb | i know nothing |
| 04:02.14 | Katty | ZAP/something |
| 04:02.22 | fujin | russellb: really? :( |
| 04:02.24 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 04:02.27 | fujin | Katty: something like that |
| 04:02.43 | Katty | fujin: would it be a suitable variable to do a transfer with? |
| 04:02.53 | fujin | Katty: does $phone not support transfers? |
| 04:03.11 | Katty | fujin: i'm thinking more of a... |
| 04:03.59 | Katty | fujin: $person dials 911, person connected, their extension and bridgedpeer dumped into meetme911 |
| 04:03.59 | *** join/#asterisk salzh (n=salzh@124.77.127.89) |
| 04:03.59 | Katty | fujin: in the meantime, .call files go out to other people, possible some cellphones |
| 04:03.59 | Katty | fujin: plays an audio file, then transfers them into meetme 911 as well |
| 04:04.01 | fujin | and they're all bridged into meetme911? |
| 04:04.05 | Katty | indeed. |
| 04:04.09 | fujin | that's an odd setup, but definitely possible |
| 04:04.12 | fujin | Hold just a moment. |
| 04:04.17 | Katty | i can't think of any other way to do it. |
| 04:04.39 | fujin | Katty: fwiw, ${BRIDGEPEER} = the person you are talking to |
| 04:04.48 | Katty | fabulious. |
| 04:04.54 | Katty | so some NoOp |
| 04:04.57 | Katty | hmm. |
| 04:05.03 | fujin | Katty: http://rafb.net/p/fRRrxW11.html |
| 04:05.05 | fujin | That's how I do n-way conferencing. |
| 04:05.05 | Katty | I guess if you transfer one, and not the other... |
| 04:05.15 | fujin | see nway-start |
| 04:05.32 | Katty | good lord. |
| 04:05.36 | Katty | this looks complicated. |
| 04:05.42 | fujin | Sorry - that's AEL. |
| 04:05.57 | Katty | i saw something like that up on wiki |
| 04:06.00 | fujin | ChannelRedirect(${BRIDGEPEER},dynamic-nway,${CONFNO},1); |
| 04:06.02 | Katty | but i can't follow all the variable stuffs. |
| 04:06.03 | fujin | That's the important line. |
| 04:06.17 | Katty | fujin: perhaps you can help me with this later in the week (= |
| 04:06.25 | fujin | which pretty much == redirect bridgepeer to context dynamic-nway extension CONFNO priority 1 |
| 04:06.43 | fujin | well, sure, if I'm around |
| 04:06.47 | Katty | naturally. |
| 04:06.55 | fujin | I *only* write AEL - so don't expect any ugly extensions.conf shite |
| 04:07.12 | Katty | at this point, i'm still trying to get down the ugly bits. |
| 04:07.21 | Katty | wrap my brain around them. |
| 04:07.40 | Katty | fujin: my next project will have to do with cron, and sporatic reminders. |
| 04:07.48 | fujin | ew. |
| 04:07.57 | Katty | i think it'll be quite fun, really. |
| 04:07.59 | fujin | AMI makes this very easy, fortunately |
| 04:08.04 | Katty | in a migrante torture sorta way (= |
| 04:08.05 | fujin | even .call files |
| 04:08.15 | fujin | I'd do it with AMI, because I *like* ami |
| 04:08.21 | fujin | but others might do it with .call files |
| 04:08.24 | Katty | i've never used it. |
| 04:08.33 | Katty | of course i've never used an API either |
| 04:09.12 | fujin | hehe. |
| 04:10.34 | Katty | 7 or 8 .call files would get messy. |
| 04:11.09 | Katty | i even had ideas about passing some info into smbclient |
| 04:11.19 | Katty | and blasting fun windows xp popups everywhere |
| 04:12.04 | fujin | yay! |
| 04:12.27 | Katty | echo -e "'hey everybody, bettie's daughter just called her again at work.'" | smbclient -M repeatasrequired |
| 04:12.38 | Katty | her daughter calls at least 12 times a day. |
| 04:13.13 | Katty | i can only imagine the Wonderful Things that could come of this. |
| 04:13.23 | Nugget | a decade ago when I worked at hfdirect we had an employee who would constantly sleep at his desk. |
| 04:13.42 | Nugget | so I made a web app on the intranet that anyone could visit. there was a button that would make his machine play reville.wav |
| 04:13.56 | Nugget | it was comedy gold. |
| 04:14.02 | Katty | nice. |
| 04:14.22 | Katty | i should do some sort of add sip header auto answer thing and pipe old 80s music into it |
| 04:14.35 | Katty | that'd be all sorts of fun. |
| 04:15.10 | *** join/#asterisk vicram (n=vr@bb219-74-52-30.singnet.com.sg) |
| 04:15.50 | fujin | heh |
| 04:15.53 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 04:16.01 | fujin | Just had a request from a colleague to setup an extension which 'rickrolls' people. |
| 04:16.05 | fujin | ;> |
| 04:16.39 | Katty | I'd like an extension that locks my door |
| 04:16.59 | Katty | all i need is open ssh keys |
| 04:17.02 | Katty | a stand |
| 04:17.06 | Katty | and a computer with a cdrom drive |
| 04:17.12 | Katty | ssh to machine next to door |
| 04:17.15 | Katty | eject cdrom drive |
| 04:17.20 | Katty | *poke* door locked. |
| 04:17.24 | fujin | heh. |
| 04:17.41 | Katty | that could work. right |
| 04:17.54 | Katty | guess i could tape a screwdriver onto it |
| 04:19.36 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 04:20.36 | Katty | anywho. naptime. later (= |
| 04:24.23 | vicram | hello from voiceroute |
| 04:25.03 | russellb | vicram: huh? |
| 04:25.53 | vicram | russellb: just introducing myself |
| 04:26.14 | russellb | ok, just please don't advertise in here :) |
| 04:26.49 | vicram | russellb: won't even dream of it :) |
| 04:26.55 | russellb | ok. |
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| 04:28.40 | znoG_ | anyone know if I should be able to run NoOp(${CDR(src)}); from an AGI script? |
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| 04:50.06 | vicram | anyone working on being able to call a macro from custom features in features.conf ? |
| 04:51.27 | fujin | vicram: you can already |
| 04:51.28 | fujin | I do; |
| 04:51.39 | fujin | [applicationmap] |
| 04:51.40 | fujin | nway-start => *0,caller,Macro,nway-start |
| 04:51.55 | vicram | hmm is this 1.6 |
| 04:52.43 | fujin | 4 |
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| 04:54.49 | vicram | thanks maybe i was doing something wrong |
| 04:55.37 | vicram | the "important note" in features.conf says otherwise specifically that macro does not work |
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| 05:00.21 | *** join/#asterisk kissas (n=notelo@r190-64-17-227.dialup.adsl.anteldata.net.uy) |
| 05:00.27 | kissas | hi |
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| 05:02.33 | mchou | anyone happen to have a dlink vta ATA here? |
| 05:02.53 | kissas | nop |
| 05:05.15 | vicram | fujin: asterisk actually crashes when you call an ael macro from applicationmap |
| 05:06.04 | *** join/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 05:06.24 | CrashSys | Anyone ever connected to CBeyond sip trunks? |
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| 05:09.39 | fujin | vicram: no it doesn't, I do it. |
| 05:10.25 | fujin | http://rafb.net/p/cb0DVO96.html |
| 05:10.26 | fujin | see. |
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| 05:10.40 | vicram | fujin: it does for me, maybe its something to do with all the funky ael in the macro |
| 05:10.42 | kissas | why can dial through TE210P to my pbx this number 816, but not this 86002039, and the dialplan is _58. so is not problem fromt the extensions.conf. i'm getting Message type: RELEASE COMPLETE (90) in the pri debug. |
| 05:11.14 | vicram | funjin: thanks i'll post a bug report with the coredump |
| 05:11.34 | kissas | i'm cutting the 5 with exten = _58.,1,Dial(Zap/g2/${EXTEN:1}|20|dr) |
| 05:12.10 | kissas | anyone can helpme? |
| 05:13.05 | kissas | no one? |
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| 06:06.59 | tzafrir_home | vicram, hi |
| 06:07.21 | vicram | hi |
| 06:07.38 | tzafrir_home | Good to see you guys here |
| 06:08.18 | vicram | only here to find out more about 1.6 |
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| 06:50.25 | obnauticus | If one runs a T1, and the termination that that T1 line provides is being used as a trunk on another PBX, should the link speed between the T1 PBX gateway and the Gateway which is providing termination to end users be the same as the speed of the total amount of channels * 64kbps? |
| 06:50.45 | obnauticus | + that the T1 provides. |
| 06:51.34 | obnauticus | for example, you have 23B channels that can be used on the T1 which is a total of 1472 kbps on 23 concurrent calls. Should the link between that box and the other one using it as a trunk be 1.472mbps? |
| 06:52.31 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:53.39 | jql | what kind of link? IP? |
| 06:53.43 | jql | ethernet? |
| 06:53.47 | obnauticus | on which interface? |
| 06:53.55 | obnauticus | yes ip to the pbx |
| 06:53.57 | jql | box <=> box |
| 06:54.00 | obnauticus | ip |
| 06:54.19 | jql | 1840kbps for IP |
| 06:54.23 | obnauticus | why is this? |
| 06:54.40 | jql | 64kbps audio uses 80kbps IP bandwidth due to headers |
| 06:54.46 | obnauticus | oh ya |
| 06:54.48 | obnauticus | lulz. |
| 06:55.16 | obnauticus | so |
| 06:56.24 | obnauticus | (channels*n)+(n*16) = the link speed needed for 64kbps audio. |
| 06:56.40 | obnauticus | on ip between the two... k |
| 06:56.42 | obnauticus | that makes more sense. |
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| 07:28.09 | SwK | obnauticus, just use http://www.bandcalc.com/ |
| 07:28.22 | SwK | and ULAW only uses 80K/sec at 20ms packetization |
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| 07:46.39 | marthag | anyone around who can help with a weird IAX2 problem? |
| 07:47.21 | marthag | I've upgraded from 1.2 to 1.4, and now most of the incoming IAX2 calls are being dropped. and there is no error in the log. |
| 07:47.43 | marthag | it is just looks like this: http://pastebin.com/d28f42317 |
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| 08:40.37 | tzafrir | marthag, are you trying to connect to a different asterisk server? |
| 08:40.59 | tzafrir | AUTHMETHODS : 2 - what does that mean? |
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| 08:51.08 | marthag | tzafrir: this is an incoming call from our provider |
| 08:51.26 | marthag | according to the RFC, authmethod 2 is MD5 |
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| 09:49.02 | endre | ping |
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| 09:57.51 | Dovid | when asterisk does a core dump whre does it go (/tmp?) and how do i read those files ? |
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| 10:06.23 | stimpie | Dovid, try the following places: path of binaray, directory of binary or just / |
| 10:06.38 | stimpie | Dovid, you can read the core dump with gdb |
| 10:06.53 | stimpie | just run 'gdb binary core' |
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| 10:07.59 | Dovid | stimpie: how do i find out the directory of binary |
| 10:08.33 | Dovid | ah ok. i found it |
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| 10:10.42 | Dovid | stimpie: Can you please help me with this dump ? http://pastebin.ca/940850 |
| 10:11.38 | stimpie | can you five a backtrace? (just enter 'bt' in gdb) |
| 10:11.43 | stimpie | five=give |
| 10:12.12 | Dovid | so just put in gdb and bt ? |
| 10:12.46 | Dovid | ah ok. i am learning ;) |
| 10:12.47 | stimpie | run 'gdb binary core' and in the gdb prompt type 'bt' |
| 10:13.24 | Dovid | http://pastebin.ca/940854 |
| 10:13.45 | Dovid | i know it has something to do with memory. but thats about it ;) |
| 10:21.33 | yang | I am having some trouble with zapata I think the incoming calls are distorted, now when i changed from ulaw to alaw codecs |
| 10:21.49 | yang | the zapata is euroisdn |
| 10:22.48 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:22.54 | mort_gib | Morning |
| 10:23.35 | mort_gib | -How do I see if an extension is engaged? my Snom phones provide more than one line so the second call just rings! |
| 10:25.48 | Dovid | sip show channels |
| 10:26.13 | Dovid | mort_gib: What are you trying to accomplish ? |
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| 10:26.31 | vicram | mort_gib there is no such thing as engaged many sip phone support multiple channels |
| 10:26.40 | vicram | you can try using chanisavail |
| 10:26.45 | vicram | and set limitincoming |
| 10:28.23 | KermitTheFragger | mort_gib: have you tried the call-limit option in sip.conf ? |
| 10:29.07 | KermitTheFragger | iirc you need to set it for sip busy notification to work |
| 10:30.28 | mort_gib | Well, a user wants calls forwarded to his PA when he is busy |
| 10:30.46 | mort_gib | I tried call limit in SIP it didn't seem to work |
| 10:31.02 | mort_gib | Trouble is he needs that/those additional lines |
| 10:31.35 | mort_gib | vicram: in extensions.conf ?? |
| 10:32.30 | mort_gib | It's a bugger though, one of the nice features of their current Asterisk setup is that they all have multiple lines |
| 10:32.50 | vicram | you should try sharedline |
| 10:33.21 | vicram | shared-line support is in asterisk |
| 10:38.27 | mort_gib | Thanks, I'll look at this... limitincoming was 1.2 -right?? |
| 10:39.13 | vicram | look it up i'm not sure if it was removed from 1.4 |
| 10:39.35 | vicram | i think shared-lines is the best option for your boss, secretary kind of setup |
| 10:40.07 | mort_gib | Well when bosses are involved it means trade-offs ;-) |
| 10:40.51 | vicram | http://www.asteriskpbx.com/node/48342 |
| 10:43.01 | beasty_ | anyone here use a gotoiftime in his extentions.conf ? |
| 10:45.31 | *** join/#asterisk DonAlex (n=DonAlex@mirafiori.demon.co.uk) |
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| 10:45.52 | tzafrir | Hi DonAlex |
| 10:45.55 | DonAlex | Morinig all. |
| 10:46.05 | DonAlex | tzafrir: Heya.. |
| 10:46.22 | tzafrir | Morning? a .uk-er ?? |
| 10:46.46 | DonAlex | Anyone here have any experience with Cable & wireles PRI IDSN and asterisks? |
| 10:46.58 | gego | mort_gib: I use GROUP and GROUP_COUNT to limit the lines. |
| 10:47.03 | DonAlex | tzafrir: Yeah.. I thought I explain... You know Rony Cohen? |
| 10:47.25 | tzafrir | From Atelis? |
| 10:47.43 | DonAlex | tzafrir: yes.. |
| 10:47.53 | gego | mort_gib: smth like ,GotoIf($[ ${GROUP_COUNT(${EXTEN:6})} > 1 ]?busy:free) |
| 10:48.00 | DonAlex | tzafrir: he has one of your XR3000 here we are testing for a company.. |
| 10:48.08 | marthag | beasty_: I use gotoiftime |
| 10:48.27 | DonAlex | :tza having real headaches with getting to talk to the C&W isdn and a Meridian M1 |
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| 10:50.58 | DonAlex | No one at all used Cowboy & Wireless PRI's? How about Meridian M1's? |
| 10:51.56 | tzafrir | DonAlex, try asking more specific questions |
| 10:52.15 | DonAlex | tzafrir: Sure.. one mo |
| 10:52.19 | *** join/#asterisk ManxPower (n=manxpowe@196.sub-70-223-169.myvzw.com) |
| 10:54.19 | DonAlex | I am getting this Error when trying to dialout on a C&W PRI line.. " Ext: 1 Cause: Unknown (90), class = Invalid message (e.g. parameter |
| 10:54.19 | DonAlex | out of range) (5) ]"" |
| 10:54.41 | DonAlex | Now looking at he ISDN error codes it could be CUG error.. right? |
| 10:55.10 | DonAlex | but Is that really what C&W have set up.. because they are unwilling (or incompetent not sure which) to be able to tell us. |
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| 10:59.35 | ManxPower | CUG? |
| 11:00.48 | DonAlex | Closed User Groups |
| 11:01.34 | beasty_ | marthag: can you show me how you do it ? |
| 11:01.51 | beasty_ | cause i tried several ways |
| 11:01.56 | beasty_ | but always get an error back |
| 11:02.35 | DonAlex | manx: Maybe I am wrong.. but I was told it was an ISDN error code.. by you if I remember right ;) hehe |
| 11:06.33 | ManxPower | It must be a EuroThing |
| 11:06.50 | DonAlex | ManxPower: Can't disagree with you there.. |
| 11:07.09 | tzafrir | DonAlex, can you provide a more complete trace? |
| 11:07.16 | DonAlex | ManxPower: Could even be a C&W specific thing for all i know. |
| 11:07.19 | tzafrir | pri debug span 1 |
| 11:07.21 | DonAlex | of a call? |
| 11:07.22 | DonAlex | sure.. |
| 11:07.29 | tzafrir | assuming that this is span 1 |
| 11:07.30 | DonAlex | tzafrir: pastebin ? |
| 11:07.34 | ManxPower | DonAlex: this will be complicated to fix |
| 11:07.38 | ManxPower | or even figure out the cause |
| 11:07.44 | tzafrir | right |
| 11:07.56 | DonAlex | manx: Hehe.. well I never get easy jobs if that is what you mean ? |
| 11:08.15 | *** join/#asterisk xenonex (n=xenonex@89.218.236.150) |
| 11:08.30 | ManxPower | tzafrir: Me too |
| 11:10.20 | DonAlex | Ok pasted |
| 11:13.30 | ManxPower | it would help if you told us the link |
| 11:15.09 | DonAlex | sorry phoen call |
| 11:15.21 | DonAlex | http://pastebin.com/d273e83e2 |
| 11:18.39 | DonAlex | Now there are some other interesting titbits of info I can give. |
| 11:19.11 | DonAlex | Atm the PRI is plugged into a Meridan M1 via BNC connectors. |
| 11:20.29 | DonAlex | When we plug the meridian into the XR3000 it all goes completely screwy on the PRI side and the Nortel side.. in fact totally disrupting the PRI connection completely. |
| 11:21.12 | DonAlex | course could be we just have a crappy adaptor but hey ho.. Meridian works with the C&W PRI |
| 11:24.19 | DonAlex | So anyone know anything about Closed User Groups and asterisk then? Assuming this is the correct error message |
| 11:25.35 | ManxPower | I doubt Asterisk supports CUG, but I've been wrong before. |
| 11:26.47 | DonAlex | ManxPower: really? Ahh ha... |
| 11:26.50 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
| 11:27.31 | DonAlex | ManxPower: Hmmm.. well that would be a bit of a bummer |
| 11:27.45 | jblack | http://www.telegraph.co.uk/news/main.jhtml?xml=/news/2008/03/13/wtoilet113.xml has got to be a joke |
| 11:30.19 | DonAlex | Ok all Gotta pop out for a bit.. I'll leave the channel open in case anyone has any brain waves.. |
| 11:30.21 | DonAlex | Cheers.. |
| 11:32.14 | marthag | beasty_: http://pastebin.com/d6f76a073 |
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| 11:45.11 | eusbreeves | Quick question for the experts, when using TOUCH_MONITOR is there a variable like MONITOR_EXEC that can be used to execute only if the recording was done? |
| 11:45.32 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
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| 12:00.28 | marthag | ok, this is driving me crazy. periodically, this server is starts replying with INVAL to incoming IAX calls |
| 12:00.37 | marthag | and I can't figure out why |
| 12:01.45 | jblack | marthag: I've never seen that before. |
| 12:02.06 | jblack | worksfine, worksfine, worksfine, doesntwork ? |
| 12:02.18 | beasty_ | marthag: thanks alot |
| 12:02.37 | beasty_ | marthag: can you show the other stuff to ? |
| 12:03.13 | beasty_ | if possible |
| 12:04.22 | ManxPower | marthag: what version of Asterisk? |
| 12:05.15 | marthag | 1.4.9 |
| 12:05.20 | marthag | I also tried 1.4.18 |
| 12:05.53 | marthag | I just installed this new server, replacing a 1.2 server on which the IAX stuff worked fine |
| 12:06.00 | ManxPower | 1.4.9 would be expected to have issues |
| 12:06.13 | ManxPower | They did MAJOR IAX2 updates sometimes fairly early in 1.4.x |
| 12:06.24 | marthag | 1.4.9 works fine on my other server |
| 12:06.32 | ManxPower | I don't know the exact version the changes were completed. |
| 12:09.13 | *** join/#asterisk Forai (n=Forai@70.89.49.205) |
| 12:10.54 | hi365 | how do you do or in a variable? |
| 12:11.26 | hi365 | Gotoif($[${user} = 100)]?e,1 <----- i want it read 100 or 101 or 102 |
| 12:11.31 | *** join/#asterisk Uatec (n=uatec@adsl.ntsols.com) |
| 12:11.33 | Uatec | hi there |
| 12:12.10 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 12:12.32 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 12:13.05 | eric2 | hi365: GotoIf($[$["${DIALSTATUS}"="CHANUNAVAIL"] | $["${DIALSTATUS}"="CONGESTION"]]?3:3) |
| 12:13.11 | eric2 | there's an example |
| 12:13.15 | hi365 | thanks |
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| 12:19.24 | Uatec | what the hell are "Line plant details"? |
| 12:19.45 | Uatec | i'm trying to port some BT numbers to my SIP provider and they need these... |
| 12:19.50 | Uatec | i'm rather confused |
| 12:23.01 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:25.32 | ManxPower | Uatec: never heard of it. I suspect BT just are putting up roadblocks to the porting 8-) |
| 12:25.49 | ManxPower | "plant" in telecom usually means "wiring". |
| 12:26.08 | Uatec | hmm, ok |
| 12:26.28 | Uatec | i don't know why BT care, we're just trying to port from one BT wholesaler to another, so they're not losing the number or anything |
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| 12:27.59 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:28.17 | *** part/#asterisk marthag (n=martha@61.28.184.226) |
| 12:28.34 | coppice | in telecoms plant means anything which doesn't grow |
| 12:29.52 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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| 12:32.15 | DonAlex | Hi Back again |
| 12:32.35 | DonAlex | What is the current release of Asterisk? 1.4.17 ? |
| 12:32.55 | DonAlex | Just grep-ing the source code see if I can find ANY reference to CUG's |
| 12:33.20 | DonAlex | but you know.. now I come to think of it .. If it DID support them then the message would not say Unknown Right? |
| 12:33.23 | DonAlex | *sighs* |
| 12:34.21 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
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| 12:35.39 | [TK]D-Fender | DonAlex: perhaps you should consider reading the topic.... |
| 12:36.30 | DonAlex | D-F: Hehe would that be as I entered or as I left? ;) |
| 12:36.50 | ManxPower | DonAlex: type "/topic" |
| 12:37.27 | ManxPower | DonAlex: I suspect you will have to turn off that feature in the other equipment. |
| 12:37.33 | DonAlex | ManxPower: Yeah Kinda got that.. But hang on .. is this actually going to be part of the main trunk or in a zap driver somewhere? |
| 12:37.51 | ManxPower | Huh? |
| 12:37.56 | DonAlex | ManxPower: I am not even sure why it is a 'feature' in the first place. |
| 12:38.16 | DonAlex | ManxPower: just something the Telco thinks is a good idea I suspect. |
| 12:38.23 | ManxPower | DonAlex: There are MANY "optional" features with PRI support. |
| 12:38.48 | ManxPower | libpri does not support the less common ones . |
| 12:39.01 | DonAlex | ManxPower: I am starting to realise that.. There is a second PIR ISDN link from BT and that works much nicer.. |
| 12:39.33 | DonAlex | manx: Though has a funny way of dialling out perhaps. |
| 12:39.50 | *** join/#asterisk styelz (n=yoohoo@2001:388:f000:0:0:0:0:20b) |
| 12:44.53 | DonAlex | Ok So how about some thoughts on the other end of the equation? |
| 12:45.22 | DonAlex | Anyone have any experience with Nortel Meridian M1 and connecting it up to asterisk? |
| 12:46.52 | DonAlex | System Type:M1 - 11C Cabinet |
| 12:47.08 | DonAlex | Release:4.00T |
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| 13:09.17 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:09.30 | JayTee52 | Hi all. I'm going to be installing Asterisk on another server today. I'm installing on CentOS 5.1. I'm not going to be doing virtualization so do I need to install the kernel-xen-devel package? |
| 13:09.39 | *** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 13:10.04 | tzafrir | no |
| 13:10.05 | Katty | morning |
| 13:10.13 | maszlo | can someone tell be if this is the way that call fwd is supposed to behave. The inbound call is connected to the asterisk, then asterisk called the fdw number and merges the two calls. In total there are three lines active when a call is fwd. We are running 1.2.18 looking to update this soon, but unsure if this will even change our problem. |
| 13:10.14 | JayTee52 | tzafrir, thank you |
| 13:10.14 | Dovid | morning Katty |
| 13:10.20 | Dovid | Boker Toker Tzafrir |
| 13:10.59 | ManxPower | maszlo: where is the 3rd line? |
| 13:11.22 | tzafrir | Dovid, you are in US time now? |
| 13:11.36 | ManxPower | and do you mean "FWD" as in "Call Forwarding" or "FWD, the free VoIP provider"? |
| 13:12.05 | maszlo | call foward, sorry |
| 13:12.09 | tzafrir | oh, and good $WHATEVER |
| 13:12.10 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
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| 13:12.53 | maszlo | the three lines are the real number, the call fwd number, and the caller |
| 13:13.27 | maszlo | we get use minutes on our voip lines and out mobile minutes |
| 13:13.33 | maszlo | doesn't seem correct |
| 13:18.08 | ccesario | tzafrir, hi, I applied your patch http://bugs.digium.com/view.php?id=9048 in 1.4 branch, but I don't see colors.... any suggestion ? idea ? |
| 13:18.18 | Sci_05 | anyone setup an audiocodes mp118 with asterisk before? I am having problems getting the caller ID CNAME and stutter tone working correctly |
| 13:19.02 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:20.51 | ManxPower | maszlo: it is correct for the most part |
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| 13:24.44 | maszlo | there is not a way to transfer the number to a location rather than keep the extra connection open? |
| 13:25.12 | cpm | g'morning Katty |
| 13:26.01 | Katty | cpm: allo |
| 13:26.32 | ManxPower | maszlo: You cannot move the SIP channel off your server, but you can keep the audio from being transfered thru your server, but only if you are not using NAT |
| 13:27.05 | ManxPower | maszlo: WHY is this a problem? |
| 13:28.09 | ManxPower | Sci_05: none of the 19 google hits for mp118 on the mailing list archives were helpful? |
| 13:28.53 | maszlo | we are using minutes on two of out lines for just recieveing a call |
| 13:28.58 | maszlo | via call forward |
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| 13:29.36 | ManxPower | Even if the channels were handed off the telco and no longer on your server, you would still be using those mins. |
| 13:29.38 | *** part/#asterisk airjump (n=zielonka@62.159.95.82) |
| 13:29.50 | Sci_05 | ManxPower: no, checked them all. |
| 13:30.13 | ManxPower | maszlo: How are you connecting to the telco? |
| 13:30.34 | maszlo | a verizon pri |
| 13:31.04 | ManxPower | There is something called 2BCT, but I've never heard of anyone getting it to work. |
| 13:31.14 | ManxPower | maszlo: you pay for INCOMING mins? |
| 13:31.19 | ManxPower | that is very unusual |
| 13:31.47 | maszlo | I was just seeing if this was normal, to the pri line active as well as the cell phone. |
| 13:32.01 | ManxPower | It is normal. |
| 13:32.07 | ManxPower | what cell phone are you talking about? |
| 13:32.25 | maszlo | if this is the way that the system works I can accept this, just was looking to see if during updates this could be altered |
| 13:32.32 | maszlo | there are several |
| 13:32.41 | ManxPower | not connected to asterisk there isn't. |
| 13:32.50 | maszlo | at&t sprint and verizon |
| 13:33.16 | ManxPower | You have PSTN -> PRI -> Asterisk -> PRI -> PSTN when you have a call forwarding situation |
| 13:33.42 | ManxPower | the two PRIs are the same line, BTW. |
| 13:34.14 | maszlo | okay, I saw that but I guess I was just oblivious to what was going on |
| 13:34.26 | maszlo | thank you for the help |
| 13:34.39 | ManxPower | maszlo: Look into 2BCT (search the mailing lists), but there is very little information out there about it. |
| 13:35.09 | ManxPower | That will free up the two channels that are in use on the PRI in a "call forward" (the telco sees it as a transfer) situation. |
| 13:35.51 | ManxPower | In a 2BCT setup the outgoing call will still be billed to you if you pay per min, same with the incoming call -- even though the actual channels are not active on the PRI |
| 13:36.40 | maszlo | yeah it is not that we need the channels |
| 13:36.55 | maszlo | it was basicly to save on paid minutes |
| 13:37.18 | ManxPower | *nod* That won't help you with free mobile-to-mobile stuff. |
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| 13:38.18 | ManxPower | Cell companies are starting to sell/lease equipment that lets you do real cell integration with PBXs. Sprint has one that lets you call your cellphone as an extension, as well as forward MWI, etc from the PBX to the cell phone. I don't know anything else about it. |
| 13:38.37 | ManxPower | It |
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| 13:53.43 | Uatec | hey, is there a specific way to hide your caller ID when dialing over a sip connection? |
| 13:54.02 | Uatec | i tried setting use Set(CallerID(all)=Number Withheld<>) |
| 13:54.02 | *** join/#asterisk Warp4 (n=robert@207.251.204.194) |
| 13:54.10 | Uatec | but it presented the number 088888888888888 |
| 13:54.21 | Warp4 | good morning all. is anyone running asterisk under debian etch without any problems? |
| 13:54.43 | Warp4 | getting an error when i run genzaptelconf and ztcfg |
| 13:58.04 | ManxPower | Uatec: you can't block callerid to the telco that way |
| 13:58.27 | ManxPower | first of all you want a space before the < |
| 13:58.46 | Warp4 | anyone have any idean how to fix the "Unable to open master device '/dev/zap/ctl'" error under Debian Etch? (installed asterisk and zaptel from DEB packages) |
| 13:58.50 | ManxPower | But better yet is to use the CallingPres app (I assume it works for VoIP and not just for PRI) |
| 13:59.07 | ManxPower | Warp4: you would have to talk to the DEB packager about that. |
| 13:59.18 | Uatec | you can't? why not? (i'm not being difficult, i'm just intrigued) |
| 13:59.40 | ManxPower | Uatec: it's not a valid phone number so the telco will stuff in a phone number for you. |
| 14:00.05 | Uatec | oh right |
| 14:00.26 | Uatec | well, my phone company is pretty stupid, they let me send the sip name of my phone once |
| 14:00.32 | Uatec | unfortunately they don't let me send black |
| 14:00.38 | Uatec | let's investigate this CallingPres app then |
| 14:00.54 | ManxPower | Uatec: Many will allow you to do that, but once the call leaves their network the terminating carrier will overide the name. |
| 14:01.12 | *** join/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 14:01.25 | Uatec | hmm, bummer |
| 14:01.34 | *** join/#asterisk Reputation101 (n=IceChat7@212.95.118.250) |
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| 14:02.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:02.21 | Reputation101 | hello there! does someone know a bit about callfiles and answering machine detection? |
| 14:03.06 | Uatec | hmm, i tried the SetCallerPres() app |
| 14:03.11 | Uatec | but that appeared to do nothing at all |
| 14:03.15 | Uatec | i called SetCallerPres(prohib) |
| 14:04.44 | *** join/#asterisk Badran (n=mohammed@41.235.117.183) |
| 14:04.58 | Uatec | everything i read seems to lead me to believe that callingpres only works for pri |
| 14:05.38 | ManxPower | Uatec: it might. |
| 14:05.56 | ManxPower | Check the mailing list archives to see if you can find any info about callingpres on SIP |
| 14:06.01 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:06.05 | Reputation101 | i use a callfile to contact a local channel. in the extension i dial a 'real' number and use amd() to find out if there is a human or a machine on the other line |
| 14:06.09 | ManxPower | Reputation101: Asterisk does not support answering machine detection |
| 14:06.31 | Reputation101 | but asterisk has app_amd... |
| 14:06.38 | ManxPower | AMD does support it. there is NOTHING special about call files in that situation |
| 14:06.42 | ManxPower | Reputation101: what version of Asterisk |
| 14:06.47 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-142-116.dsl.stlsmo.sbcglobal.net) |
| 14:06.53 | Reputation101 | 1.4.18 |
| 14:07.03 | Uatec | ManxPower, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+callingpres |
| 14:07.04 | Uatec | there's that |
| 14:07.08 | Uatec | but it doesn't say what that is |
| 14:07.11 | Uatec | is it an application |
| 14:07.15 | *** part/#asterisk Badran (n=mohammed@41.235.117.183) |
| 14:07.17 | Uatec | a configuration option or what |
| 14:07.24 | Uatec | it's a "setting" |
| 14:07.29 | Uatec | but i can't see what ti's a setting of |
| 14:07.30 | ManxPower | Must be new in 1.4 |
| 14:08.04 | *** join/#asterisk PepOSX (n=angeldav@190.72.148.27) |
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| 14:08.22 | LemensTS | Im having a heck of a time with a polycom 550 sip phone. I cannot get it to transfer calls. Ive set the atxfer => #, in the cli when they try it from ext 1137 i get: [Mar 13 09:02:03] NOTICE[4817]: chan_sip.c:13794 handle_request_invite: Failed to authenticate user "1137" <sip:1137@pbx.myserverhere.com>;tag=F123F14B-54F8786 |
| 14:08.23 | Uatec | hmm |
| 14:08.23 | Reputation101 | <ManxPower> i just mentioned the call files because i don't know if my problem has something to to with call files or with app_amd |
| 14:08.44 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 14:08.57 | Uatec | the annoying this is that when calling my mobile is presents Unknown |
| 14:09.07 | Uatec | but when calling my ISDN it presents 0888888888888 |
| 14:09.47 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.227.137) |
| 14:10.02 | ManxPower | did you read carefully "core show application callingpres" and callerpres? |
| 14:10.17 | ManxPower | well "core show application setcallingpres" |
| 14:10.27 | Katty | in a .call file, what does the 'waittime' refer to? |
| 14:10.52 | [TK]D-Fender | LemensTS: You shouldn't be using * to transfer calls via features.conf anyways... |
| 14:10.58 | *** join/#asterisk angryuser[A] (n=nononon@df01t2-195-36-136-82.d4.club-internet.fr) |
| 14:11.33 | Reputation101 | katty: waittime tells asterisk how long to wait for a call to be answered |
| 14:11.47 | [TK]D-Fender | Katty: Mew. |
| 14:11.54 | Uatec | i just tried, i don't appear to have the application |
| 14:13.05 | *** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net) |
| 14:13.23 | Reputation101 | is it possible that amd() does not set the variable amdstatus when there was a hangup? |
| 14:13.54 | ManxPower | Katty: analog or non-analog? |
| 14:14.08 | LemensTS | TKD-FENDER: You mean features.conf has nothing to do with it? |
| 14:14.11 | *** join/#asterisk potsboy (n=chrisg@196.212.121.210) |
| 14:14.40 | ManxPower | LemensTS: it has nothing to do with the transfer key on the phone. |
| 14:14.49 | ManxPower | it has only to do with DTMF based stuff |
| 14:15.05 | LemensTS | ManxPower: Oh its for analoug phones? |
| 14:15.16 | ManxPower | LemensTS: no, analog phones use FLASH transfers. |
| 14:15.25 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 14:15.36 | ManxPower | It's for special situations and for phones too stupid to have their own transfer features |
| 14:16.06 | ManxPower | I will never understand the fetish people have for DTMF transfers. |
| 14:16.27 | LemensTS | ManxPower: Ok I see. I tried it with the transfer key, I get: [Mar 13 09:12:35] NOTICE[4817]: chan_sip.c:13794 handle_request_invite: Failed to authenticate user "1137" <sip:1137@pbx.myasteriskserver.com>;tag=3C140B00-5BDC1887 |
| 14:16.47 | LemensTS | I tried to transfer from 1137 |
| 14:16.48 | ManxPower | so you have a [1137] section of sip.conf? |
| 14:16.54 | LemensTS | Yes |
| 14:17.43 | ManxPower | LemensTS: systems I manage do HUNDREDS of transfers per day using Polycom phones and Asterisk. It's some sort of config or design issue on your side. Unfortunately, I don't know where to suggest you look. |
| 14:18.37 | LemensTS | ManxPower: Thanks for the help, I at least know where not to look now. |
| 14:18.52 | ManxPower | LemensTS: try the mailing list archives. |
| 14:19.10 | ManxPower | prepend site:lists.digium.com in your google search |
| 14:19.14 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 14:19.40 | ManxPower | LemensTS: I assume you can make calls from phone 1137 |
| 14:19.52 | BobLutz | My Asterisk box is having trouble picking up DTMF input in the IVR dialplan. It appears if I key the DTMF too fast, Asterisk cant pick them up. Are there any suggested tweaks? |
| 14:19.56 | ManxPower | and you do NOT have more than one [1137] section? |
| 14:20.14 | LemensTS | ManxPower: correct and correct |
| 14:20.20 | ManxPower | BobLutz: I suggest you tell us what phone you are using, what lines, what protocol. |
| 14:20.29 | ManxPower | LemensTS: weird |
| 14:21.17 | Katty | fujin: http://angela.sleekgeek.org/2008/03/13/e911/ <- in simplest form. |
| 14:21.37 | BobLutz | AT&T phone, channel = zap, 1 fxo module plugged into a TDM400P |
| 14:21.55 | Katty | fujin: i'm still thinking about some popups via smbclient on that... |
| 14:22.18 | Katty | fujin: and perhaps muting people upon entry. duno yet. |
| 14:22.19 | ManxPower | BobLutz: issues with that setup are almost always audio level issues. |
| 14:22.34 | ManxPower | try increasing or decreasing your rxgain |
| 14:22.55 | BobLutz | ManxPower: Thank you ... any other keywords I might want to Google? |
| 14:22.58 | ManxPower | and whatever you do don't put in relaxdtmf |
| 14:23.16 | BobLutz | ahh! I just put that in cause I read someones blog that suggested it, lol |
| 14:23.26 | ManxPower | BobLutz: I suppose you could google audio gain levels in asterisk, or you could just play with the gains. |
| 14:23.28 | coppice | they should have a tensedtmf |
| 14:23.39 | BobLutz | ManxPower: Thanks |
| 14:23.42 | *** join/#asterisk kannan (n=kann@123.201.60.116) |
| 14:23.53 | ManxPower | BobLutz: relaxdtmf makes asterisk more tollerant of long dtmf tones that are cut into two tones by audio problems |
| 14:23.59 | JackEStorm | whats out there that can handle voice+fax+data with asterisk that uses GSM for transport? |
| 14:24.33 | ManxPower | JackEStorm: start with some good hallucinogens because what you want is a fantasy. |
| 14:24.48 | JackEStorm | I was afraid of that |
| 14:24.52 | BobLutz | lol |
| 14:25.00 | [TK]D-Fender | ManxPower++ |
| 14:25.39 | ManxPower | It doesn't matter if you mean "GSM cellular service from a provider" or "GSM codec in VoIP", the answer is the same. |
| 14:26.16 | ManxPower | JackEStorm: I would not be surprised if there is NOTHING out there from any vendor that would do what you want. |
| 14:26.52 | ManxPower | BobLutz: relaxdtmf is one of those things like DTMF transfers that many people use and almost none of them need it. |
| 14:27.50 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 14:28.03 | ManxPower | BobLutz: relax dtmf may solve the problem if one digit being seen as two of the same digits on the IVR, but it can cause two of the same dtmf digits in a row to be only seen as only one digit. |
| 14:28.41 | ManxPower | it just MASKS the issue, it doesn't solve it. Like putting make up over skin cancer. |
| 14:28.48 | ManxPower | It looks better, but you are still screwed. |
| 14:29.13 | *** join/#asterisk [VIRGOS]Ricardo (n=kvirc@189.7.85.214) |
| 14:30.49 | BobLutz | There are parts in the IVR where the caller enters their PIN or account number...say "1234", mostly it works, but other times it only picks up "124" (missing the 3) |
| 14:30.53 | coppice | i thought they fixed the bug in the DTMF decodee that caused that |
| 14:30.55 | JackEStorm | ManxPower: I just need to get phone service to this one building by noon on Saturday, and bell hasn't even done the buildout to the building yet, I could have sworn I saw a voice adapter for that awhile back (but I need data support for the alarm systems) |
| 14:31.55 | *** join/#asterisk chronos (n=fcku@189.7.85.214) |
| 14:32.03 | ManxPower | JackEStorm: never going to happen |
| 14:32.27 | ManxPower | BobLutz: sounds like a job for rxgain=2 more than you have right now. |
| 14:32.44 | ManxPower | JackEStorm: no data service either, I assume. |
| 14:32.51 | BobLutz | ManxPower: Thank you, I am going to read up on it right now |
| 14:32.55 | JackEStorm | no, it's a new building |
| 14:33.06 | JackEStorm | if I had my T's up, it would be a no brainer |
| 14:33.25 | ManxPower | Any building nearby with data service you can use? |
| 14:33.28 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:33.39 | ManxPower | Phones -> Asterisk -> WiFi -> Internet |
| 14:33.42 | JackEStorm | 2 miles away, and no LOS |
| 14:33.48 | ManxPower | JackEStorm: it sucks to be you. |
| 14:34.09 | ManxPower | I suggest you stop spending time here and start spending it with your telco sales rep. I suggest weapons. |
| 14:34.41 | ManxPower | If your client does not understand there is nothing you can do until the telco installs their lines, then it's time to find a new customer. |
| 14:36.29 | JackEStorm | I'm on the phone with my rep, and his manager, and bell build out tech. My customer understands the issue, to a point, but god knows I can't get the fire marshal to understand |
| 14:36.50 | ZPertee | anyone know how I can do a find and replace text on my files in linux |
| 14:37.04 | eric2 | use sed |
| 14:39.34 | BobLutz | ZPertee: `sed s/[old text]/[new text]/ file1 > file2` |
| 14:41.16 | ZPertee | this sed 's/<MaxRetries: 0>/<Channel: Zap/7>/g' *.call >*.call isn't working what do I need to change? |
| 14:41.27 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 14:41.44 | *** join/#asterisk Dovid (n=Dovid@bzq-79-181-139-153.red.bezeqint.net) |
| 14:42.00 | BobLutz | ZPertee: I am not a sed expert...picked it up the other day for the first time, but i have noticed you cant do "file1 > file1" |
| 14:42.11 | methods | i try to register my soft phone to asterisk with 'sip set debug' on and i get no output and failed registration... i'm simply using localhost as the registrar ... should i not be seeing output on my astconsole ? |
| 14:43.01 | draygon-w | Hello |
| 14:43.10 | draygon-w | Where can I find a good doc on allowing incoming calls with asterisk? |
| 14:43.30 | *** part/#asterisk maszlo (n=reckenro@65.223.240.146) |
| 14:43.54 | ManxPower | draygon-w: um, they are enabled by default |
| 14:43.56 | johndo | incoming calls in what way? sip, IAX, hardline? |
| 14:44.25 | *** join/#asterisk P4C0 (n=lau@200.124.22.34) |
| 14:44.48 | JenniferAkemi | draygon-w: if you haven't read the book it's a great doc. |
| 14:44.59 | JenniferAkemi | ~book |
| 14:44.59 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:45.29 | P4C0 | hello guys, I have an asterisk server and multiple clients, when a call gets in, in my extensions I answer it, and then do a multi dial for all the internal extensions, however the calling party doesn't heard the ring... I'm using r option in the my dial command: exten => s,1,Dial(SIP/2&SIP/3&SIP/1&SIP/4&SIP/5&SIP/771&SIP/772,,r) |
| 14:47.48 | methods | help |
| 14:48.02 | BobLutz | ~help |
| 14:49.06 | ManxPower | P4C0: remove the "r" option, make sure you have a valid /etc/asterisk/indications.conf |
| 14:49.31 | ManxPower | P4C0: "r" requests a FAKE ringback |
| 14:49.52 | P4C0 | ManxPower, without it i doesn't get any ringback either |
| 14:50.05 | *** join/#asterisk ph0ne (n=ph0ne@76-10-153-160.dsl.teksavvy.com) |
| 14:50.06 | P4C0 | according to doc it must be used when dialing multiple channels |
| 14:50.18 | draygon-w | with SIP johndo |
| 14:50.19 | ManxPower | P4C0: that is usually because indications.conf does not exist |
| 14:50.26 | ManxPower | P4C0: the doc is WRONG |
| 14:50.50 | P4C0 | ManxPower, i have default indications.conf |
| 14:51.16 | johndo | draygon-w: incoming from a soft/hardphone or another pbx? |
| 14:52.14 | P4C0 | ManxPower, should I need to answer the call before doing the dial to multiple channels? |
| 14:52.37 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:53.25 | P4C0 | ManxPower, still no ringback :( |
| 14:57.56 | eric2 | what's the best way to test the stability of the network and ensuring that there are not 0.5 second spots of dead air? |
| 14:58.26 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:58.26 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:58.38 | lmadsen | ~skype |
| 14:58.39 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 14:58.58 | lmadsen | darn, was hoping for that hacked channel driver link :) |
| 14:59.09 | johndo | ~fwd |
| 14:59.10 | jbot | [~fwd] Free World Dialup, created by Jeff Pulver, is a free SIP server for P2P style that does not involve the PSTN (there is a charged option for this as well though). http://www.freeworlddialup.com/ |
| 14:59.11 | kannan | ~jabber |
| 14:59.11 | jbot | hmm... jabber is at http://www.jabber.org, http://www.jabber.com, http://www.jabbercentral.com and http://www.sourceforge.net/projects/jabberpy or an instant messaging system, or jabberstudio.org and planet.jabber.org |
| 14:59.25 | johndo | ~jingle |
| 14:59.36 | lmadsen | ~book |
| 14:59.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:59.39 | lmadsen | heh |
| 14:59.50 | kannan | ~google |
| 14:59.50 | jbot | extra, extra, read all about it, google is a search engine found at http://www.google.com/ |
| 14:59.58 | lmadsen | ooo... someone added a ~buybook link :) |
| 15:00.13 | johndo | I think the book is now a little dated |
| 15:00.20 | lmadsen | there is a 2nd edition |
| 15:00.24 | lmadsen | it is based on 1.4 |
| 15:00.32 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 15:00.44 | johndo | hmm, I may have to look into that |
| 15:00.53 | lmadsen | ;) |
| 15:00.58 | lmadsen | it's twice as thick as the first book |
| 15:01.08 | johndo | really? |
| 15:01.12 | Juggie | lmadsen, i thought you were hiding off irc today! |
| 15:01.16 | lmadsen | 3 pages short of double to be exact |
| 15:01.17 | johndo | ~buybook |
| 15:01.18 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 15:01.25 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:01.30 | lmadsen | Juggie: I am, but I just needed to try and find a link, but now I'm gone again :) |
| 15:02.39 | johndo | its tempting |
| 15:02.44 | ManxPower | And everyone knows size really does matter! |
| 15:02.54 | jameswf-home | http://www.trixbox.org/forums/trixbox-forums/open-discussion/another-asterisk-gui-better-trixbox << fanboys allready on the attack :)) |
| 15:03.31 | johndo | buy 2 get 1 free offer from oriely... |
| 15:03.34 | ManxPower | Warning: Site has word "forum" in URL, you risk lowering of your IQ if you go to this site! Continue?" |
| 15:03.37 | ZPertee | how do I fix this Rejected connect attempt from x.x.x.x, request 'x@y' does not exist error? |
| 15:03.53 | cpm | ManxPower, Nice! |
| 15:03.59 | ManxPower | ZPertee: create a type=friend of [x] with a context=y entry |
| 15:04.56 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 15:05.17 | ZPertee | ManxPower: I'm trying to connect to Teliax and am getting that error. not in system? |
| 15:07.02 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 15:15.55 | *** join/#asterisk ddunavant (n=David@pool-71-163-234-27.washdc.east.verizon.net) |
| 15:17.50 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 15:17.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:20.12 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:21.15 | *** join/#asterisk xenonex (n=xenonex@89.218.237.221) |
| 15:25.27 | *** join/#asterisk mitcheloc (n=mitchelo@216.207.245.1) |
| 15:26.27 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:26.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:28.06 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:28.19 | *** join/#asterisk andrebarbosa (n=andrebar@212.13.49.67) |
| 15:28.49 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 15:29.45 | khronos | Anybody know how to use the conference feature on aastra 9133i phones? |
| 15:29.58 | *** join/#asterisk Lsodi (n=Lsodi@213.168.26.50) |
| 15:30.06 | *** part/#asterisk datachomper (n=russ@75.146.194.59) |
| 15:30.16 | khronos | Tried hitting conference button, dialing in number I want to do three way with and hitting send but that doesn't really seem to be the way to do this. |
| 15:30.35 | khronos | Can't find the way to bridge the two calls together. |
| 15:31.18 | [TK]D-Fender | khronos: Go read your phone's manual. |
| 15:36.34 | methods | can i turn off skinny ? |
| 15:36.43 | methods | i tried to uncomment the bind and port lines but it still starts up |
| 15:36.55 | mvanbaak | in modules.conf: noload => chan_skinny.so |
| 15:37.29 | methods | by default it just loads everything ? |
| 15:37.38 | mvanbaak | yes |
| 15:38.06 | methods | cool |
| 15:38.08 | methods | thanks |
| 15:38.36 | ManxPower | I hate geometry!!! |
| 15:39.03 | ManxPower | khronos: Oddly enough, you should read the manual for that phone |
| 15:39.55 | *** join/#asterisk ceres (n=ceres@ariane.digicomp.ch) |
| 15:40.50 | tzafrir | florz, here? |
| 15:49.17 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 15:51.30 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:52.34 | khronos | Here's how you do the conference with aastra phones. |
| 15:52.41 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 15:52.41 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 15:52.59 | khronos | Once you have two calls going hit line number, the conf button. |
| 15:53.24 | khronos | line 1 for example, then conf, line 2 then conf again and the two calls are now bridged. |
| 15:55.29 | jameswf-home | heh rons house http://www.youtube.com/watch?v=R6P8qssKrSw |
| 15:57.13 | methods | exten => s,1,Wait(1) |
| 15:57.17 | methods | that is extension 1 right ? |
| 15:57.25 | Katty | no |
| 15:57.28 | Katty | that's extenion s |
| 15:57.40 | Katty | exten => 1,1,DoSomething |
| 15:57.44 | Katty | that's extension one. |
| 15:57.57 | Katty | exten => 1,2,DoSomethingElse |
| 15:58.16 | jameswf-home | 1,n,Even |
| 16:00.53 | tzanger | heh |
| 16:00.59 | methods | so wtf is s ? |
| 16:01.10 | jameswf-home | start |
| 16:01.21 | jameswf-home | ~ch5 |
| 16:01.22 | jbot | Read about extensions DialPlans etc.. in Chapter 5 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 16:04.24 | *** part/#asterisk LemensTS (i=CustomGT@adsl-70-238-142-116.dsl.stlsmo.sbcglobal.net) |
| 16:05.03 | ice_croft | jameswf-home> hehe |
| 16:05.13 | ice_croft | jameswf-home> that's too suck |
| 16:05.33 | Katty | methods: i see it most often with IVRs |
| 16:05.51 | Katty | methods: exten => s,1,Answer ~ exten => s,1,Background(prettymenu) |
| 16:06.52 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 16:06.56 | Katty | methods: then just below that extensions 1 through whatever. |
| 16:07.04 | Katty | methods: which does somethign when they hit 1 |
| 16:07.27 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-124-209.se.biz.rr.com) |
| 16:07.32 | methods | to dial an extension from a sip device the extension has to also exist in sip.conf ? |
| 16:07.41 | BobLutz | Anyone know what module is needed to pick up the callers phone number? |
| 16:07.42 | Katty | oh boy. |
| 16:07.56 | Katty | jbot: thebook? |
| 16:07.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:08.02 | Katty | methods: go read some stuff. |
| 16:08.50 | florz | tzafrir: kindof |
| 16:09.33 | BCS-Satori | Does anyone haev any documentation on how set asterisk to router calls with a specific cnum or cname such as anonymous? |
| 16:09.53 | *** join/#asterisk nirz (n=nir@mail2.tikalnetworks.com) |
| 16:09.59 | BCS-Satori | wow...have* & route*, typing is bad today lol |
| 16:10.18 | *** join/#asterisk jbigB (n=jBigB@216.207.245.1) |
| 16:10.24 | nirz | hello, what's the command to convert WAV files to smaller wav format (wav49) ? |
| 16:10.43 | nirz | iv been playing with sox, but havnt got any results yet. |
| 16:10.45 | *** join/#asterisk doolph (n=doolph@201.218.103.170) |
| 16:10.47 | doolph | hi |
| 16:11.14 | doolph | anyone know how to install a program with rpath/asterisknow? there's no yum or apt-get |
| 16:11.25 | ManxPower | doolph: Try the CORRECT CHANNE |
| 16:11.34 | BobLutz | lol |
| 16:11.47 | ManxPower | BobLutz: He's been told this like 10 times |
| 16:11.48 | doolph | ah ok |
| 16:13.17 | methods | when i dial 100 though my client says user not found when extensions.conf has a 100 extension registered |
| 16:14.22 | BCS-Satori | doolph: conary |
| 16:15.01 | [TK]D-Fender | methods: "extensions" do not "register". SIP Devices "register", and this has absolutely nothing to do with your dialplan. |
| 16:15.44 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:16.05 | hmodes | contexts, extensions, and peers, oh my! |
| 16:16.53 | doolph | BCS-Satori thanks I just discovered it, thank you :) |
| 16:17.02 | BCS-Satori | doolph: no problem |
| 16:17.47 | *** join/#asterisk latebind (i=latebind@196.2.118.11) |
| 16:18.08 | BobLutz | Anyone know what module is needed to pick up the callers phone number? |
| 16:18.14 | BobLutz | ANSWER = func_callerid.so |
| 16:18.19 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 16:18.20 | latebind | hi all, has anybody used sipp load test tool? |
| 16:18.22 | BobLutz | just in case someone else runs into that |
| 16:19.26 | Kobaz | sipp works pretty well |
| 16:19.44 | *** join/#asterisk ^Migs^ (n=Migs@205.122.50.69) |
| 16:19.52 | latebind | Kobaz : really? I've been stuck on a problem the whole week |
| 16:20.11 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 16:20.14 | Kobaz | you need to get the commandline options right and give it a nice xml |
| 16:20.42 | latebind | Kobaz : the cmd options are fine I think, my problem is definitely the xml |
| 16:21.57 | latebind | Kobaz : I seem to have a problem when I try to do more than 3 dtmf tones using the <nop> tag, I just took the normal uac_pcap example and literrally copied and pasted the <nop> with the pause 3 times, and the sipp program aborts on that xml. |
| 16:22.16 | BobLutz | Is there some kind of Asterisk certification? |
| 16:23.13 | Kobaz | latebind: http://pastebin.ca/941268 http://pastebin.ca/941270 |
| 16:23.14 | Qwell | BobLutz: dcap |
| 16:23.23 | Kobaz | oh |
| 16:23.27 | Qwell | ~dcap |
| 16:23.27 | jbot | it has been said that dcap is Digium Certified Asterisk Professional. See http://www.voip-info.org/tiki-index.php?page=Asterisk+dCAP |
| 16:23.33 | Kobaz | i haven't done anything fancy other than dialing a number |
| 16:23.35 | Kobaz | but that's my setup |
| 16:23.44 | Kobaz | sipp lacks good error reporting |
| 16:24.37 | latebind | the xml is valid, but there is something I am missing with the amount of pauses allowed, because I can have a few <nop> dtmfs but when I try to add a pause after each one it breaks after about 3 pauses. the funny thing is that i can play the g711a.pcap with 8 ms pause as many times as I want, but the moment I change any of them to a dtmf pcap then sipp doesnt like it |
| 16:25.23 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 16:25.32 | latebind | is there something I am missing? I even tried to do dtmf as an info send with content type "application/dtmf-relay" and that also breaks after 3 or more |
| 16:26.06 | jbigB | Qwell, wow... that page is a bit out of date |
| 16:26.27 | Qwell | jbigB: it's a wiki - update it :) |
| 16:26.33 | jbigB | yay!!! |
| 16:26.44 | *** join/#asterisk angryuser (n=nononon@df01t2-195-36-136-82.d4.club-internet.fr) |
| 16:26.55 | jbigB | I'm in the bootcamp right now... so I'll work on it later tonight |
| 16:30.30 | Kobaz | dunno |
| 16:31.00 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:31.21 | latebind | Kobaz : what scenarios have you tried? just the defaults or did you make your own? |
| 16:32.16 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:34.38 | *** join/#asterisk jmls (n=jmls@mail.tessera.co.uk) |
| 16:34.54 | *** join/#asterisk xenonex (n=xenonex@89.218.237.221) |
| 16:34.57 | jmls | is there any way of disconnecting / logging off a manager connection from the cli ? |
| 16:35.16 | Qwell | jmls: unplug the network cable? :) |
| 16:35.33 | jmls | that would disconnect *all* manager connections .... |
| 16:35.35 | jmls | :) |
| 16:35.47 | Qwell | would it, or would it not accomplish what you wanted? ;) |
| 16:35.56 | jmls | not quite. |
| 16:36.06 | Qwell | out of scope of specifications. |
| 16:36.42 | jmls | heh |
| 16:36.45 | jmls | get that all the time |
| 16:37.46 | *** join/#asterisk Zuchmir (n=dddddd@ool-4570ab19.dyn.optonline.net) |
| 16:38.16 | Zuchmir | is there a way to "busy" a zap channel without losing existing calls? |
| 16:39.12 | [TK]D-Fender | Zuchmir: place a call against it, and don't hang up. |
| 16:40.18 | Kobaz | latebind: just number dialing down the lis |
| 16:40.19 | Kobaz | t |
| 16:40.43 | latebind | oh, ok |
| 16:41.05 | latebind | Im trying to use it to go through an ivr menu |
| 16:41.17 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 16:41.21 | latebind | and i wish i knew what pcap was |
| 16:41.45 | funxion | I'm having a problem with asterisk not accepting calls from a cisco gateway |
| 16:42.04 | bkruse | latebind: pcap? |
| 16:42.20 | bkruse | latebind: Explain the context in which it is used |
| 16:42.29 | *** join/#asterisk efort (n=efort@74-86-100-202.lx-vs.net) |
| 16:42.37 | funxion | Im using realtime had this working, the peer from the sip_buddies table was deleted and I readded same as before but now inbound calls no longer work get 407 auth required |
| 16:43.02 | Kobaz | pcap is a packet capture library |
| 16:43.02 | funxion | I've tried insecure=very in db |
| 16:43.03 | latebind | bkruse : it is a "Packet Capture (pcap)" file used by sipp testing tool. |
| 16:43.38 | bkruse | latebind: That is just a raw packet dump |
| 16:43.42 | bkruse | analyze with wireshark |
| 16:43.49 | latebind | bkruse : I dont know anything about pcaps other than they are usually captured with packet sniffers like wireshark |
| 16:44.08 | funxion | bkruse any clue? |
| 16:44.42 | bkruse | latebind: What do you want to know? Or need to know? |
| 16:44.52 | latebind | my big problem is that I dont know if it is a problem in my xml or if it is a problem with the pcap files, Im leaning toward xml since sipp just exits instantly upon analyzing the xml |
| 16:45.26 | latebind | bkruse : I just want to automate someone going through an ivr |
| 16:45.34 | ^Migs^ | http://www.dothetest.co.uk |
| 16:45.38 | bkruse | latebind: That has nothing to do with pcap........ |
| 16:46.00 | latebind | bkruse : the sipp dtmf files are saved as pcap files |
| 16:46.18 | hwt | does video work with eyebeam and asterisk? |
| 16:46.41 | hwt | i have allow: alaw, ulaw, h263, h263p, but still get "can't send video". |
| 16:46.45 | latebind | bkruse : but I tend to also think its not anything to do with the files, I think its a sipp problem |
| 16:46.47 | ManxPower | hwt: your google search did not give you an answer? |
| 16:47.10 | bkruse | latebind: You really do not make sense, I am quite positive you are not sure exactly what you want to do, or any way to do it |
| 16:47.51 | latebind | bkruse: do you know what sipp is? |
| 16:47.51 | funxion | this is weird |
| 16:48.05 | bkruse | latebind: Yes. |
| 16:48.19 | bkruse | latebind: Just use another asterisk box man, it is simple! |
| 16:48.28 | ManxPower | latebind: perhaps you could start out be explaining what you are trying to accomplish |
| 16:48.43 | funxion | if i go into my cisco and do csim start my * ext rings but if I call it from my nortel extension throught eh same gateway I get 407 auth req |
| 16:48.46 | bkruse | ManxPower: Really |
| 16:48.53 | hwt | ManxPower: nah, not really. only some stuff from 2005, which i already did. |
| 16:48.58 | latebind | ok, here is what I want to do : I want to automate someone calling asterisk and pushing 1,2,3,4,5,6,7,8,9 |
| 16:48.58 | funxion | I've tried insecure=very and yes neither worx |
| 16:49.12 | latebind | and I want to do it 3000 times\ |
| 16:50.17 | funxion | hwt try h245 |
| 16:50.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:51.19 | *** join/#asterisk CVirus (n=GoD@196.205.192.125) |
| 16:51.21 | latebind | not to get into technicalities, but I have an agi pointer in the dial plan that points to a java agi server that is an ivr. The server works fine. the problem is now with the testing and load generation phase of the project, I need a tool to simulate 3000 concurrent calls, and the normal sipp uac_pcap example doesnt cut it for me, and when I try extend it and add more dtmfs then sipp doesnt like it |
| 16:52.18 | latebind | here is the normal uac_pcap example : http://sipp.sourceforge.net/doc/uac_pcap.xml |
| 16:52.19 | ACiDV | latebind, search for pbx-test 0.1.0 from atis, require a 2nd linux server |
| 16:54.42 | _ShrikE | latebind: also look at winsip from touchstone. You do have to buy it, but is great for what you want to do. |
| 16:54.59 | funxion | is there anyone familiar with realtime sip? |
| 16:55.00 | latebind | ACiDV : thanks! |
| 16:55.30 | hwt | funxion: thanks, i will. |
| 16:55.32 | _ShrikE | funxion: what is your question? |
| 16:55.41 | funxion | if i go into my cisco and do csim start my * ext rings but if I call it from my nortel extension throught eh same gateway I get 407 auth req |
| 16:56.18 | funxion | the sip peer in sip_buddies for my pstn gateway was accidently deleted from the database |
| 16:56.28 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:56.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:56.33 | funxion | I recreated it with the same entries |
| 16:56.41 | funxion | and it works outbound but not inbound |
| 16:57.00 | _ShrikE | insecure=port,invite? |
| 16:57.06 | funxion | the weird part is if i create a call ont he gateway it will ring my * ext |
| 16:57.27 | funxion | insecure was null is null I've tried yes and very neither work |
| 16:57.37 | funxion | port is 5060 |
| 16:58.04 | _ShrikE | try insecure=port,invite? |
| 16:58.35 | funxion | um |
| 16:58.41 | funxion | not sure i understand |
| 16:58.45 | funxion | its realtime |
| 16:59.05 | funxion | so in the insecure field I should populate it with '5060,INVITE'? |
| 16:59.15 | *** join/#asterisk xenonex (n=xenonex@89.218.237.221) |
| 16:59.34 | funxion | this was working yesterday |
| 16:59.45 | funxion | same exact entry in the database |
| 16:59.50 | funxion | just recreated |
| 17:00.19 | _ShrikE | funxion: in my sip realtime implementations I just put "port,invite" exactly like that in the insecure field and it works fine. |
| 17:00.31 | _ShrikE | funxion: that is when I dont want auth |
| 17:01.03 | funxion | Ill try but I did not hav that before and it was working |
| 17:01.50 | funxion | nope there is not enough room to put port,invite |
| 17:01.55 | funxion | only poirt show up |
| 17:01.57 | funxion | trunkated |
| 17:02.13 | _ShrikE | funxion: extend the field length in your database |
| 17:02.20 | _ShrikE | funxion: what version of *? |
| 17:02.22 | funxion | it workd before |
| 17:02.36 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 17:02.40 | funxion | this is in production Im not hosing it |
| 17:03.07 | funxion | anyone else ever seen "port,invite" in the insecure field using realtime? |
| 17:03.51 | _ShrikE | funxion: try to read just a little... http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure |
| 17:04.07 | funxion | I've used the field before I know how it werx |
| 17:04.21 | funxion | I just did not need to use it yesterday when this config was working |
| 17:04.28 | funxion | why should I need to use it today |
| 17:05.09 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 17:06.02 | _ShrikE | funxion: agreed, you shouldnt.. but something obviously changed. |
| 17:07.49 | funxion | agreed but I can't seem to figure it out because I restored the db I've double checked everything and everything else work except for the gateway on node 1 |
| 17:07.56 | funxion | its a * cluster btw |
| 17:08.19 | funxion | but this problem is only happening on 1 node |
| 17:12.18 | ManxPower | Maybe you don't need it and something is screwed up |
| 17:12.53 | funxion | when i do a sip show peers I dont see the peer for the gateway |
| 17:13.10 | funxion | ut the gateway does not register |
| 17:13.25 | *** join/#asterisk adjohn (n=adjohn@p2103-ipad512marunouchi.tokyo.ocn.ne.jp) |
| 17:19.32 | znoG_ | does anyone know how I could possibly lookup the IP address of the originating call? the idea is to only allow outbound calls if the source IP of the IAX/SIP user is within the local subnet. |
| 17:19.48 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 17:20.21 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:20.28 | EmleyMoor | Is anyone available to take a call now on a number I can access via sipbroker? |
| 17:22.46 | EmleyMoor | Is sipbroker's US toll free service busy? I have just implemented it as a route option in my dialplan but it gives a circuit-busy |
| 17:22.50 | *** join/#asterisk xenonex (n=xenonex@89.218.237.221) |
| 17:23.12 | EmleyMoor | I could do with trying something else over sipbroker |
| 17:24.04 | robeph | what could possibly cause asterisk to take forever between dial and to actually send the sip invite (36 seconds at this time) on a direct to device extension |
| 17:26.12 | [TK]D-Fender | robeph: DNS lookup |
| 17:26.41 | *** join/#asterisk hillct (n=H@cpe-024-211-242-232.nc.res.rr.com) |
| 17:26.58 | hillct | Good afternoon all |
| 17:27.15 | hillct | Can anyone point me to an example dialplan fragment which forwards calls received from POTS, to a target POTS number and also reports the proper call originator caller ID information when forwarding? |
| 17:27.29 | hillct | my SIP provider is forwarding calls reporting my VOIP number in the caller ID data rather than the number of the call originator and I want to be able to provide them an example of a proper forwarding dialplan |
| 17:30.52 | TJNII | Does your VOIP provider allow you to set outgoing caller ID info? |
| 17:30.58 | TJNII | Most don't for fraud reasons. |
| 17:31.42 | hillct | nope |
| 17:32.38 | hillct | but I'm talking about automatic forwarding of incoming calls. when forwarded, for example, to my cell phone, the caller ID info, indicates my VOIP number, not the number of the caller |
| 17:32.50 | robeph | [TK]D-Fender: even if its extension is to an ip not hostname? |
| 17:33.22 | TJNII | You can't, because you provider won't let you set the callerID info. So all outgoing calls will be set to the caller ID specified by your provider. |
| 17:34.06 | hillct | I know I can't change it. I mean I want to 'suggest' that they fix what I regard as a bug |
| 17:34.35 | [TK]D-Fender | robeph: pastebin a call attempt at verbose 10, SIP debug enabled |
| 17:34.38 | hillct | they may or may not fix it but I want to provide them an example dialplan that sets the proper caller ID info (of the caller) |
| 17:35.46 | TJNII | hillct: What do you consider to be a bug? |
| 17:35.55 | hillct | it should be a matter of just setting a few vars in the dialplan, but I'm not so sure about exactly what those settings would be as I've never tried it |
| 17:36.04 | funxion | hey tk |
| 17:36.13 | funxion | I got a question for you |
| 17:36.34 | hillct | The forwarded caller ID information shouldd reflect the number of the caller, not my VOIP number (which the is the number the caller dialed) |
| 17:37.23 | hillct | TJNII: to be clear, this is not an Asterisk bug. This is a configuration errror, I want to suggest they fix. I just want to be able to provide them the fix itself |
| 17:37.25 | funxion | I have * realtime with pstn calls going through a cisco 5400 outbound calls work fine but inbound calls aren't working if I do a csim start from the gateway I can make my * exten ring but if I call through the pstn i get 407 auth req |
| 17:37.28 | TJNII | No, your voip provider doesn't allow that for fraud reasons. If the provider allowed you to set caller ID, what's to stop you from setting someone else's caller ID info and calling 911? |
| 17:37.39 | TJNII | Which is a problem. It's called "SWATting" |
| 17:38.22 | hillct | I don't care about arbitrarily setting it. I want them to automatically set it properly when calsl are forwarded |
| 17:38.26 | TJNII | It's a legal thing. The outgoing caller ID is set based off the information you've given for billing, which they can verify. |
| 17:38.54 | file | calls aren't being forwarded from their perspective... it's a brand new call if you have Asterisk in the mix |
| 17:38.58 | hillct | sure, but the problem is the data is in fact invalid |
| 17:39.00 | TJNII | Exactly |
| 17:39.16 | *** join/#asterisk mocker (n=kyle@mocker.org) |
| 17:39.20 | hillct | aH |
| 17:39.22 | mocker | ~itsp |
| 17:39.22 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 17:39.29 | mocker | ~itsplist-us |
| 17:39.29 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 17:39.44 | hillct | I see. they want to bill me for this crap. That blows |
| 17:39.45 | file | you are forwarding the call to a local extension that *may* then place an outgoing call |
| 17:40.16 | [VIRGOS]Ricardo | succes |
| 17:40.25 | TJNII | hillct: You may be able to get something to work using SIP forwarding messages like "busy here," but you'll probably have to work with your ITSP to make it reliable. |
| 17:41.23 | hillct | TJNII: regardless of whether I can convince them do do it, can you point me to a dialplan fragment that will accomplish the goal? |
| 17:41.34 | *** join/#asterisk xenonex (n=xenonex@89.218.237.221) |
| 17:41.45 | TJNII | I have no idea how to implement that. I just think it might be possible. |
| 17:41.59 | hillct | K thanks. |
| 17:44.00 | funxion | [TK]D-Fender you there |
| 17:44.03 | file | if you Dial() to a device that has a call forward on it the call will go back into the dialplan, and unless you explicitly set the Caller ID before Dialing somewhere else... the number will be preserved |
| 17:44.12 | [TK]D-Fender | funxion: Yes, I was waiting for you to actually ask something. |
| 17:44.23 | [TK]D-Fender | funxion: Am a little buried here... |
| 17:44.28 | funxion | sry |
| 17:44.41 | file | setting number is accomplished using the Set application with the CALLERID dialplan function, an example would be Set(CALLERID(num)=8005551212) |
| 17:44.41 | funxion | Im wondering why I cant call through the gateway |
| 17:45.13 | TJNII | file: That won't work for him though, because hit ITSP doesn't support setting of the caller ID info. |
| 17:45.49 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 17:46.14 | [TK]D-Fender | funxion: if it doesn't show up in sip show peers then your base config is FUBAR'd |
| 17:46.21 | file | please note that if you set fromuser in sip.conf for a peer entry this *will* not allow caller ID to properly pass unless you set sendrpid=yes in the general section and the remote side has to support receiving RPID |
| 17:46.25 | funxion | it does now |
| 17:46.28 | funxion | jsut took a little |
| 17:46.36 | file | that is all. |
| 17:47.09 | funxion | this was all working fine yesterday until someone deleted the pstn gateway entry from sip_buddies |
| 17:47.17 | mocker | Is ~itsplist-us still a pretty accurate list of the top (most reliable, hah) ITSPs? |
| 17:47.24 | mocker | Or is it outdated? |
| 17:47.32 | funxion | when I readded it exactly as it was it no longer werx |
| 17:48.03 | funxion | the weird part is it werx from csim start on the gateway but not when calling through the gateway |
| 17:49.16 | plik | mocker , did you write the into to AGI on mocker.org? |
| 17:49.23 | robeph | [TK]D-Fender: http://pastebin.ca/941397 |
| 17:49.58 | mocker | plik: Yeah, don't yell at me though. |
| 17:50.04 | robeph | all seems to follow through as normally should, but the time between dial execution / and invite is rather long imo... |
| 17:50.20 | plik | nah, I was just gonna say" Thank you" |
| 17:50.26 | plik | so Thanks :) |
| 17:50.29 | [TK]D-Fender | robeph: Oh, you think I trust your AGI? Please ge me some of that crack while you're at it... oh, and do a REAL test... |
| 17:50.29 | mocker | Heh, glad it helped someone. |
| 17:50.30 | mocker | :) |
| 17:50.35 | [TK]D-Fender | robeph: :p |
| 17:50.38 | mocker | I wrote it because I knew *jack* about AGI programming. |
| 17:50.38 | robeph | hahah |
| 17:50.39 | robeph | p |
| 17:50.44 | robeph | well i don't think thats the case |
| 17:50.53 | robeph | [TK]D-Fender: I didn't really need someone to debug that ;) |
| 17:50.57 | BobLutz | `reload chan_zap.so` in *CLI> will reload zapata.conf? (tweaking rxgain) |
| 17:51.06 | [TK]D-Fender | BobLutz: Yes |
| 17:51.09 | robeph | I just wondered what possible causes for long times between dial / invite could be |
| 17:51.10 | BobLutz | danke |
| 17:51.24 | plik | well, thanks to that I have now increased my knowledge to Jack about AGI, but it's encouraging to learn ;) |
| 17:51.26 | robeph | the agi sends the dial to asterisk before it sits for the 30 seconds |
| 17:51.36 | robeph | its on the asterisk end I presume |
| 17:51.37 | [TK]D-Fender | robeph: I see no call to dial in there... |
| 17:51.50 | robeph | ? it answers.. |
| 17:52.19 | [TK]D-Fender | robeph: NVM, I'm blind.. |
| 17:52.21 | robeph | ;p |
| 17:52.43 | [TK]D-Fender | robeph: But do try it directlyt. |
| 17:52.53 | robeph | I mean its working fine.. heh. I really just dunno what the cause of such an inordinate time wait between the dial sent to asterisk and actually sending sip invites could be |
| 17:53.15 | robeph | cos asterisk is getting the invite (if I turned on agi debug you'd see that ) |
| 17:53.20 | robeph | er getting the dial |
| 17:53.23 | mocker | plik: Heh, I need to stop being lazy and write another paper. |
| 17:53.37 | mocker | s/lazy/busy |
| 17:53.42 | robeph | I'm just guessing it may be as was said earlier, either dns lookup or maybe some rdns stuff for some reaosn |
| 17:53.52 | robeph | cos if anything is slow over there...its their dns |
| 17:54.04 | hillct | file, TJNII: yes, Set(CALLERID(num)=8005551212) would certainly do the trick, though that contains the caller ID info for the incoming call? as in this case I don't want to set it arbitrararily, but set it to the value from the incoming call |
| 17:54.26 | mocker | ~itsplist-us |
| 17:54.27 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 17:54.41 | file | like I said... unless it is modified by that, it will be preserved |
| 17:54.47 | file | you do not have to do anything. |
| 17:55.00 | *** part/#asterisk mocker (n=kyle@mocker.org) |
| 17:55.12 | hillct | so hat you're saying it they're already statically modifying it |
| 17:55.18 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 17:55.20 | hillct | er s/hat/what/ |
| 17:55.21 | *** join/#asterisk Nor2 (n=nbagley@aerospace.webpipe.net) |
| 17:55.36 | TJNII | hillct: The problem as I see it is that _anything_ you do will be ignored by the ITSP. The only way to have them forward the info is to tell their system that your system hasn't touched it. If the call gets to the dialplan you can't go that. |
| 17:55.44 | ^Migs^ | GREETINGS Nor2! |
| 17:55.54 | Nor2 | what's up Migs |
| 17:56.12 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 17:56.13 | ^Migs^ | you know, just sitting around being awesome |
| 17:56.49 | funxion | hey [TK]D-Fender pleaes take a look at http://pastebin.ca/941402 see if you see anything that shouldn't be |
| 17:56.51 | hillct | TJNII: yes, I don't have direct access to update the dialplan they use and I recognize it would need to be updated on their asterisk box. I jsut want to present both the problem and the solution ot them as a support request and see if they act on it |
| 17:56.54 | TJNII | "just sitting around being awesome" I'll have toremember that one. |
| 17:57.10 | ^Migs^ | Sorry, TJNII. That's MY thing. |
| 17:57.19 | TJNII | Heh. |
| 17:57.48 | ^Migs^ | "just sitting around being awesome" is copyright (c) 2008 Migs Enterprises LLC. All Rights Reserved. No duplication of this statement may be made without the express written permission of Migs Enterprises LLC. |
| 17:58.12 | plik | ... based on an idea previously plagarised by someone else |
| 17:58.14 | *** join/#asterisk mocker (n=kyle@mocker.org) |
| 17:58.17 | Nor2 | oh yeah... "Just sitting around being awesome" Ha |
| 18:00.02 | plik | reminds me of http://www.flickr.com/photos/sugarfreak/1715199256/ |
| 18:01.05 | *** join/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 18:01.07 | anarcat | hi |
| 18:01.18 | anarcat | i'm having problems with international dialing: it's just taking the first digit |
| 18:01.33 | mocker | Any Junction Networks users? |
| 18:01.43 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 18:01.43 | anarcat | i'm using a Grandstream budgetone ATA which was hacked (with firewall rules on the server) to have "dial as you go" working |
| 18:02.18 | TJNII | The CLI will help you greatly here. |
| 18:02.52 | anarcat | basically, it's just taking the first digit |
| 18:03.06 | anarcat | (ie. dialing 90113 instead of 90113301... |
| 18:03.37 | anarcat | i tried setting SET(TIMEOUT...) but that applies only after the path in the conetext is chosen... |
| 18:03.48 | anarcat | is there a way to specify the timeout in iax2.conf? |
| 18:04.16 | anarcat | oh wait, it's sip.conf... |
| 18:05.05 | Katty | anyone know how to make asterisk pass some stuff into a text file ? |
| 18:05.26 | Katty | echo -e "' some stuff '" | /path/to/some/file |
| 18:05.29 | Katty | would that work? |
| 18:05.33 | *** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 18:05.34 | [TK]D-Fender | anarcat: you are hitting a legit pattern match.... you should be doing early-dial on a variable len dial like that. |
| 18:05.43 | [TK]D-Fender | Katty: When? |
| 18:05.43 | BobLutz | > ? |
| 18:05.55 | Katty | [TK]D-Fender: when i dial extension 301 |
| 18:06.03 | Katty | [TK]D-Fender: i want to put some text into a text file. |
| 18:06.10 | TJNII | A Q&D AGI would do it. |
| 18:06.14 | [TK]D-Fender | Katty: System(echo blah>afile) |
| 18:06.27 | Katty | [TK]D-Fender: does cron have a 'file' |
| 18:06.27 | *** join/#asterisk |STORM| (n=nbagley@aerospace.webpipe.net) |
| 18:06.40 | anarcat | [TK]D-Fender: did you mean "shouldn't" there? |
| 18:06.40 | Katty | [TK]D-Fender: or maybe can you tell cron to execute a bash script or something |
| 18:06.54 | Katty | [TK]D-Fender: and asterisk could dump to the bash thing |
| 18:06.59 | [TK]D-Fender | anarcat: Correct... should not. oops |
| 18:07.08 | justdave | does a Polycom IP430 use the IP_500 config options instead of IP_400 for some reason? |
| 18:07.50 | anarcat | [TK]D-Fender: hum... i expected * to wait for a timeout, since the string is arbitrary lenght... |
| 18:07.50 | Katty | [TK]D-Fender: system (echo stuff > /etc/crontab |
| 18:07.55 | Katty | [TK]D-Fender: would that work, mayhaps? |
| 18:07.58 | justdave | have an IP430 phone that's trying to use the idle graphic from the IP501s, even though the IP_400 block is blank for no graphic. |
| 18:08.03 | TJNII | justdave: It might. I *think* the order os 400 < 500 < 430. |
| 18:08.07 | [TK]D-Fender | Katty: something like that... |
| 18:08.11 | Katty | [TK]D-Fender: cheers. |
| 18:08.21 | TJNII | justdave: It's been a while since I've researched the Polycoms. |
| 18:08.23 | *** join/#asterisk smash- (n=smash@c-24-21-116-224.hsd1.mn.comcast.net) |
| 18:08.28 | justdave | but the 430's screen is smaller than the 501, so only half of the graphic is showing and it looks stupid :) |
| 18:08.34 | anarcat | [TK]D-Fender: so how can i disable early-dial? |
| 18:08.36 | smash- | Hey i got a question what type of headset do i need for a polycom 501 phone |
| 18:08.50 | [TK]D-Fender | anarcat: un-hack your ATA |
| 18:08.51 | smash- | it has a telephone looking jack for it, does it need a device to lift the receiver off the hook? |
| 18:08.54 | ice_croft | u talkin bout videophone? |
| 18:09.20 | smash- | me? |
| 18:09.26 | anarcat | [TK]D-Fender: ouch, this sucks. this means that local dialing will suck... |
| 18:09.34 | [TK]D-Fender | smash-: No, doesn't require a lifter. Then again, if you want the headst to able to trigger an answer then yes. Only sueful if you are useing a wireless solution |
| 18:09.39 | anarcat | [TK]D-Fender: i can't disable it just for some extensions? |
| 18:09.47 | anarcat | or prefixes, actually |
| 18:09.49 | [TK]D-Fender | anarcat: "suck" how? Dial like the rest of us |
| 18:09.49 | ice_croft | all of u |
| 18:09.50 | ice_croft | L( |
| 18:09.55 | ice_croft | :) |
| 18:10.17 | smash- | no its mainly outbound dialing, so they can hit the send botton |
| 18:10.37 | smash- | and the phone usally goes to speaker phone so it would instead goto the headset vs. speaker |
| 18:10.39 | [TK]D-Fender | smash-: Typically you'd use it with an RJ9 amplifier like the Plantronics M-22, and an attached headset. |
| 18:10.45 | funxion | hey in * 1.2.9 what should the insecure value be for sip peers? |
| 18:10.49 | anarcat | [TK]D-Fender: well, there's going to be a long delay for regular dialing |
| 18:11.01 | [TK]D-Fender | funxion: 1.2.9 is insecure by default ;) |
| 18:11.06 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-10-32.nyc.res.rr.com) |
| 18:11.07 | smash- | Thanks TK |
| 18:11.10 | anarcat | [TK]D-Fender: i'm just expecting this device to behave like a regular phone :) |
| 18:11.13 | smash- | i think thats what i wanted to know |
| 18:11.15 | funxion | tk thats what I thought |
| 18:11.16 | ice_croft | did anybody try to watch video via sip? |
| 18:11.17 | [TK]D-Fender | anarcat: only if your dialplan sucks. |
| 18:11.21 | ^Migs^ | I think it's too rowdy in here. Everyone calm down. |
| 18:11.29 | anarcat | [TK]D-Fender: ah, maybe that's the problem... |
| 18:11.32 | smash- | then use the adapter and plug the mic/headphone jack into the RJ9 Amp.? |
| 18:11.32 | ^Migs^ | haha |
| 18:11.48 | [TK]D-Fender | anarcat: On any other sane setup I can have fast 10-11 digit dialing |
| 18:11.53 | funxion | [TK]D-Fender did you get a chance to look at my pastbin? |
| 18:12.02 | funxion | http://pastebin.ca/941402 |
| 18:12.05 | [TK]D-Fender | funxion: Didn't say anything to me... |
| 18:12.21 | funxion | I can't figure this out |
| 18:12.49 | [TK]D-Fender | smash-: The amp goes to RJ9 as well though you can get a variety of headsets for it, and some adapters, etc. |
| 18:12.50 | funxion | if I make a call from a gateway with no ani it works if I provide caller id it no longer works |
| 18:12.50 | funxion | any clues? |
| 18:13.35 | anarcat | [TK]D-Fender: i expected that... :) |
| 18:13.42 | anarcat | we're overflowing poor [TK]D-Fender here :) |
| 18:14.02 | anarcat | i pasted part of my dialplan here: http://pastebin.ca/941425 |
| 18:14.04 | anarcat | selected extracts |
| 18:14.25 | justdave | I guess it's not that surprising... the 430 is a trimmed down version of the 501, not a beefed up 401 :) |
| 18:14.27 | [TK]D-Fender | anarcat: I didn't say ASTERISK's dialplan.... |
| 18:14.34 | anarcat | oh |
| 18:14.36 | [TK]D-Fender | anarcat: your ATA's <- |
| 18:14.41 | anarcat | ouch |
| 18:14.59 | [TK]D-Fender | justdave: Neither, its a souped up IP 301 <- |
| 18:15.26 | justdave | ah, so maybe I need to override the IP_300 block in the bitmaps... |
| 18:15.33 | anarcat | so you're saying my dialplan is fine and my ATA is broken |
| 18:16.01 | anarcat | or rather, that i can't do dial as i go with (potentially) international dialing in asterisk |
| 18:16.23 | anarcat | i mean, i would have the same issue with Zap channels... they would just hang up on me on the first digit... |
| 18:17.20 | iCEBrkr | [TK]D-Fender: psst |
| 18:17.20 | alrs | how about a testimonial for the improved quality of the Digium PCI Express cards, anyone? |
| 18:17.45 | justdave | ok, nope, it was IP_400. But I have to override the <bitmaps> and not the <Animations> |
| 18:18.03 | [TK]D-Fender | justdave: That would help ;) |
| 18:18.15 | justdave | but on the IP501, it won't even show up if you don't define an Animation for it as well |
| 18:18.31 | |STORM| | HELP-- Can anybody help.. I receive these errors when I try to call out. Using 2 Grandstream GXP2000 Phones connected to the AsteriskNow server.. The phones can talk to each other but just not call out.. I have a Digium TDM400p card and Port 4 (FXO) connected to the phone line. Take a look at these errors please.. http://pastebin.com/m5cf85ad |
| 18:18.34 | justdave | guess the 430 isn't that picky :) |
| 18:19.01 | |STORM| | Both phones also produce a 503 error when calling out. |
| 18:19.02 | tzafrir | anarcat, "hang up after the first digit" - badly-configured dialplan? |
| 18:19.12 | tzafrir | incorrectly-detected DTMF? |
| 18:19.26 | ^Migs^ | how do they tell you it's a 503 error, btw? |
| 18:19.29 | ^Migs^ | |STORM| |
| 18:19.30 | smash- | hey thanks TK i got it all figured out, reading is a good ability |
| 18:19.37 | justdave | |STORM|: looks like you don't have an outbound route defined |
| 18:19.56 | |STORM| | the Grandstream phone says 503 and gives you the busy tone |
| 18:19.59 | *** part/#asterisk pkunkra (n=chris@cpe-74-73-10-32.nyc.res.rr.com) |
| 18:20.01 | iCEBrkr | [TK]D-Fender: You still experience echo these days? I thought it was a think of the past. |
| 18:20.11 | ^Migs^ | oh okay |
| 18:20.25 | justdave | |STORM|: it's in asterisk. there's no outbound route, so it's trying to send your dial string to a non-existant device/channel |
| 18:20.25 | [TK]D-Fender | |STORM|: And AsteriskNOW is not supported here, please use thier channel |
| 18:20.34 | anarcat | tzafrir: i don't know, you can see part of the dialplan there: http://pastebin.ca/941425 |
| 18:20.38 | ^Migs^ | there's an AsteriskNOW channel |
| 18:20.38 | ^Migs^ | ? |
| 18:20.46 | [TK]D-Fender | ~asterisknow |
| 18:20.47 | jbot | somebody said asterisknow was based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 18:20.49 | [TK]D-Fender | ^^^^^^^^ |
| 18:20.57 | anarcat | tzafrir: and local, fixed-length dialing works properly, it's just variable length dialing that is broken |
| 18:21.05 | ^Migs^ | what's the difference between AsteriskNOW and Asterisk? |
| 18:21.16 | [TK]D-Fender | ^Migs^: Asterisk NOW = full distro + GUI <- |
| 18:21.23 | justdave | AsteriskNOW is a linux distro that happens to include Asterisk and some other goodies in it |
| 18:21.25 | tzafrir | AsteriskNOW is a Linux Distribution that includes Asterisk |
| 18:21.27 | justdave | Asterisk is just Asterisk |
| 18:21.28 | [TK]D-Fender | ^Migs^: Add-on crap is "not our problem" |
| 18:21.30 | ^Migs^ | OH okay |
| 18:21.31 | anarcat | hehe |
| 18:22.12 | tzafrir | anarcat, you can log the detected dtmfs . Look for "dtmf" in logger.conf |
| 18:22.13 | justdave | the main reason it's hard to support in here is because AsteriskNOW's GUI does a bunch of strange things to the config files |
| 18:22.31 | justdave | so you really need people that can help you use the GUI |
| 18:22.32 | ^Migs^ | ugh |
| 18:23.11 | |STORM| | AsteriskNow is a pre-configured Asterisk Installationokay so let's pretend that I am using a Certified copy of MY own install of Asterisk... Can anyone point me to the correct config files? |
| 18:23.15 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 18:23.28 | BobLutz | ? |
| 18:23.30 | TJNII | Heh. Makes me think of when I tried to set up an IAX trunk to my friend running trixbox. Spent an hour just trying to figure out the configs it generated. |
| 18:23.40 | anarcat | tzafrir: i just added "dtmf" to my "console" logging line and i'm not actually seeing anything different |
| 18:23.49 | ^Migs^ | we'll probably end up installing Asterisk from scratch anyway |
| 18:24.02 | russellb | i hate the attitude towards GUIs in this channel ... |
| 18:24.03 | anarcat | tzafrir: i feel like the digits are sent through SIP directly |
| 18:24.20 | iCEBrkr | russellb: if the GUI's didn't suck :-D |
| 18:24.34 | ^Migs^ | russellb, it's all the Linux fanbois. They're the same ones who think vi and irssi are SUPERIOR because they are all text-based |
| 18:24.47 | tzafrir | anarcat, you need to log debug messages |
| 18:25.06 | tzafrir | IIRC |
| 18:25.12 | ^Migs^ | never mind the fact that GUIs make things 100x easier. It's all about geek cred masquerading as superior products |
| 18:25.36 | TJNII | ^Migs^: I have nothing against GIUs, but when someone has no idea what is in the config file to begin with, us telling them to check a setting is kind of pointless. |
| 18:25.41 | iCEBrkr | ^Migs^: Typically, yes. But at the moment, the Astrisks GUIs really miss the mark. |
| 18:25.49 | iCEBrkr | ^Migs^: Geeks shouldn't be doing UI design. |
| 18:25.54 | [TK]D-Fender | ^Migs^: it generates your config for you based on shit you can't readily show us and the scripts that do the generating itself can be screwed up. Who wants to spend all day debug shit that some shmuch who needs a click-and-go setup can't figure out? |
| 18:26.14 | [TK]D-Fender | ^Migs^: This is like making a "Color -by-numbers For Dummies" book. Its REDUNDANT |
| 18:26.26 | |STORM| | ^Migs^ We will end up installing Asterisk from Scratch.. funny though because you would think that someone here could point us in the right direction.. But I slipped and said AsteriskNow.. Instead of something like Ubuntu with Asterisk, etc |
| 18:26.29 | *** join/#asterisk Alexus27 (n=alexus@104-60.wlan.cpe.InfoSky.net) |
| 18:26.32 | iCEBrkr | and ON top of it.. the GUIs are designed be 'geeks' who just don't understand Asterisk |
| 18:27.18 | BobLutz | It is not that hard to open up a .conf file and change "yes" to "no" |
| 18:27.26 | [TK]D-Fender | ^Migs^: Some things are jsut fine to admin with some GUI interface. * is not a PBX. Its a telephony toolkit from which you can make a PBX. Thinking that we need to support some cookie-cutter admin system because you think thats the way things should be run is not our issue. |
| 18:27.26 | iCEBrkr | If I weren't such a lazy bastard. I'd design a GUI |
| 18:27.27 | anarcat | tzafrir: ok, well, i see a bunch of new stuff, but no digits... |
| 18:27.41 | anarcat | tzafrir: i really don't think that the problem is dtmf-detection, that is done on the ATA-side |
| 18:28.01 | anarcat | tzafrir: and [TK]D-Fender seemed to be saying that early-dialing doesn't work with variable-length extensions, is that true? |
| 18:28.07 | iCEBrkr | *tap* *tap* *tap* Testing.. 1.2.3... Is this thing on? |
| 18:28.31 | tzafrir | <iCEBrkr> *tap* *tap* *tap* Testing.. 1.2.3... Is this thing on? |
| 18:28.41 | iCEBrkr | [TK]D-Fender: CAN YOU HEAR ME!! |
| 18:28.46 | ^Migs^ | First of all, I never said that was the way things should be. |STORM| here just set up AsteriskNOW as a way to quickly test if he could get things working. Second of all, you're forgetting many people are new to Asterisk. It's not that they're "schmucks" who "need click-and-go setups." |
| 18:28.54 | ^Migs^ | They're just trying to figure it out one step at a time. |
| 18:28.55 | [TK]D-Fender | anarcat: Early dial works if there are multiple items that will hit the in-progress dial until 1 is unique & complete. Thing is that a variable length exten could be "complete" at any time. |
| 18:29.17 | justdave | but it does a good job of parsing and editing config files so you can still edit them by hand without breaking things |
| 18:29.29 | justdave | and it'll figure out what you did and reflect it in the GUI |
| 18:29.32 | ^Migs^ | and I fail to see how this is any way analogous to your "color-by-numbers for dummies" example. It's not redundancy, it's an attempt at implementing user-friendliness, even though it may be shoddy for now. |
| 18:29.42 | tzafrir | What do you mean by "early dial"? |
| 18:30.02 | [TK]D-Fender | ^Migs^: Forgetting? No, I spend untold hours a week here helping people doing it from scratch.... and its a staggering amount of people who expect and demand everything be pretty and a 2-click solutions.... |
| 18:30.19 | BobLutz | It is not that hard to open up a .conf file and change "yes" to "no" |
| 18:30.26 | ^Migs^ | I don't see why that's necessarily an unrealistic expectation. |
| 18:30.28 | iCEBrkr | ^Migs^: Well, unfortunately, most GUIS are EPIC FAIL |
| 18:30.40 | anarcat | tzafrir: this http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching |
| 18:30.43 | iCEBrkr | ie. the designers should give up |
| 18:30.46 | Alexus27 | Hi all. I'm running 1.4.18 and after reboot I hear silence while calling to analog PSTN lines. Box also doesn't hear incoming calls. No errors in console. Card is Wildcard TDM2400P. Could you please suggest what can be wrong? |
| 18:30.50 | hmodes | if my 55yo father can understand extensions.conf and run his own phone system... i fail to see the need for guis ;p |
| 18:30.50 | TJNII | ^Migs^: It isn't, and there are solutions for it. This is not the place, though. |
| 18:30.53 | iCEBrkr | all it does is just confuse people more |
| 18:31.03 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 18:31.06 | anarcat | [TK]D-Fender: from what i understand from the (lack of) documentation is that there should be a timeout in there, unless it's a clear match |
| 18:31.15 | iCEBrkr | Lack of documentation? |
| 18:31.17 | johndo | http://xkcd.com/386/ |
| 18:31.17 | iCEBrkr | Are you kidding me? |
| 18:31.17 | ^Migs^ | understood |
| 18:31.28 | justdave | |STORM|: it'll be somewhere in your extensions.conf or something referenced by it |
| 18:31.31 | justdave | assuming it writes it out |
| 18:31.35 | ^Migs^ | haha, I love that one, johndo |
| 18:31.43 | justdave | I seem to recall it storing everything in a database instead of the config files |
| 18:31.46 | iCEBrkr | I learned to configure and program my Asterisk box from voip-info.org |
| 18:31.48 | anarcat | this is a pretty good description of my problem: http://lists.digium.com/pipermail/asterisk-users/2004-June/042727.html |
| 18:31.55 | justdave | but maybe that's FreePBX |
| 18:32.00 | iCEBrkr | and what I couldn't figure out there, I either found out by trial and error or I'd ask here. |
| 18:32.09 | iCEBrkr | asking here was last ditch effort :) |
| 18:33.14 | BobLutz | russellb: Are you the maintainer of Zaptel as well? |
| 18:33.31 | hmodes | there's a reason so many people use ~thebook in here, methinks |
| 18:33.39 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 18:33.47 | hmodes | if you can't be bothered to read readily available documentation on something... |
| 18:33.51 | hmodes | you need a support contract. |
| 18:33.51 | russellb | BobLutz: not really, no |
| 18:33.57 | [TK]D-Fender | anarcat: it is a clear match, thats the issue |
| 18:34.01 | iCEBrkr | hmodes: hahah |
| 18:34.10 | russellb | BobLutz: mainly asterisk and asterisk-addons |
| 18:34.10 | [TK]D-Fender | anarcat: It isn't in clash with any OTHER exten. |
| 18:34.13 | |STORM| | [TK]D-Fender Thanks for pointing me to the Asterisknow channel. I happen to love command line interfaces, but found that I could install AsteriskNow in the fraction of the time.. to see if Asterisk is even right for our company... Little did I know that the gui Scimps on the Asterisk install and prevents it from working right... I will return once I have an Ubuntu server install containing Asterisk. |
| 18:34.33 | BobLutz | russellb: Just wondering..I saw that you announce the releases on your site |
| 18:34.47 | russellb | yeah, i usually make the releases |
| 18:34.50 | TJNII | |STORM|: It's suprisingly easy to do from scratch. Give it a shot. |
| 18:34.58 | russellb | but our hardware and driver team does all the work and decides when they should be done for the most part |
| 18:35.12 | anarcat | [TK]D-Fender: so would there be a way for me to hack my * dialplan to fix that? |
| 18:35.16 | TJNII | |STORM|: If you're not afraid of CLI's and config files, you system will probably be better for it. |
| 18:35.22 | iCEBrkr | You know, extensions.conf can be intimidating at first.. Just have to step back and take a deep breath and just dive in. |
| 18:35.31 | |STORM| | thanks TJNII |
| 18:35.32 | iCEBrkr | If you can't dedicate time to learning how all this crap works, then you probably shouldn't be doing it |
| 18:35.46 | anarcat | [TK]D-Fender: the guy in GB mentionned that he tried explicitely mentionning all the prefixes... |
| 18:35.56 | ManxPower | |STORM|: The other advantage of using a self installed, non-GUI Asterisk is that you can ask for help here. We throw the GUI users to the aligators. |
| 18:36.04 | [TK]D-Fender | anarcat: closest thing I can think of is do a "011" exten on both sides, and have * answer in an unpromted IVR to listen for the remaining digits. |
| 18:36.34 | [TK]D-Fender | anarcat: Might be workable. Your CDR's will look really bad, but you may be able to hack some more stuff in thre fo compensate. |
| 18:36.43 | iCEBrkr | ManxPower: Yeah, I don't think anyone's head has been bit off when asking how to make their dialplan work via textpad :) |
| 18:36.45 | hmodes | speaking of releases, is there a good reason i get release-announce emails before the releases are on downloads.digium.com? |
| 18:37.00 | russellb | the mirror has been a PITA recently |
| 18:37.08 | russellb | you can always get releases out of svn very easily |
| 18:37.19 | anarcat | [TK]D-Fender: thanks |
| 18:37.20 | russellb | svn co http://svn.digium.com/svn/asterisk/tags/1.4.19-rc2 for example |
| 18:37.27 | hmodes | after about the third time, i gave up on 1.6-beta releases, my attention span isn't long enough to remember to look for a tarball > 5mins after getting the announcement ;p |
| 18:37.41 | iCEBrkr | lol |
| 18:37.56 | russellb | hmodes: svn co http://svn.digium.com/svn/asterisk/tags/1.6.0-beta5 |
| 18:38.22 | hmodes | duely noted for next time :) |
| 18:38.54 | hmodes | i try to avoid svn unless i'm working with trunk tho'... |
| 18:39.39 | [TK]D-Fender | hmodes: The channel topic here is usuallyt he best place for up to date stuff. We tend to know about it before asterisk.org even gets updated. |
| 18:39.49 | anarcat | [TK]D-Fender: i tried changing the extensions... http://pastebin.ca/941465 |
| 18:40.08 | *** part/#asterisk |STORM| (n=nbagley@aerospace.webpipe.net) |
| 18:40.13 | anarcat | now it *does* seem like there shouldn't be a clear mathc, but it's still matching the 9011. :( |
| 18:40.29 | hmodes | well, this is more a commentary about sending out release notices that don't include links to the release ;p |
| 18:40.51 | *** join/#asterisk |STORM| (n=nbagley@aerospace.webpipe.net) |
| 18:40.54 | [TK]D-Fender | anarcat: Where in there should I be looking? I don't see anything like what I described. |
| 18:41.26 | anarcat | [TK]D-Fender: lines 5 and 6 |
| 18:41.38 | [TK]D-Fender | anarcat: and you didn't provide any backup for your failed attempt |
| 18:41.40 | anarcat | [TK]D-Fender: i added the 901133 |
| 18:41.43 | ^Migs^ | pretty risky coming back here, |STORM| |
| 18:42.02 | [TK]D-Fender | lol |
| 18:42.34 | iCEBrkr | [TK]D-Fender: damnit, stop ignoring me. |
| 18:42.36 | anarcat | [TK]D-Fender: hmm... failed attempt? |
| 18:42.43 | anonymouz666 | anyone know why I should answer an analogic channel, for after that, I detected the busy tone? |
| 18:43.02 | anonymouz666 | must be always that way? dsp can't detect the busy tone if I don't answer it? |
| 18:43.09 | hmodes | and now that i've had my say and reminded myself, time to go upgrade to beta5 ;p |
| 18:43.22 | [TK]D-Fender | iCEBrkr: Sorry, missed it all. Nope... I don't get echo... |
| 18:43.27 | *** part/#asterisk |STORM| (n=nbagley@aerospace.webpipe.net) |
| 18:43.44 | [TK]D-Fender | anarcat: SIP debug & CLI of the attempt that doesn't work the way you expected it to... |
| 18:43.58 | iCEBrkr | hehe |
| 18:44.16 | robeph | [TK]D-Fender: I made a seperate context that simply dials that device directly, still, 30-40 seconds after sending the dial to asterisk before asterisk responds with invite. so AGI is not the factor, any other plausible ideas come to mind? |
| 18:44.20 | anarcat | [TK]D-Fender: i'm sorry, you mean that you want the CLI trace of the attempt? |
| 18:44.20 | iCEBrkr | [TK]D-Fender: We have a location that's complaining of echo.. I haven't heard complaints of echo in years. |
| 18:44.38 | [TK]D-Fender | robeph: check all of your networking... |
| 18:44.42 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 18:44.44 | Kernel_Core | hi all |
| 18:45.12 | Kernel_Core | I have a SIP connection between my asterisk and my cisco 2600 with 4 FXO ports , |
| 18:45.44 | Kernel_Core | I can register my eyebeam to Asterisk and make a call to my Cisco2600 and PSTN... |
| 18:45.55 | Kernel_Core | but I have a problem ... I disconnects after 2 minutes |
| 18:46.06 | Kernel_Core | when I issue , show channels |
| 18:46.09 | Qwell | Cisco actually has a 2600 model number? |
| 18:46.16 | Qwell | that's silly |
| 18:46.33 | BobLutz | teehee |
| 18:46.35 | Kernel_Core | I get : |
| 18:46.36 | Kernel_Core | SIP/cisco-home-081e4 44532273@blocked:1 Down AppDial((Outgoing Line)) |
| 18:46.36 | Kernel_Core | SIP/saber-081e08c8 44532273@saber-con:1 Ring Dial(SIP/44532273@cisco-home) |
| 18:47.09 | Kernel_Core | imagine I get DOWN and RING in asterisk , but I can speak , and channel stablished |
| 18:47.11 | znoG_ | does anyone know how I could possibly lookup the IP address of the originating call? the idea is to only allow outbound calls if the source IP of the IAX/SIP user is within the local subnet. |
| 18:47.39 | [TK]D-Fender | Kernel_Core: pastebin "core show channels concise" |
| 18:47.47 | iCEBrkr | znoG_: Isn't that something you set in sip.conf? |
| 18:47.57 | iCEBrkr | znoG_: Don't they have to register before making calls? |
| 18:48.04 | johndo | znoG_: wouldn't it make sense to put the phones on that subnet in a context, and then control that context? |
| 18:48.17 | Kernel_Core | @Qwell: yea , it is 2600 |
| 18:48.20 | [TK]D-Fender | Qwell: its an ancient series.... |
| 18:48.31 | iCEBrkr | znoG_: I guess I don't understand why/what you're trying to do/prevent |
| 18:48.31 | Qwell | just a funny model number, if it's a telephony device |
| 18:48.33 | *** part/#asterisk jmls (n=jmls@mail.tessera.co.uk) |
| 18:48.36 | hmodes | sounds like you're talking over early media somehow kernel |
| 18:48.43 | iCEBrkr | Qwell: haha, 2600. makes PERRRRfect sense! |
| 18:48.44 | [TK]D-Fender | Qwell: My co used one for a 64k leased line when I started here about 13 years ago |
| 18:48.58 | Kernel_Core | @Qwell: it is 2600 with four port FXO |
| 18:49.20 | iCEBrkr | You should get a Capt'n Crunch whistle with it! |
| 18:49.27 | hmodes | and the 2600s were never meant to do voip, it's a hack at best ;p |
| 18:49.29 | Kernel_Core | @Qwell: and in my CDR , I get " NO ANSWER " |
| 18:49.31 | tzafrir | 2600 is not a router? What a shame |
| 18:49.34 | hmodes | which makes the model number even more ridiculous |
| 18:49.54 | iCEBrkr | hmodes: I still don't think you get the joke |
| 18:50.16 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 18:50.17 | Kernel_Core | tzafrir: it is a router but it has FXO WIC , and with specific IOS it can do VOIP |
| 18:50.33 | hmodes | i don't think he does either :) |
| 18:51.02 | iCEBrkr | If I were to have one, I'd have to put a Capt'n Crunch sticker on it or something! |
| 18:51.17 | anarcat | [TK]D-Fender: i figured something out... http://pastebin.ca/941481 |
| 18:51.18 | smash- | for echo cancelation |
| 18:51.22 | smash- | what is 256 echo tap's |
| 18:51.25 | smash- | what is an echo tap? |
| 18:52.00 | nny_1 | so.. |
| 18:52.35 | nny_1 | I give up trying to register the digium echo canceler again on this box.. anyone got an opinion or way to insall OSLEC? |
| 18:52.39 | anarcat | [TK]D-Fender: it's awful, but it works and it won't mess up my CDRs, i think |
| 18:52.46 | EmleyMoor | I've now got "reverse hold with return to chosen extension" |
| 18:52.58 | anarcat | i basically need to add entries for each fixed-lenght dialings out there |
| 18:53.00 | tzafrir | that's an echo canceller with a tail whose length is 256 samples. there are 8 samples per milli-second |
| 18:53.13 | smash- | hrm |
| 18:53.22 | smash- | what kinda effects does that cause |
| 18:53.23 | tzafrir | and hence this means a tail of 256/8 = 32 ms |
| 18:53.26 | smash- | lag or more bandwidth used? |
| 18:53.26 | *** join/#asterisk vap0rtranz (n=jpittman@75.110.17.157) |
| 18:53.34 | smash- | as in 32ms delay? |
| 18:53.54 | tzafrir | smash-, no. |
| 18:54.00 | smash- | whats the tail endup being |
| 18:54.12 | tzafrir | But the longer the tail is, the more processing it takes. |
| 18:54.17 | smash- | oohh |
| 18:54.20 | smash- | ok cpu is fine |
| 18:54.25 | smash- | so i can crank it up to 256 |
| 18:54.26 | smash- | then |
| 18:54.56 | tzafrir | longer tails may also take a bit longer to adapt the echo canceller to changes |
| 18:55.01 | smash- | i got a pri so echo shouldnt be that bad i think someone had there cell phone on speaker phone durring conf call |
| 18:55.10 | smash- | hrm i see tza |
| 18:55.41 | tzafrir | smash-, the author of OSLEC claims that 32 ms is probably more than enough for most people |
| 18:55.51 | tzafrir | at least with analog |
| 18:56.11 | smash- | gotcha i just put it back to default but i have no value of what defualt is |
| 18:56.39 | hmodes | yeah, except when your voip endpoint is 60ms away and the echo comes from the pstn side.. or someone has their cell phone echoing back in to a conference ;p |
| 18:57.20 | hmodes | echo cancellation is damn near a black art these days |
| 18:58.08 | johndo | what is a black art? |
| 18:58.44 | hmodes | http://dictionary.reference.com/browse/black%20art |
| 18:59.00 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:59.07 | *** part/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 18:59.26 | hmodes | requires a degree in echo cancellation, or a shitton of cpu ;p |
| 18:59.26 | iCEBrkr | I wish I could help figure out why we have echo with this one customer. |
| 18:59.43 | iCEBrkr | 20ms latency.. They have like a pristine network apparently. |
| 19:00.04 | iCEBrkr | and of course, I don't know all the details :( |
| 19:00.16 | *** part/#asterisk redax (i=redax@r6.hu) |
| 19:02.02 | tzafrir | hmodes, with analog you have to first and formost cancel the echo you generate |
| 19:02.31 | hmodes | well, yes, analog circuits you _control_ are almost easy compared to any form of termination |
| 19:03.00 | funxion | I keep getting 407 auth req when trying to call from a cisco gwy to * im using 1.2.9 with realtime can someone help with my config? |
| 19:03.00 | hmodes | pstn latency is low, comparatively |
| 19:03.05 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 19:03.26 | hmodes | once you get a call bounced around a carrier network and transcoded once or twice... ugh. |
| 19:03.29 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 19:04.24 | hmodes | especially if it's a particularly shitty carrier |
| 19:04.40 | tzafrir | Also, thanks to OSLEC, echo cancelletion is a bit less of afro-american art |
| 19:05.04 | jameswf | qua? |
| 19:05.36 | hmodes | huh, that's cute |
| 19:05.49 | tzafrir | <hmodes> echo cancellation is damn near a black art these days |
| 19:06.14 | ManxPower | these days? It's always been that way |
| 19:06.51 | hmodes | these days as compared to when everyone was using analog lines and used to the echo ;p |
| 19:06.58 | EmleyMoor | I wish I could buy a new main card without daughtercards to replace my TDM400 with the new version |
| 19:07.54 | hmodes | there's a reason 'comfort noise' exists, after all |
| 19:09.09 | Qwell | EmleyMoor: you can |
| 19:09.30 | EmleyMoor | Qwell: not found a supplier over here who will do one with less than 1 daughtercard |
| 19:09.31 | Qwell | I've seen several resellers that sell just the base board... |
| 19:09.49 | johndo | speaking of cards, does anyone know if I can get a sangoma A101 and later add a echo canceller to it? |
| 19:10.15 | nny_1 | lol so.... |
| 19:10.29 | *** join/#asterisk xenonex (n=xenonex@89.218.237.221) |
| 19:10.32 | nny_1 | anyone wnna try to tackle build_tools/genudevrules: line 3: udevinfo: command not found |
| 19:10.45 | nny_1 | lol i think i am going to apt-get install udev at this point |
| 19:10.46 | funxion | I keep getting 407 auth req when trying to call from a cisco gwy to * im using 1.2.9 with realtime can someone help with my config? |
| 19:10.56 | iCEBrkr | nny_1: udevinfo isn't found |
| 19:10.56 | iCEBrkr | lol |
| 19:10.57 | nny_1 | been dealing with the lack thereof and compiling zaptel for a fortnight now |
| 19:10.59 | Qwell | 1.2.9? upgrade |
| 19:11.11 | nny_1 | iCEBrkr: the commander of the obvious has spoken! |
| 19:11.12 | Qwell | you're only like 20 versions behind |
| 19:11.12 | tzafrir | nny_1, you're on Debian Sarge with kernel 2.6 and don't have udev installed . |
| 19:11.14 | iCEBrkr | <PROTECTED> |
| 19:11.19 | iCEBrkr | LOL! |
| 19:11.32 | nny_1 | tzafrir i am on a centos version called clarkconnect |
| 19:11.41 | tzafrir | interesting |
| 19:11.47 | nny_1 | and I know peter from CC is working on asterisk packages for it |
| 19:11.48 | iCEBrkr | tzafrir: Sledge hammer will fix anything |
| 19:11.52 | nny_1 | tzafrir lol |
| 19:12.12 | tzafrir | nny_1, do they use devfs as well? |
| 19:12.13 | jameswf | ~witchdoctor |
| 19:12.14 | jbot | Oo ee, oo ah ah, ting tang, walla walla bing bang Oo ee, oo ah ah, ting tang, walla walla bing bang Oo ee, oo ah ah, ting tang, walla walla bing bang Oo ee, oo ah ah, ting tang, walla walla bing bang |
| 19:12.14 | nny_1 | tzafrir i use CC cause it has great gateway/firewall tools built in, good for business style gateway device |
| 19:12.24 | nny_1 | tzafriri believe so, is there a way to confirm? |
| 19:13.11 | tzafrir | ls /dev/ide |
| 19:13.12 | nny_1 | tzafrir pretty sure, all the /dev/nodes are pre established files etc |
| 19:13.24 | nny_1 | no such file or directory |
| 19:13.37 | tzafrir | hmmm... |
| 19:13.47 | nny_1 | mocker: tey vitelity too... rates are good |
| 19:13.49 | nny_1 | try* |
| 19:14.00 | mocker | nny_1: I already have an account w/ Vitelity. :) |
| 19:14.05 | nny_1 | mocker: hehhee |
| 19:14.07 | mocker | Thanks though. |
| 19:14.16 | *** part/#asterisk rsc (n=robert@fedora/rsc) |
| 19:14.18 | nny_1 | mocker: wnp working on setting mine up today after i rebuild zap lol |
| 19:14.34 | nny_1 | then i get to code some tech rollover |
| 19:14.38 | nny_1 | but first.. zap! |
| 19:15.35 | BobLutz | oh? |
| 19:15.45 | iCEBrkr | I have asterisk running on FC7 with no /dev/ide |
| 19:16.19 | johndo | why would there be a /dev/ide? I have never heard of ide as a device node before. |
| 19:16.25 | *** join/#asterisk NirS (n=Nir@bzq-84-108-249-41.cablep.bezeqint.net) |
| 19:16.26 | BobLutz | /dev/hda perhaps? |
| 19:16.56 | iCEBrkr | No, they're all sda |
| 19:17.22 | BobLutz | iCEBrkr: I assume you have sata or scsi drives? |
| 19:17.30 | iCEBrkr | Yeah, but with IDE drives. |
| 19:17.35 | BobLutz | ? |
| 19:17.57 | iCEBrkr | Yeah, I was the same way |
| 19:17.59 | iCEBrkr | ? |
| 19:18.01 | tzafrir | nny_1, but you do have /etc/udev/rules.d , right? |
| 19:18.02 | BobLutz | lol |
| 19:18.19 | johndo | You have IDE (PATA) and they are showing up as sda devices? |
| 19:18.20 | nny_1 | tzafrir correct |
| 19:18.46 | tzafrir | how can I tell that you don't have udev? |
| 19:19.09 | nny_1 | http://www.clarkconnect.com/forums/showflat.php?Cat=0&Number=98874&an=0&page=9 |
| 19:19.14 | nny_1 | :D |
| 19:19.22 | ManxPower | I seem to recall some issue with udev not being installed, but the /etc/rc.d files are there and zaptel NOT liking it. I believe recent 1.4s fix that |
| 19:19.52 | iCEBrkr | johndo: yup |
| 19:19.55 | *** part/#asterisk ^Migs^ (n=Migs@205.122.50.69) |
| 19:20.07 | nny_1 | i am using 1.4.9.2 FWIW |
| 19:20.25 | johndo | iCEBrkr: I have run into similar situations where a chipset is used in both PATA and SATA drive controllers. |
| 19:20.27 | nny_1 | which, btw compiles 1 time on every system but fails on recompile |
| 19:20.33 | iCEBrkr | ManxPower: Yeah there was some permissions you had to grant for asterisk to use udev |
| 19:20.38 | nny_1 | (as far as i have seen) |
| 19:20.46 | nny_1 | haven't been able to test on a clean box yet |
| 19:20.49 | ManxPower | iCEBrkr: no, asterisk was thinking udev was installed when it wasn't |
| 19:20.56 | ManxPower | zaptel, actually |
| 19:20.56 | iCEBrkr | johndo: It wasn't until FC5 or 7.. |
| 19:21.03 | ManxPower | It was a pretty common Debian issue for a while |
| 19:21.13 | iCEBrkr | ManxPower: hrrm, possibly. This was on Fedora |
| 19:21.40 | robeph | [TK]D-Fender: its definitely the network after all heh, oddly .11 that its dialing (and somehow completes the call after the inordinate amount of waiting for the invite) is not even on the local 21.0.20.0/24 network O.o... tracepath leaves to the realworld, but ping seems to know right how to get to that box...something is awry on their network setup and I have no idea wth they're doing heh |
| 19:21.41 | NirS | hey all |
| 19:21.42 | ManxPower | iCEBrkr: no idea if Fedora also had the issue, if you are running 1.4.18 it should be fine |
| 19:22.10 | iCEBrkr | ManxPower: 1.4.6 and there's no issues |
| 19:22.14 | *** join/#asterisk ReDLe (n=rdl@200.138.29.170) |
| 19:22.16 | NirS | has anyone ever encountered a situation where 2 asterisk boxes will send RTP to one another, however, RTP will not pass back to the originating client ? |
| 19:22.35 | NirS | is there a situation where Asterisk will no be able to transcode or perform passthrough ? |
| 19:25.38 | BobLutz | I am trying to determine why sometimes DTMF input is not detected.. I am viewing my fxo module with ztmonitor ... I notice that if the (RX) hash is still moving while I key the next DTMF input, the second DTMF will not be picked up, is this correct? |
| 19:31.28 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 19:31.28 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:31.45 | *** join/#asterisk razu__ (n=razu@195.222.7.33) |
| 19:35.30 | nny_1 | so anyone got a solution for my problem outside of installing udev? Not sure how to move forward at this point |
| 19:37.52 | jameswf | ~repeat |
| 19:37.53 | jbot | from memory, repeat is probably If nobody answers your question, don't just repeat it. Spamming the channel and getting ignored/banned/silenced isn't going to get a faster answer; spamming is a function of time. More likely, nobody knows the answer (/msg apt ask), or you need to provide more information (/msg apt sicco)), or ask me about "rephrase", or ask smart ... |
| 19:38.38 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 19:38.45 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 19:42.07 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 19:42.22 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 19:43.07 | iCEBrkr | nny_1: Why not install udev? |
| 19:48.48 | NirS | ok, i'm stomped |
| 19:49.00 | NirS | I have an Asterisk that simply doesn't want to understand that SIP works |
| 19:49.05 | NirS | any idea why this would happen ? |
| 19:51.47 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 19:51.50 | BobLutz | ~smart |
| 19:51.50 | jbot | i guess smart is the opposite of stupid, or the opposite of straightforward and polite, or what I definitely am, or "Self-Monitoring, Analysis, and Reporting Technology" used by some hard drive manufacturers to detect imminent drive failure |
| 19:56.00 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 19:56.47 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 19:57.49 | jameswf | iminent drive failure.... warning your hard drive will fail in 5 4 3 2 1 too late |
| 19:58.15 | Strom_C | that joke would be funny if you didn't misspell imminent |
| 19:58.27 | BobLutz | Zing! |
| 20:01.26 | NirS | same problem applies to both 1.2 and 1.4 |
| 20:01.26 | NirS | this is weird |
| 20:01.26 | ManxPower | NirS: maybe you don't understand how Asterisk works. 8-) |
| 20:01.27 | NirS | Manx |
| 20:01.30 | NirS | thanks :-) |
| 20:01.35 | BobLutz | ~thanks |
| 20:01.35 | jbot | BobLutz: de nada |
| 20:01.37 | NirS | I have like 20 of these gateways |
| 20:01.43 | NirS | all configured exactly the same |
| 20:01.49 | NirS | this is the first one that caueses problem |
| 20:01.59 | ManxPower | for example if reverse DNS is not setup for the IP of the server, and there is incorrect info in /etc/hosts at least some versions of Asterisk won't work with SIP |
| 20:02.02 | BobLutz | Is it possible to enter DTMF inputs so fast that Asterisk cant register all of them? |
| 20:02.18 | NirS | I can see RTP coming back to Asterisk at the tcpdump level, but looks like Asterisk won't accept the RTP back |
| 20:02.19 | ManxPower | BobLutz: A human should not be able to do that, a modem might. |
| 20:02.29 | NirS | oh |
| 20:02.52 | ManxPower | BobLutz: if your gains are off all DTMF bets are off |
| 20:03.11 | BobLutz | ManxPower: I have been toying with rxgain all day and have not seen much of an impact |
| 20:04.06 | NirS | well, you learn something new every day |
| 20:05.10 | BobLutz | Has anyone here used the Linksys WIP330? |
| 20:05.44 | NirS | Bob, I did |
| 20:05.50 | BobLutz | You likey? |
| 20:05.51 | NirS | It's a crappy wifi phone |
| 20:05.54 | BobLutz | :-/ |
| 20:06.01 | BobLutz | seems cool... |
| 20:06.01 | NirS | not worth the 330$ I paid back then |
| 20:06.07 | NirS | cool on the outside |
| 20:06.11 | NirS | shitty on the performance |
| 20:06.15 | NirS | I love LinkSys |
| 20:06.22 | NirS | but that product still has a long way to go |
| 20:06.25 | BobLutz | wtf is Windows CE? |
| 20:06.32 | BobLutz | nm |
| 20:07.29 | NirS | From the looks of it, looks like my Asterisk box is simply unable to negotiate the SDP correcly |
| 20:07.46 | NirS | the packets are getting to the box for sure, but Asterisk just seems to disregard them completely |
| 20:07.54 | NirS | what the hell could be the problem ? |
| 20:07.54 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-110-9.bltmmd.east.verizon.net) |
| 20:13.18 | *** join/#asterisk josh1856 (n=cheese@static-71-113-205-226.frstil.dsl-w.verizon.net) |
| 20:13.18 | *** join/#asterisk gutz|work (n=mark@gateway.meteor-web.com) |
| 20:13.20 | NirS | I have an Asterisk that simply doesn't want to understand that SIP works |
| 20:13.53 | EmleyMoor | What evidence do you have? |
| 20:14.05 | NirS | well, I have 2 asterisk boxes |
| 20:14.12 | NirS | one is sending a call via SIP to the other |
| 20:14.28 | NirS | the second asterisk gets the call, answers it and starts playing RTP |
| 20:14.43 | NirS | I can see the RTP being received at the first Asterisk at the network layer |
| 20:14.48 | NirS | verified by tcpdump and ngrep |
| 20:15.02 | NirS | however, Asterisk claims that the RTP isn't there |
| 20:15.09 | NirS | won't negotiate SDP correctly |
| 20:15.16 | NirS | nothing |
| 20:15.31 | EmleyMoor | Is there any NAT between the machines? |
| 20:15.55 | NirS | the originating asterisk is behind a static NAT, however, the externip and localnet configuration are setup correctly |
| 20:16.19 | EmleyMoor | If they're both Asterisks, why do you need SIP |
| 20:16.20 | EmleyMoor | ? |
| 20:17.01 | NirS | I need to connect to something else, IAX works just fine |
| 20:17.05 | NirS | oh my god ! |
| 20:17.08 | BobLutz | LOL |
| 20:17.09 | NirS | I need to check something |
| 20:17.32 | EmleyMoor | Ah, it's a proof of concept then... fair enough |
| 20:17.37 | NirS | my assumption was incorrect |
| 20:17.46 | EmleyMoor | I have had numerous problems with SIP and NAT |
| 20:17.46 | NirS | not a proof of concept |
| 20:17.52 | EmleyMoor | (never solved them) |
| 20:18.00 | NirS | it started because I need to connect to a Veraz softswitch via SIP |
| 20:18.16 | NirS | so, I decided to check that Asterisk->Asterisk works, but the problem is identical |
| 20:20.48 | BobLutz | http://pastebin.ca/941588 <--- new way = best way to get dynamic DTMF input from caller? |
| 20:21.01 | BobLutz | If there is a better way, please let me know |
| 20:21.55 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 20:23.44 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:27.03 | NirS | even when I do rtp debug, I can see the rtp in the Asterisk, why doesn't it redirect back into the SIP channel ? |
| 20:28.08 | *** join/#asterisk xenonex (n=xenonex@89.218.237.221) |
| 20:29.58 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:32.56 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:33.09 | JayTee52 | when a call comes in on one of my PRI channels is the ${EXTEN} variable automatically populated with the dialed number? I know that my Nortel PBX that is currently connected to the 2 PRI's passes 4 gets passed the 4 digit extension, my local telco strips the NXX from the dialed number |
| 20:33.47 | jameswf | what was the name of the company selling IAX hard phones |
| 20:33.50 | nny_1 | um still stuck on compiling zaptel, udev problems, been talking in this channel for a bit about it, did anyone have a fix? This seems like "install udev" is a broken fix, and zaptel's compil options are wrong for this system |
| 20:35.10 | nny_1 | HAHAHAHA |
| 20:35.15 | BobLutz | ? |
| 20:35.21 | nny_1 | mv /etc/udev /etc/udevold |
| 20:35.24 | nny_1 | make install |
| 20:35.25 | nny_1 | done |
| 20:35.30 | nny_1 | so yeah thats why it works the first time |
| 20:35.38 | nny_1 | the first compile blindly makes udev rules |
| 20:36.03 | robeph | [TK]D-Fender: ok routes network everything set up as it should be. I still have no ideas of what may be causing the latency between the dial being sent to asterisk and the actual propagation of the invite to the foreign host. |
| 20:36.16 | nny_1 | and the following compile fails because it sees the udev rules folder and thinks udev so installed |
| 20:36.24 | nny_1 | so this is a bug |
| 20:36.37 | nny_1 | as the original compile shouldn;t make udev rules if udev isn't there |
| 20:36.56 | nny_1 | mind you this is like bitching about why my horse can't shit on public roads |
| 20:36.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:37.14 | JayTee52 | horses can't shit on public roads? |
| 20:37.19 | nny_1 | lol maybe they can |
| 20:37.22 | outtolunc | just his |
| 20:37.23 | JayTee52 | well, that's just plain wrong |
| 20:37.29 | JayTee52 | they were here before cars |
| 20:37.41 | nny_1 | damnit you're screwing up my euphemism.. should have made a car analogy |
| 20:37.57 | tzafrir | nny_1, there is a makefile variable for the detection of udev. sadly it can't be overriden from the command-line |
| 20:38.07 | JayTee52 | did you know it's still illegal (technically) to drive a car on the streets of Boston? |
| 20:38.13 | nny_1 | tzafrir any chance it is documented somewhere? |
| 20:38.27 | nny_1 | tzafrir future installs would be better if I just had a step to disable udev in the make file |
| 20:38.36 | nny_1 | (have to train people at one point) |
| 20:41.25 | *** join/#asterisk l2trace99 (n=asd@75.112.133.235) |
| 20:43.11 | BobLutz | ~cidsignalling |
| 20:43.53 | *** join/#asterisk NeedAsteriskHelp (n=jasonrot@user-69-73-40-171.knology.net) |
| 20:45.13 | Forai | bah, Damn you polytech 301! |
| 20:45.21 | Forai | Your complex nature has bested me |
| 20:47.11 | BobLutz | whoa, slow down there turbo |
| 20:48.37 | *** join/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 20:49.04 | ThatKidKel | using asterisk in a lab environment, is it possible to have it 200 OK every register request? |
| 20:49.51 | nny_1 | woot now zap isn't responding |
| 20:50.01 | nny_1 | man compiling from source is over rated sometimes |
| 20:50.20 | NeedAsteriskHelp | can someone field a question about 'show queues'? |
| 20:50.34 | BobLutz | ~question |
| 20:50.35 | jbot | hmm... question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 20:51.57 | billytwowilly | anyone know if asterisks upports qsig? |
| 20:52.27 | *** join/#asterisk mikecx (n=mikecx@pool-71-124-48-201.chi01.dsl-w.verizon.net) |
| 20:52.56 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 20:53.04 | nny_1 | woot fixed, permission issues on /dev/zap |
| 20:53.17 | nny_1 | i think this is the point where i review which part of this system is broken and fix the damn thing |
| 20:53.20 | mikecx | Is AGI the right thing to use for trying to get the status of all the channels? I'm writing a plugin for lcd4linux and trying to get zap status but AGI seems to halt for entry every time |
| 20:53.35 | nny_1 | either udev/dev is the issue or using dev with zaptel sucks.. i had this problem with permissions on debian too |
| 20:54.06 | mikecx | #/j asterisk-dev |
| 20:54.20 | nny_1 | and always because of hpec echo cancel causing said issues lol |
| 20:55.04 | nny_1 | eh it's five o clock, i am gonna go work on dinner, asterisk will have to wait |
| 20:55.07 | nny_1 | later all |
| 20:55.18 | *** join/#asterisk oej (n=olle@114.62-97-206.bkkb.no) |
| 20:55.18 | BobLutz | nny_1: bye |
| 20:55.39 | gutz|work | could an incorrectly configured zapata.conf make the telco believe that a T1 line is not plugged in? |
| 20:56.31 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 21:00.26 | ZPertee | anyone have a recommendation for seconds between audio for music on hold? |
| 21:02.35 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 21:03.00 | *** part/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 21:03.25 | znoG_ | johndo: around still? |
| 21:03.29 | znoG_ | or iCEBrkr |
| 21:04.53 | Zuchmir | is there a way to "busy" a zap channel without losing existing calls? (i.e. i want to busy out a single T1, so that calls roll-over to next T1) |
| 21:05.02 | *** join/#asterisk jasonWoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 21:06.54 | EmleyMoor | Have any of you ever had to write a user guide for users of your configured Asterisk system? I've just done one for the home system here - it came to 9 pages! |
| 21:07.34 | EmleyMoor | Even I can't always remember some of the codes I implemented on it. |
| 21:08.41 | clyrrad | whats that bot trigger for voip providers? anyone recall? |
| 21:08.49 | clyrrad | ~providers |
| 21:08.52 | clyrrad | or something like that |
| 21:11.01 | *** join/#asterisk efort (n=efort@74-86-100-202.lx-vs.net) |
| 21:11.20 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 21:14.24 | jpeeler | ~itsp |
| 21:14.25 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 21:14.25 | jpeeler | ? |
| 21:14.33 | clyrrad | thanks |
| 21:14.54 | clyrrad | ~itsplist-ca |
| 21:14.55 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 21:15.15 | clyrrad | ~itsplist-us |
| 21:15.17 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 21:20.00 | *** join/#asterisk joobie (n=joobie@58.108.192.59) |
| 21:20.02 | jasonWoot | is any one here using asterisk in a large call center environment? |
| 21:22.40 | *** join/#asterisk nitram (i=nitram@superblob.com) |
| 21:23.11 | *** part/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 21:25.17 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:25.21 | generalhan | hey all ! |
| 21:25.24 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:27.11 | efort | ~itsplist-china |
| 21:27.30 | efort | ~itsplist-ch |
| 21:27.48 | efort | ~itsplist-ca |
| 21:27.48 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 21:28.29 | efort | ~itsplist-asia |
| 21:28.39 | efort | ~itsplist-jp |
| 21:29.15 | efort | ~help |
| 21:29.23 | DarKnesS_WolF | i am having problem between * and NAT + snom 370 |
| 21:29.29 | DarKnesS_WolF | asterisk + snom on same LAN works |
| 21:29.43 | gutz|work | could an incorrectly configured zapata.conf make the telco believe that a T1 line is not plugged in? |
| 21:29.47 | DarKnesS_WolF | snome behind nat + asterisk on real IP don't work sometime i get 401 some i get 503 and some i get timeout .. |
| 21:29.50 | DarKnesS_WolF | any idea? |
| 21:32.47 | generalhan | anyone here have experience with Integra Telecom as a voice provider ? |
| 21:33.12 | generalhan | i am especially interested in anyone that has used a PRI from them, but any information about their reliability will help ! |
| 21:36.07 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
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| 21:38.21 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 21:38.21 | *** mode/#asterisk [+o file] by ChanServ |
| 21:47.06 | jasonWoot | <PROTECTED> |
| 21:47.17 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:50.20 | [TK]D-Fender | jasonWoot: You've got the source like the rest of us... |
| 21:51.40 | *** join/#asterisk xenonex (n=xenonex@89.218.237.221) |
| 21:53.46 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 21:54.32 | alrs | [TK]D-Fender: have you previously tangled with jasonwoot, or are you pointing to the vast mass of C knowledge that dangles between your legs? |
| 21:55.49 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 21:57.16 | [TK]D-Fender | alrs: No, I'm saying that there's no friendly way to do it and if thats what he wants then thats what it is going to take. |
| 21:57.22 | [TK]D-Fender | alrs: Just a matter of fact. |
| 21:57.54 | [TK]D-Fender | alrs: And no, I could not actually do this myself with any sense of realism attached... |
| 21:59.06 | alrs | [TK]D-Fender: That clarification takes the edge off |
| 21:59.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:00.52 | l2trace99 | ~help |
| 22:02.26 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:02.37 | *** join/#asterisk Leland (i=leland@ws2.discpro.org) |
| 22:02.50 | Leland | evening all |
| 22:06.37 | [TK]D-Fender | alrs: I'm often misinterpreted.... |
| 22:09.33 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:09.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:11.06 | [TK]D-Fender | alrighty, time to head out. BBL |
| 22:12.34 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
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| 22:21.36 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 22:41.33 | billytwowilly | can you guys point me to a good sip provider that will invoice a company? |
| 22:43.16 | jameswf | What... is the air-speed velocity of an unladen swallow? |
| 22:44.10 | billytwowilly | jameswf: somewhere between zero and terminal velocity |
| 22:46.24 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:46.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:47.16 | jameswf | ping |
| 22:54.37 | *** join/#asterisk RoyK (n=roy@ip-27-3-149-91.dialup.ice.no) |
| 23:00.10 | EmleyMoor | jameswf: European or African? |
| 23:07.20 | *** join/#asterisk mindCrime (n=chatzill@nat/redhat/x-30d5805ac90a15af) |
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| 23:13.58 | *** join/#asterisk sob0l (n=sobol@devel4.net) |
| 23:17.26 | sob0l | I have problems with g726 and PAP2 on * 1.4.18 |
| 23:17.29 | sob0l | NOTICE[24968]: chan_sip.c:5420 process_sdp: No compatible codecs, not accepting this offer! |
| 23:17.58 | EmleyMoor | Are there any particular pros/cons of alsa or oss in Asterisk? |
| 23:21.00 | plik | neither are neccessary, so it depends what you want to do |
| 23:21.45 | EmleyMoor | plik: Just wondered as mine came with oss enabled and it says I could enable alsa instead |
| 23:22.56 | plik | I think alsa is often prefered for a lot of linux sound apps, but Asterisk doesn't even use the sound card usually... |
| 23:23.17 | plik | I suoopose thats different if you want to place calls on the CLI console... |
| 23:23.45 | plik | or listen to mixmonitor recordings on the asterisk server without copying them elsewhere first |
| 23:23.51 | EmleyMoor | Hmmm, probably won't make much difference to me, then... I do use festival though... |
| 23:24.22 | plik | ah..yeah - that could be it |
| 23:24.48 | plik | but again, AFAIK you'd only need it to test festival directly on the server |
| 23:25.15 | EmleyMoor | All testing I've done with it has been through the dialplan |
| 23:25.19 | lirakis_away | how does asterisk generate the tag for To: headers? |
| 23:25.30 | lirakis_away | is it a "simple" hash .. on some part of an invite or... an md5 .. some thing like that? or am i way off base |
| 23:26.04 | plik | EmleyMoor: no need to worry then :) |
| 23:26.25 | EmleyMoor | plik: Fair enough |
| 23:26.35 | EmleyMoor | Just watching out for the future |
| 23:26.54 | EmleyMoor | I have as future-proof a dialplan as I can at present as well |
| 23:30.59 | EmleyMoor | I'm still on 1.2.13 (that's what's in Debian) but I have already eliminated at least one if not two apps from it that only become deprecated in 1.4 |
| 23:32.01 | *** join/#asterisk rkeene (i=1011@oc9.org) |
| 23:32.12 | angryuser | hi all |
| 23:32.17 | *** join/#asterisk sm9 (n=sm9@83-103-90-63.ip.fastwebnet.it) |
| 23:32.33 | *** part/#asterisk sm9 (n=sm9@83-103-90-63.ip.fastwebnet.it) |
| 23:32.42 | jameswf | high |
| 23:33.02 | EmleyMoor | l'eau <g> |
| 23:33.21 | jameswf | you know you have had to much to drink when you depricate on your shoes |
| 23:33.26 | angryuser | druid has releases it's opensource version at http://www.voiceroute.org/ have someone tested it out yet ? |
| 23:33.46 | rkeene | I'm running the latest stable Asterisk (1.4.18); This is my initial setup and I'm having a problem connecting to extensions. I get this error: [Mar 13 18:30:15] NOTICE[24538]: chan_sip.c:13879 handle_request_invite: Call from '1234' to extension '500' rejected because extension not found. but according to the output of `dialplan show' at the console I would think extension '500' does exist. |
| 23:33.56 | jameswf | Our tester is installing druid right now... we have played with their commercial version |
| 23:33.58 | Wayhigh | angryuser: I heard it totally puts out.. unf unf unf |
| 23:34.28 | EmleyMoor | rkeene: In the same context as the default context of the phone? |
| 23:34.32 | angryuser | <rkeene> pastein your extensions.conf&sip.conf |
| 23:34.53 | jameswf | according to a trix post druid will provision 300 phones in 2 seconds walk your dog and mow your grass |
| 23:35.06 | angryuser | <jameswf> i have installed on vm, got errors when typing 'add new trunk" |
| 23:35.18 | angryuser | or more 'clicking" |
| 23:35.27 | jameswf | well the software is perfect so must be you |
| 23:35.41 | jameswf | :) |
| 23:35.49 | angryuser | <jameswf> perfect ? hehe |
| 23:36.18 | *** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com) |
| 23:36.52 | angryuser | <Wayhigh> do you consider it a bad or good thing ? |
| 23:37.58 | rkeene | EmleyMoor, Yes ... I think |
| 23:38.06 | rkeene | angryuser, Just a second. |
| 23:38.18 | Wayhigh | angry: dude.. anyone or anything that puts out has to be a good thing.. how else would desperate guys get laid? |
| 23:38.24 | EmleyMoor | Pop the files in a pastebin somewhere and I'll have a look |
| 23:38.57 | angryuser | <Wayhigh> ah sorry, it was new expression to me 'to put out' , so i asked |
| 23:39.49 | ZPertee | how can I make so that I can dial even once I answer a call. for example what I want to do is be able to answer a call and if they ask for extension 16 I want to be able to dial on the phone *16 and send them to extension 16 VM? |
| 23:40.46 | angryuser | <ZPertee> clear you mind, explain clearly what do you want to do |
| 23:41.05 | rkeene | http://www.rkeene.org/tmp/asterisk/ -> http://www.rkeene.org/tmp/asterisk/extensions.conf http://www.rkeene.org/tmp/asterisk/sip.conf |
| 23:41.48 | angryuser | <ZPertee> and it is called transfer (T ot t option in dial) |
| 23:42.35 | EmleyMoor | rkeene: You have no context "from-internal" but the phone is in it |
| 23:43.33 | angryuser | ^^ he got the reason |
| 23:43.41 | rkeene | Why is the phone in it ? |
| 23:44.04 | EmleyMoor | Because of the line "context=from-internal" in its definition in sip.conf |
| 23:44.15 | angryuser | <rkeene> who knows, ufo intervention |
| 23:44.32 | rkeene | D'oh, I didn't see that in the definition |
| 23:44.32 | rkeene | Thanks! |
| 23:44.41 | ZPertee | angryuser: this http://www.voip-info.org/wiki/view/Asterisk+Voicemail+Dialplan is what I was looking for |
| 23:45.16 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:45.25 | angryuser | <ZPertee> so i suppose you found it |
| 23:46.39 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 23:46.59 | rkeene | EmleyMoor, I've deleted the "context=" line from the [1234] entry... and I get the same behaviour, is this expected ? |
| 23:47.02 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:47.18 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:48.19 | EmleyMoor | rkeene: I'd put "Âcontext=internal" or "context=default" if I were you |
| 23:48.24 | EmleyMoor | Erk! |
| 23:48.33 | EmleyMoor | context=internal even |
| 23:49.41 | rkeene | EmlyMoor, Okay, I added "context=default" (and restarted the asterisk daemon for good measure), and I still get the same behaviour |
| 23:49.57 | Nivex | IAX2/guest@voip.cerient.net/40 |
| 23:50.22 | EmleyMoor | rkeene: Hmmm... tried a "show dialplan"? |
| 23:52.21 | rkeene | EmleyMoor, http://www.rkeene.org/viewer/tmp/asterisk/dialplan-show-output.txt.htm |
| 23:52.28 | EmleyMoor | Ah, yes |
| 23:52.55 | EmleyMoor | OK, try "context=internal" |
| 23:53.22 | rkeene | Oops, I see the typo ... include => default from default :-P |
| 23:53.38 | EmleyMoor | Yes - that can't be helping |
| 23:53.46 | EmleyMoor | include => internal might have helped |
| 23:53.59 | rkeene | Thanks! |
| 23:54.55 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 23:55.34 | ZPertee | how can I configure a hotline in asterisk? so that when I pick up the handset it auto dials |
| 23:55.42 | EmleyMoor | rkeene: Before you know it you'll have a dialplan about the size of mine if not larger |
| 23:56.16 | EmleyMoor | ZPertee: "batphone" is certainly supported for Zap - not sure about other technologies |
| 23:56.39 | rkeene | EmleyMoor... I hope not.. I only have 2 PSTN networks to deal with and my internal network :-P |
| 23:57.04 | EmleyMoor | rkeene: I am actually surprised at the size of mine |
| 23:57.14 | EmleyMoor | 328 extensions (965 priorities) in 62 contexts. |
| 23:57.32 | rkeene | (Public phone network, DSN network, and then the 500 or so handsets that will be deployed) |
| 23:58.28 | rkeene | Excellent.. I'm able to call another SIP phone connected to the same server |
| 23:59.44 | *** join/#asterisk draygon (n=draygon@76.185.106.151) |
| 23:59.52 | draygon | Hi, does anyone in here use Festival? |