| 00:06.26 | GuyOCanada | drmessano: i purchased a toll free NUmber From LES what should i use as Peer Type? |
| 00:07.06 | MacWinner | is the CDR system something native to asterisk? i'm just trying to get an idea of whether it's more like a 3rd party system, or a core feature of asterisk |
| 00:28.41 | GuyOCanada | I'm wondering what i do wrong |
| 00:29.07 | GuyOCanada | while setting it up |
| 00:35.48 | GuyOCanada | when using IAX2 to connect to a DID number provider do you have to register it on both sides? |
| 00:44.25 | GuyOCanada | bah it works now the did but still les.net shows as not registered while my box says it is registered |
| 00:44.59 | *** join/#asterisk MrFollies (n=Miranda@60-242-243-193.static.tpgi.com.au) |
| 00:45.34 | MrFollies | Good evening all... |
| 00:53.28 | *** join/#asterisk adjohn (n=adjohn@211.9.39.161) |
| 00:54.10 | *** join/#asterisk mrdigital (n=mrdigita@65-78-103-248.c3-0.drf-ubr2.atw-drf.pa.cable.rcn.com) |
| 00:54.34 | mrdigital | whats a good voip supplier |
| 00:54.36 | mrdigital | like phones etc |
| 00:56.46 | *** join/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 00:57.01 | *** join/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 00:57.24 | WilliamK | telephonydepot is good |
| 00:58.53 | tessier | Teliax is overloaded again. :( |
| 01:01.04 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-73f44066824b268d) |
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| 01:04.02 | mrdigital | basically i wanna $40ish phone to play with |
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| 01:05.15 | *** mode/#asterisk [+o mog] by ChanServ |
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| 01:20.46 | *** join/#asterisk ManxPower (n=manxpowe@242.sub-75-200-115.myvzw.com) |
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| 01:41.45 | teknoprep | hey [TK]D-Fender you here on christmas day |
| 01:45.45 | dacs | Merry christmas everyone |
| 01:57.31 | *** join/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 02:00.49 | lmadsen | GuyOCanada: I've had some success with voicenetwork.ca |
| 02:03.28 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 02:05.06 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 02:05.14 | tengulre | hi,all |
| 02:05.40 | dacs | tengulre: hi |
| 02:05.44 | tengulre | anybody can tell me how to test stun server, I don't known how to connect other client? |
| 02:08.52 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:09.52 | dacs | tengulre: stunserver.org |
| 02:11.03 | dacs | i just installed * 1.2.13 i need help configuring it |
| 02:12.59 | tengulre | dacs: yes, I use this stun, but I don't know how to test it! |
| 02:17.21 | ManxPower | dacs: why did you pick 1.2.13? |
| 02:17.38 | *** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-73f44066824b268d) |
| 02:17.55 | dacs | ManxPower: i am using debian and this is the plg i found |
| 02:18.12 | dacs | ManxPower: do you have any suggestione please? |
| 02:19.21 | ManxPower | Then you did not install asterisk. You installed some package someone built. |
| 02:19.39 | ManxPower | We always recommend you install Asterisk from source. |
| 02:20.40 | dacs | ManxPower: ok i will remove 1.2.13 and install which ver you tell me |
| 02:21.23 | ManxPower | Either the latest 1.2.x or the latest 1.4.x |
| 02:22.56 | *** join/#asterisk Tux889 (n=Tux889@mctnnbsa28w-142167102006.pppoe-dynamic.nb.aliant.net) |
| 02:24.17 | mrdigital | ManxPower: pm? |
| 02:25.22 | ManxPower | mrdigital: only if you are going to offer me a very profitable consulting job. |
| 02:28.38 | mrdigital | ManxPower: yes |
| 02:29.45 | ManxPower | Then pm away. |
| 02:31.19 | *** join/#asterisk ozus (n=ozus@202.77.104.197) |
| 02:47.39 | rvhi | hi, what's the difference bewteen callerid(ani) and callerid(num) |
| 02:48.45 | ManxPower | rvhi: You need to understand what ANI is first. Google is your friend |
| 02:49.33 | rvhi | i looked up ani, but not very sure, is this another kind of number identify, but not the same as caller id? |
| 02:49.56 | ManxPower | ANI is generally only available on incoming toll free numbers. |
| 02:50.34 | rvhi | can i set ani on outbound calls using set(callerid(ani)=xxxx)? |
| 02:50.52 | rvhi | what is the difference of setting callerid(num)? |
| 02:51.09 | ManxPower | ANI is not really supposed to be customer settable. |
| 02:51.42 | dacs | is there is a bot for acronyms? |
| 02:51.49 | dacs | !ATA |
| 02:51.54 | ManxPower | You can set it, but the telco will almost always override it. ANI us not supposed to be spoofable. |
| 02:51.59 | ManxPower | ~ata |
| 02:52.10 | jbot | well, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 02:52.10 | dacs | ~ata |
| 02:52.10 | jbot | somebody said ata was Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 02:52.29 | dacs | ManxPower: thanks :) |
| 02:52.35 | dacs | ~ANI |
| 02:52.36 | jbot | from memory, ani is Automatic Number Identification Systems |
| 02:53.26 | rvhi | 911 center seems to use ani instead of caller id |
| 02:53.46 | rvhi | i'd like to be able set ani |
| 02:54.15 | ManxPower | or BTN (billing telephone number), which can never be set by the customer. |
| 02:57.16 | mvanbaak | mornin |
| 02:57.21 | mvanbaak | Wed Dec 26 03:57:21 CET 2007 |
| 02:57.21 | rvhi | how asterisk support e911? http://www.voiceipsolutions.com/Asterisk_E911.htm |
| 02:57.40 | *** join/#asterisk Tux889 (n=Tux889@mctnnbsa28w-142167102006.pppoe-dynamic.nb.aliant.net) |
| 02:58.23 | rvhi | looks like setting ani and then update revese lookup database? |
| 03:00.05 | ManxPower | It TOTALLY depends on what the e911 service center uses for locating the caller. |
| 03:00.28 | ManxPower | And each e911 service center decides how they will do that. |
| 03:00.33 | mvanbaak | spoken word ! |
| 03:00.55 | mvanbaak | "I'm at home" |
| 03:01.20 | mvanbaak | "oh, so you are at <street>, <number> <city> <state>" |
| 03:01.24 | mvanbaak | "yeah !" |
| 03:01.46 | mvanbaak | sorry |
| 03:01.50 | mvanbaak | had xmas dinner |
| 03:01.56 | mvanbaak | started at 4 PM |
| 03:02.04 | mvanbaak | and it's 4 AM now |
| 03:02.11 | *** join/#asterisk vetetix (n=vetetix@83.222.34.219) |
| 03:02.11 | mvanbaak | Wed Dec 26 04:02:11 CET 2007 |
| 03:04.17 | ManxPower | One issue we had recently was that when an office moved, the telco never updated the service address they had on file. |
| 03:05.15 | *** join/#asterisk ozus (n=ozus@202.77.104.197) |
| 03:05.41 | mvanbaak | we use ISDN for e911 |
| 03:05.54 | ManxPower | I believe the issue was discovered during routine 911 testing. |
| 03:06.08 | mrdigital | ManxPower: check pm |
| 03:06.25 | mvanbaak | every call is routed with IAX2 |
| 03:06.27 | Nugget | http://macnugget.org/photos/tws200711/IMG_11653 <-- routine 911 testing |
| 03:06.47 | mvanbaak | but e911 is routed on the local ISDN interface |
| 03:07.04 | *** join/#asterisk suvir (n=suvir@ppp-124.120.232.190.revip2.asianet.co.th) |
| 03:07.24 | mvanbaak | I had a car like that |
| 03:08.20 | mvanbaak | but tire replacement was more then my salary |
| 03:08.25 | mvanbaak | so I had to sell it |
| 03:08.32 | Nugget | yeah, that can be sort of painful. |
| 03:09.58 | mvanbaak | ah well |
| 03:10.06 | mvanbaak | I got over it |
| 03:10.14 | mvanbaak | saving for a DB9 now |
| 03:11.03 | Nugget | heh |
| 03:11.14 | Nugget | I've only had to buy two tires so far, but they were rears. :) |
| 03:11.22 | mvanbaak | they are almost the same price now |
| 03:14.58 | mvanbaak | I'm off to bed |
| 03:14.58 | ManxPower | ~book |
| 03:14.59 | 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 |
| 03:15.17 | mvanbaak | had about 10 shots of irish whisky |
| 03:15.25 | mvanbaak | so it's time to sober up |
| 03:16.19 | mvanbaak | damn, daytime job to cleanup the house |
| 03:16.40 | mvanbaak | latero all |
| 03:22.14 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 03:26.51 | dacs | ManxPower: i compiled and installed * 1.4.16.2 |
| 03:38.00 | dacs | ~pbx |
| 03:38.01 | jbot | i heard pbx is a Private Branch eXchange |
| 03:42.58 | dacs | ~ATA vs FXO |
| 03:45.13 | ManxPower | ATA vs FXO is like asking Car vs Tire |
| 03:47.40 | [TK]D-Fender | indeed brilliant |
| 03:57.03 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 04:03.54 | dacs | ManxPower: so, you saying that an * will not work with ATA |
| 04:06.18 | [TK]D-Fender | dacs, No it means that you used 2 things that can't be compared |
| 04:06.28 | [TK]D-Fender | ~ata |
| 04:06.28 | jbot | hmm... ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 04:06.43 | [TK]D-Fender | dacs, it PROVIDES either a FXS or an FXO interface |
| 04:07.09 | [TK]D-Fender | dacs, And if they ATA speaks a protocol * supports then yes you can use it. |
| 04:08.26 | [TK]D-Fender | dacs, Currently the Linksys SPA series ATAs are the ones we recommend for use with * |
| 04:08.30 | dacs | [TK]D-Fender: thank you, that explain it to me |
| 04:09.33 | dacs | [TK]D-Fender: i just finished compiling and installing * 1.4.16.2, this is my 1st install of * never messed with it before |
| 04:09.34 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 04:09.52 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f403f384d4916735) |
| 04:10.15 | dacs | and i am trying to configure it |
| 04:10.18 | [TK]D-Fender | dacs, Congratulations. Be sure to download the BOOK, and have the link to the WIKI handy for other reference. Here's a simple basic config to help get you started : |
| 04:10.19 | [TK]D-Fender | ~sipnat |
| 04:10.20 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:10.23 | [TK]D-Fender | then : |
| 04:10.24 | [TK]D-Fender | ~book |
| 04:10.24 | 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 |
| 04:10.26 | [TK]D-Fender | ~wikis |
| 04:10.27 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 04:10.40 | [TK]D-Fender | oops, strike that first guide! Bad reflex! |
| 04:10.48 | [TK]D-Fender | dacs, THIS guide rather : |
| 04:10.50 | [TK]D-Fender | ~jerjerguide |
| 04:10.51 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 04:12.09 | dacs | [TK]D-Fender: nice thank you |
| 04:12.36 | dacs | however i still have a little problem that i need to solve befoare i get starting on this project? |
| 04:12.49 | [TK]D-Fender | dacs, Do you? |
| 04:14.03 | dacs | my * box has 2 eth card, on is a wireless card which i use to connect to internet, the 2nd NIC (which i have to config and re-routr internet from wireless card to that 2nd NIC) |
| 04:15.08 | [TK]D-Fender | dacs, you mean you want your * box to act as a router for your home network? |
| 04:17.42 | dacs | [TK]D-Fender: hmmm, i have my DSL modem connected to Linksys wireless router, then i have my windows box connects to the internet wirelessly thru the wireles router, and i have my 2nd box"Debian" which connects to the internet also wirelessly thru the same linksys router |
| 04:18.22 | dacs | like right now i am using my windows box to ssh to debian to chat with you |
| 04:18.29 | [TK]D-Fender | dacs, ok, no big deal. So it uses its wireless NIC to connect to your Linksys Router. |
| 04:18.34 | dacs | am i making sense |
| 04:18.45 | [TK]D-Fender | dacs, nothing wrong with that. |
| 04:19.24 | [TK]D-Fender | dacs, You will need that "~sipnat" guide I linked by accident for your * to work with other VoIP resources outside of your LAN if thats one of your needs |
| 04:19.30 | dacs | [TK]D-Fender: yes, and i installed * in my debian box and i want to use the 2NIC for my ATA |
| 04:19.55 | mrdigital | DACS: hook up the ATA to the network |
| 04:20.18 | [TK]D-Fender | dacs, Sure, nothing wrong with that. |
| 04:20.22 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 04:20.38 | [TK]D-Fender | dacs, You could also wire it to your router. Doesn't matter really.... |
| 04:20.47 | dacs | mrdigital: can i hookup the ATA to the wireless router |
| 04:20.57 | mrdigital | yes through a WIRED port |
| 04:21.06 | dacs | cool cool |
| 04:21.16 | [TK]D-Fender | dacs, I've never see a WIRELESS ATA, but if your router has a free Rj45 jack on it then yes |
| 04:21.21 | *** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au) |
| 04:21.26 | dacs | so i think i am ready to start setting up my box |
| 04:21.26 | Thazza | hey all |
| 04:22.47 | dacs | hi Thazza |
| 04:29.58 | mrdigital | ~book |
| 04:29.58 | 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 |
| 04:29.59 | Thazza | Anyone here had any issues with asterisk 1.14.17.2 yet? |
| 04:30.53 | Thazza | sorry i mean 1.14.16.2 |
| 04:31.36 | Thazza | it totaly broke my copy of asterisk. loads of segment errors when making a call from a sip channel to another outbound sip channel |
| 04:33.37 | *** join/#asterisk patrickv0x (n=patrick@67.131.93.17) |
| 04:35.08 | Thazza | trying to backdate to 1.14.16 |
| 04:36.05 | patrickv0x | Does anyone interest in purchasing a NACT Switch + NTS2000 billing for a give-away price ? please contact me privately. |
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| 04:48.57 | dacs | can someone please help me config sip.conf this is my first time |
| 04:50.17 | [TK]D-Fender | dacs, I gave you a quick guide which includes most of that and you've got the boook. You should work on this a bit yourself and not attempt a 100% hand-holding learning experience here.... |
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| 04:50.59 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:51.28 | dacs | yes you did, but i am confused |
| 04:53.41 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 04:55.56 | dacs | anyways i am heading to bed now , thank you [TK]D-Fender for all the help, talk to you guys tomorrow |
| 05:10.23 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 05:11.38 | WilliamK | good evening coppice |
| 05:12.17 | coppice | good afternoon |
| 05:12.46 | WilliamK | give me 1 more hour and I can say morning :) |
| 05:13.20 | WilliamK | so how's things on the opposite side of the globe? |
| 05:14.48 | coppice | pleasant, sunny, a little cool |
| 05:17.01 | WilliamK | coool |
| 05:17.02 | WilliamK | :) |
| 05:17.58 | *** join/#asterisk Abydos313 (n=abydos31@adsl-76-214-9-14.dsl.lsan03.sbcglobal.net) |
| 05:18.05 | coppice | oh, we're seriously cool, but its 22C at the moment |
| 05:20.08 | WilliamK | that's warm |
| 05:20.16 | WilliamK | you should try here |
| 05:21.01 | coppice | warm is when I don't need a duvet |
| 05:21.01 | WilliamK | it's between 0c and 6c here |
| 05:22.03 | WilliamK | wanna trade? :) |
| 05:22.34 | coppice | it hasn't been 0C here for 15000 years |
| 05:23.01 | coppice | at least at ground level. there are frosts in the hills |
| 05:23.25 | WilliamK | see you're lucky |
| 05:24.03 | coppice | not really. it does get down to maybe 5C for a week or two each year, and nobody has central heating |
| 05:24.19 | WilliamK | wow |
| 05:24.40 | coppice | so much for living on a tropical island :-) |
| 05:24.52 | WilliamK | I'd still trade I think |
| 05:24.53 | WilliamK | :) |
| 05:25.17 | WilliamK | I got to put my hand in the swimming pool the other night |
| 05:25.21 | WilliamK | was coooooolllllldddd |
| 05:25.36 | coppice | does it freeze over? |
| 05:25.59 | WilliamK | not the pool, but the grass was certainly starting to freeze |
| 05:26.13 | WilliamK | takes a very cold week to freeze the pool |
| 05:26.31 | WilliamK | pump stays running constantly moving about 18,000/gal of water |
| 05:27.18 | WilliamK | cept my problem the other night was that leaves got in covering the skimmer and air locked it to where it wasn't moving any water with the pump on |
| 05:27.21 | coppice | a couple of hundred years ago they barbequed on the frozen river thames in london. |
| 05:27.22 | WilliamK | so I had to unblock it |
| 05:27.47 | WilliamK | wow |
| 05:28.27 | coppice | the had a winter fair right on the thames each winter. must have been fun |
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| 06:04.35 | GuyOCanada | why would someone want to freeze their pool? |
| 06:08.58 | coppice | for skating, of course |
| 06:10.45 | Qwell | would be better off empty |
| 06:11.14 | coppice | how can you skate on tiles? |
| 06:15.57 | *** join/#asterisk clusco (i=clusco@203.114.50.232) |
| 06:15.58 | Qwell | what do you mean how? O.o |
| 06:16.06 | Qwell | search youtube for hundreds of examples :p |
| 06:16.28 | clusco | hello everyone! |
| 06:16.32 | Qwell | bed |
| 06:18.39 | WilliamK | actually freezing a pool or emptying a pool for a season here would be many thousands in repairs |
| 06:18.47 | WilliamK | and I do mean thousands if not tens of thousands |
| 06:18.55 | WilliamK | enough the insurance company would not like someone |
| 06:19.31 | coppice | why is emptying bad? |
| 06:19.40 | WilliamK | soil will shift |
| 06:19.44 | WilliamK | thus causing it to crack |
| 06:20.06 | coppice | you mean jsut the change of weight upsetting things? |
| 06:20.08 | clusco | everyone.... im here as damn newbies about PBX |
| 06:20.11 | WilliamK | yep |
| 06:20.28 | WilliamK | can only leave it empty for about a week max |
| 06:20.39 | WilliamK | and you better hope it doesn't rain |
| 06:21.15 | coppice | lots of places empty pools for long periods in the winter, sometimes as a matter of practice, and sometimes for repairs |
| 06:21.24 | WilliamK | I'm preparing to empty mine this spring so I can powerwash the gunite and refill the pool with fresh water |
| 06:21.44 | WilliamK | not here in TX, it's a BAD idea to even dream the thought |
| 06:21.51 | WilliamK | soil is mostly made from clay |
| 06:22.34 | clusco | WilliamK: newbies here would like to ask... which step should i start first to setup PBX ??? |
| 06:22.44 | coppice | i live a few minutes work from a beautiful complex of pools, on the sea front, looking out across HK harbour to HK island. I avoid all these personal hasssles :-) |
| 06:22.48 | clusco | i was thinking to buy a Sangoma card |
| 06:23.04 | clusco | is that a right choice ??? |
| 06:23.17 | GuyOCanada | I ordered my spa box to setit up with my old pbx :) see what will happen |
| 06:23.29 | coppice | clusco: depends what you want to do |
| 06:27.49 | GuyOCanada | what fun things do people do with *? |
| 06:30.23 | suvir | fun things are like call the radio station in a contest |
| 06:31.24 | GuyOCanada | :) |
| 06:31.40 | GuyOCanada | like doing what the old phreakers did? |
| 06:32.04 | GuyOCanada | the 100th caller will win the grand prize ok (I have asterisk and voip outbound trunks) let me call 100 times :D |
| 06:32.04 | GuyOCanada | ? |
| 06:32.11 | GuyOCanada | is that what you mean |
| 06:32.32 | WilliamK | coppice... have you heard of a company by the name of Attractel (re: T.38 stack for *)? |
| 06:32.39 | suvir | no like when the radio station plays a DTMF sequence on the air that is time for people to call in right away |
| 06:32.52 | suvir | we have asterisk listening for that DTMF sequence |
| 06:32.54 | coppice | yes. zoa, who is often here, is the guy behind that |
| 06:33.15 | suvir | as soon as it hears the dtmd it calls the radio station and at the same time calls your extension |
| 06:33.30 | GuyOCanada | :) ooh good |
| 06:33.42 | GuyOCanada | any other fun things? |
| 06:35.16 | WilliamK | know if it works well and/or pricing info? |
| 06:35.51 | GuyOCanada | I need to do some stuff that people would say wow lets change to * :) |
| 06:36.12 | WilliamK | I haven't been able to get g711u pass-through working on this one box....what gets under my skin is it was working before the last 5-6 SVN updates |
| 06:36.46 | WilliamK | Guy, I think T.38 is a major roadblock for *; it's been talked about for years |
| 06:37.36 | coppice | well, G.711 is never going to work with any predictability, whatever some people who've seen a few pages pass successfully might say. |
| 06:37.53 | WilliamK | I'd rather toss $ at doing programming VS tossing it at hardware/telco for dedicated fax equipment |
| 06:38.14 | suvir | home and office device automation using asterisk and lirc sounds fun |
| 06:38.36 | suvir | control anything with an IR remote throught asterisk |
| 06:38.40 | WilliamK | the odd thing is the end user can fax outbound all day long without issues, even refill the dang postage machine without issues |
| 06:38.41 | coppice | the attractel stuff apparently does termination and gateway T.38 up to 14,400bps. I have no idea |
| 06:38.43 | coppice | - how well tested it is against other T.38 boxes |
| 06:38.44 | coppice | - how much patching of asterisk is required |
| 06:38.46 | coppice | - how much it costs |
| 06:38.47 | WilliamK | which I'd think would be more complex than a fax |
| 06:39.17 | GuyOCanada | suvir: what hardware do you need for that? |
| 06:40.02 | WilliamK | I'm truely wondering how much they're making off the product VS offering a way for digium to aquire the product |
| 06:40.09 | coppice | WilliamK its pot luck, and that luck may change with temperature |
| 06:40.11 | suvir | guyocanada - look at the LIRC project (linux infrared remote control) |
| 06:40.18 | WilliamK | true |
| 06:40.54 | coppice | why would digium acquire the product? |
| 06:41.11 | WilliamK | to get a full T.38 stack which * lacks currently |
| 06:41.31 | WilliamK | if someones already done the work for it, may be cheaper and more effective for marketing * and hardware |
| 06:42.07 | coppice | I don't see why that product would be the most attractive way to achieve that goal, and I'm pretty sure it wouldn't go in the free version |
| 06:42.49 | WilliamK | even if it was a pay per mod like the g729 codec I think it would still have the desired affect |
| 06:43.38 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 06:43.40 | WilliamK | honestly I don't look at * itself being free if you're buying hardware to support an * server (digium boards) which many do |
| 06:43.58 | coppice | this is how digium has lockin. they can do this smoothly, while its really hard for anyone else |
| 06:45.20 | WilliamK | I understand, I'm just saying that something that's truely free and open wouldn't require a set platform which * relies upon digium boards, or a heck of alot of patching |
| 06:45.27 | WilliamK | so I don't look at it as being totally free |
| 06:45.30 | WilliamK | the cost is the hardware |
| 06:45.57 | coppice | if some of the software isn't open source, you might as well use a totally closed solution |
| 06:46.21 | WilliamK | as Microsoft put it one month when they mailed out several guest meal tickets..... "There is no such thing as a free lunch." |
| 06:47.21 | WilliamK | I'm just saying that further development of software; free or not in digiums case is a win-win to be able to say we have this and others are still struggling |
| 06:47.41 | WilliamK | instead when I search the net on this specific issue, I see others doing it and digium struggling since 2005/6 |
| 06:48.23 | coppice | while I have no direct knowledge of how well the attractel stuff works, 2 people trying it said my stuff is well ahead :-) |
| 06:48.36 | WilliamK | kinda see where I'm going with this? |
| 06:49.05 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 06:49.09 | WilliamK | honestly, I like your coding and I've gotten to use it once before |
| 06:49.11 | WilliamK | worked well |
| 06:49.24 | coppice | digium aren't struggling. they just aren't doing anything about it |
| 06:50.03 | WilliamK | yeah and that's the scary approach as it results in some loss of market share to others that can do it |
| 06:50.24 | WilliamK | that's kinda like hoping the issue will go away and some things just don't go away |
| 06:50.43 | WilliamK | I just see it as a major roadblock at the moment |
| 06:50.50 | coppice | do you know if attractel offer T.37? |
| 06:51.33 | WilliamK | don't know, I have an email into them and if they call in the morning I'll ask |
| 06:52.34 | WilliamK | the other thing I'm anxious to see is the encryption of SIP sessions |
| 06:53.21 | coppice | * doesn't implement SIP. The SIP spec says any compliant server must implement TLS :-) |
| 06:55.23 | WilliamK | :) |
| 06:59.05 | *** join/#asterisk MACscr (n=Mark@adsl-75-23-66-10.dsl.peoril.sbcglobal.net) |
| 06:59.14 | MACscr | hello everyone |
| 07:03.34 | GuyOCanada | does anyone have experience with text to speech or speech to text? |
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| 10:11.32 | lucent | I rather think that a label printer just swore at me |
| 10:12.42 | lucent | by some course of events I managed to bump into it in such a way that it turned on, cleared its settings, set a large font, entered in a string of text and striking non-alphanumeric glyphs, and then hit the print + confirm buttons |
| 10:12.48 | lucent | all on accident |
| 10:12.56 | lucent | no, I dare say the bloody thing just swore at me |
| 10:13.28 | GuyOCanada | Everyone is sleeping lucent |
| 10:13.39 | lucent | apparently the aliens are not |
| 10:13.47 | lucent | they are having their way with my dignity. |
| 10:14.41 | GuyOCanada | maybe aliens are making fun of you? |
| 10:14.45 | lucent | I mention it here because the message was: MMM [glyph of a telephone] |
| 10:14.47 | GuyOCanada | by doing all that cool stuff? |
| 10:15.12 | lucent | It's fucking E.T. phoning home alright? |
| 10:15.25 | GuyOCanada | ooh e.t. knows how to phone? |
| 10:15.41 | GuyOCanada | does he use analog phones or VoIP? |
| 10:16.05 | lucent | VOLP |
| 10:16.11 | GuyOCanada | hey check your dialplan maybe he messed around with it to send messages to his own planet |
| 10:16.23 | lucent | grr bloody thing |
| 10:16.27 | lucent | ;/ |
| 10:16.33 | lucent | how are you anyways |
| 10:16.45 | lucent | in Turkey? |
| 10:16.58 | lucent | wonder what you're also doing up at this hour |
| 10:19.44 | GuyOCanada | im in Canada right now |
| 10:20.23 | GuyOCanada | its 2 am will be going to futureshop at 4 am waiting in line or hoping and trying to be the first in line :D |
| 10:20.33 | GuyOCanada | 6 am doors open i need to get some drives :P |
| 10:21.20 | lucent | ah |
| 10:21.30 | lucent | best of luck and wishes to you mate |
| 10:21.40 | lucent | sleepy time for me again |
| 10:21.42 | GuyOCanada | thanks |
| 10:21.52 | GuyOCanada | well 1.5 tb harddrives usb2 for 150 $ |
| 10:21.56 | GuyOCanada | is a good deal :) |
| 10:22.06 | GuyOCanada | that makes a person stay inline on boxing day |
| 10:22.13 | lucent | bloody memory going farked need to type the right command |
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| 12:42.19 | MrFollies | Does anyone know where to find the release notes for the various * releases? |
| 12:43.40 | mvanbaak | http://downloads.digium.com/pub/asterisk/releases/ |
| 12:44.05 | MrFollies | Thanks! |
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| 14:09.30 | *** join/#asterisk dkatz334 (i=121mhz@189.sub-75-195-132.myvzw.com) |
| 14:10.00 | dkatz334 | Morning everyone |
| 14:11.10 | dkatz334 | Anybody find an option to have asterisk not send InUse hints when sending 484 responses? |
| 14:13.36 | *** join/#asterisk ManxPower (n=manxpowe@151.sub-75-203-53.myvzw.com) |
| 14:21.07 | *** join/#asterisk achu (n=achu@125.17.244.2) |
| 14:21.16 | achu | Hi All |
| 14:21.33 | dkatz334 | howdy |
| 14:21.39 | dkatz334 | Seems quiet this AM |
| 14:22.23 | achu | my asterisk server (Asterisk 1.2.25) was wroking very good, It is running as a virtual host on VMware Server |
| 14:22.57 | achu | today I had taken a snapshot and tested revert feature |
| 14:23.37 | achu | from that time when I get a user's voice mail or in a conference room , the system pre recorded messages |
| 14:23.41 | achu | are lagging |
| 14:23.50 | achu | I can't hear it clearly |
| 14:24.14 | ManxPower | We don't really support Asterisk on VMs. It works for some people, it doesn't for others. |
| 14:24.24 | achu | can anybody help me to solve this issue ? |
| 14:24.54 | ManxPower | achu: if the USB or RTC are messed up, that would cause that issue with Asterisk. |
| 14:24.54 | achu | ManxPower, anyway you have any idea , which section I have to check ? |
| 14:25.25 | achu | also I have tested a fresh configuration on another machine |
| 14:25.27 | ManxPower | achu: I'm sorry, I don't know VMware well enough to make any suggestions on how to fix the issue with VMware. |
| 14:25.34 | achu | at beginning it was working great |
| 14:25.56 | achu | but after copiying the modules and configuration files its start the problem |
| 14:26.28 | achu | ManxPower, can you please expain about USB or RTC ? |
| 14:26.43 | achu | so that I can check it |
| 14:27.08 | ManxPower | achu: Asterisk times many of it's operations, including audio playback by using the RTC timer or USB timer. |
| 14:27.26 | achu | k |
| 14:27.30 | ManxPower | The "ztdummy" zaptel driver is what does this. |
| 14:28.03 | achu | its loaded successfully |
| 14:28.21 | ManxPower | then unload it. |
| 14:28.34 | achu | k |
| 14:28.49 | ManxPower | you won't be able to use meetme, but maybe other stuff will start working. |
| 14:29.33 | achu | you mean rmmod ztdummy right ? |
| 14:29.49 | ManxPower | correct, then start asterisk |
| 14:29.58 | achu | k |
| 14:30.07 | ManxPower | But as I said and as everyone else here will say, it is not recommended to run Asterisk in a virtual machine. |
| 14:30.32 | ManxPower | Even VMware says to not run applications that require realtime operation. |
| 14:30.34 | achu | hmm , still the same |
| 14:30.45 | ManxPower | achu: then there is nothing else I can suggest. |
| 14:30.55 | achu | k |
| 14:31.35 | achu | why you are recommending not to run asterisk in a virtual machine ? |
| 14:32.41 | achu | I was doubtful about g729_a codec , because as I said the fresh machine was working good |
| 14:32.53 | achu | and when I copied the licences and codec to the new one |
| 14:32.59 | achu | its start problematic |
| 14:33.00 | ManxPower | Because Asterisk may not get enough CPU time to service all the audio playback. |
| 14:33.08 | achu | oh |
| 14:33.13 | dkatz334 | Yeah, I wouldn't think asterisk on vmware would do very well either. Unless your only running it for a few phones. |
| 14:33.19 | achu | but all the calls are working great |
| 14:33.30 | dkatz334 | How many concurrent calls? |
| 14:33.35 | ManxPower | Also Asterisk may not get enough CPU time to process incoming audio as well. |
| 14:33.41 | achu | about 6-8 |
| 14:34.07 | ManxPower | Do you get any console messages that say "request to schedule in the past" |
| 14:34.19 | achu | let me check it |
| 14:34.21 | dkatz334 | Achu, I gotta agree with Manx, your past experience may not be indicative of future performance. |
| 14:36.03 | achu | ManxPower, no there is no message like that |
| 14:36.34 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-8e4200b978e8e23c) |
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| 14:44.00 | dkatz334 | Anybody know how to make a hint that only shows when the channel is in a call? I have SIP phones that are showing InUse when they are dialing... quite annoying and lots of extra traffic |
| 14:44.35 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
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| 14:52.51 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 14:57.53 | dkatz334 | Repost: Anybody know how to make a hint that only shows when the channel is in a call? I have SIP phones that are showing InUse when they are dialing... quite annoying and lots of extra traffic |
| 14:59.00 | *** join/#asterisk MagInfo (n=mlx@gw-MagInfoCenter-magn.suttk.ru) |
| 14:59.06 | MagInfo | Hi there. I have a problem. Does anybody can help with this? |
| 14:59.15 | MagInfo | I have device with FXS an FXO ports. When I have incoming call from FXO asterisk reject it and says comething line chan_sip.c: username mismatch, have <1507>, digest has <0010> |
| 15:00.09 | *** join/#asterisk Laureano (n=chatzill@200.59.172.38) |
| 15:00.10 | MagInfo | FXS port has username 1507 and FXO port has username 0010 |
| 15:00.31 | lmadsen | dkatz334: modify the code |
| 15:01.38 | russellb | making a call ... sounds "in use" to me :) |
| 15:02.10 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 15:02.15 | *** join/#asterisk Deeewayne (n=Deeewayn@ool-43507e21.dyn.optonline.net) |
| 15:02.15 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:02.25 | Laureano | MagInfo: What is the problem that you are experiencing? |
| 15:02.40 | MagInfo | Laureano: I have device with FXS an FXO ports. When I have incoming call from FXO asterisk reject it and says comething line chan_sip.c: username mismatch, have <1507>, digest has <0010> |
| 15:03.19 | Laureano | I have the same problem with one of my VoIP providers. Do you have a sip peer defined for the FXS port, right? |
| 15:04.24 | MagInfo | Laureano: Yes. FXS port has username 1507 and FXO has username 0010 |
| 15:04.49 | MagInfo | When call comes from FXO I have this warning in asterisk |
| 15:06.42 | Laureano | MagInfo: What is happening is this: Asterisk tries to find a peer definition for the host that is trying to authenticate, and the first definition found is the FXS port (the username 1507. Asterisk tries to compare the digest, but fails, because the username for the peer found using the host is the FXS peer; but the call is being generated by another user from the same host. |
| 15:07.27 | Laureano | I haven't found yet a way to solve this problem :-( |
| 15:07.27 | MagInfo | Laureano: I know it. But How can I solve it? |
| 15:08.24 | Laureano | I don't know. I have seen this topic in the -users list, but without any solution. |
| 15:08.40 | MagInfo | I tried to set insecure=very in user.conf for those extentions. Calls from FXO goes correctly but in PBX it comes from extention 1507, instead of FXO has username 0010 |
| 15:09.41 | Laureano | Yeap, I tried that too. I haven't found a way to use different users for the same host, without any collateral effect. |
| 15:10.49 | MagInfo | Ok. I think that I can't find the answer here. I'm going to modify source. Give me your mail and I will send ptch to you when I complete |
| 15:11.22 | Laureano | Thank you MagInfo. |
| 15:12.42 | achu | ManxPower, its works now |
| 15:12.49 | achu | I don't know how |
| 15:12.58 | MagInfo | I need to now. Thank you |
| 15:15.42 | jer | any way i can get * to ask an ip phone to make a distinctive ring based on which channel a call comes in over? |
| 15:17.33 | *** join/#asterisk dkatz334 (i=121mhz@189.sub-75-195-132.myvzw.com) |
| 15:18.08 | tzafrir_laptop | jer, hmm.. what specific IP phone? |
| 15:18.20 | jer | tzafrir_laptop, it's a snom370 |
| 15:18.39 | jer | i'm just not sure how to craft the sip notify message |
| 15:20.20 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:21.28 | ManxPower | Jer: the Wiki should have details |
| 15:21.51 | jer | ManxPower, voip-info? |
| 15:21.58 | ManxPower | ~wiki |
| 15:22.12 | ManxPower | correct |
| 15:22.18 | jer | ok |
| 15:22.23 | ManxPower | However, the info MAY be out of date. |
| 15:26.12 | lmadsen | and probably is :) |
| 15:26.19 | jer | ManxPower, yes, i've often found that, but in this case, it looks promising anyway. i'll just add to the sip header with a http uri pointing at a wav file, and see if that works |
| 15:30.30 | Laureano | #asterisk-bugs |
| 15:30.33 | Laureano | Ups |
| 15:49.45 | tzafrir_laptop | ~wiki wikipedia |
| 15:50.14 | Qwell | well, that wasn't helpful |
| 15:50.33 | tzafrir_laptop | that bot is used in other channels as well |
| 15:50.49 | *** join/#asterisk shadebob (n=chatzill@84.16.28.38) |
| 15:50.51 | tzafrir_laptop | its dictionary is not specific to here, from what I understand |
| 15:50.54 | shadebob | hi, |
| 15:50.59 | tzafrir_laptop | Or maybe it was just formed? |
| 15:52.08 | shadebob | hi tzafrir_laptop, I don't known if you remember my problem of dropped calls. I had actived FULL debug, and I find (app_dial.c: Unable to forward voice frame) in the log just before dropped call... Maybe more verbose for you? |
| 15:52.51 | tzafrir_laptop | sorry, I don't remember the problem... |
| 15:54.53 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 15:54.53 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.16.2 (2007/12/20), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.7.1 (2007/12/13), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
| 15:55.10 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:55.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:06.31 | *** join/#asterisk CapRicORN^80 (n=you@209.8.41.61) |
| 16:07.49 | CapRicORN^80 | hi ! i am new to asterisk . i just configured asterisk on Centos 5 and then i did configuration in sip.conf and extension.conf . After that i download xlite and added SIP Accounts from SIP Accounts setting |
| 16:08.32 | CapRicORN^80 | i gave display name , username ( which is in sip.conf n password ) and in domain i mentioned the ip address of my asterisk server |
| 16:08.33 | lmadsen | sounds right so far |
| 16:08.45 | lmadsen | not domain -- use host= |
| 16:09.33 | CapRicORN^80 | i dont have any option like host= in xlite menu |
| 16:09.51 | CapRicORN^80 | its giving me Registering error |
| 16:10.23 | CapRicORN^80 | one more thing that on my centos box . i can only check asterisk is running by ps -ef | grep asterisk |
| 16:10.33 | lmadsen | oh, I thought you meant sip.conf |
| 16:10.42 | lmadsen | asterisk -rvvv |
| 16:10.43 | Qwell | isn't it like proxy address in xlite? |
| 16:10.46 | CapRicORN^80 | and asterisk is running |
| 16:10.58 | CapRicORN^80 | yes there is option proxy in xlike |
| 16:11.02 | CapRicORN^80 | xlite * |
| 16:11.09 | lmadsen | CapRicORN^80: iptables -L -v |
| 16:11.16 | lmadsen | I bet the firewall is blocking |
| 16:11.57 | CapRicORN^80 | i dont configure any iptables |
| 16:12.04 | CapRicORN^80 | anyways i just run that command . let see |
| 16:12.25 | CapRicORN^80 | no same problem |
| 16:13.04 | lmadsen | that command just shows you the rules for iptables |
| 16:13.23 | lmadsen | does it return information other than 3 lines for FORWARD, INPUT, and OUTPUT with blank rules? |
| 16:13.44 | lmadsen | that command wasn't supposed to do anything but give you information towards your problem |
| 16:14.10 | CapRicORN^80 | i have no security enabled on my linux box |
| 16:14.13 | *** join/#asterisk mog (n=mog@nat/digium/x-dc7dcbba6ce954d0) |
| 16:14.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:14.19 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 16:14.42 | funxion | anyone here today |
| 16:14.52 | lmadsen | nope |
| 16:15.38 | funxion | damn I've got a weird problem |
| 16:17.05 | lmadsen | yes you do |
| 16:17.24 | funxion | lol |
| 16:18.05 | funxion | do you have any experience with european based * boxen? |
| 16:18.21 | shadebob | hi, someone have some experience with "Unable to forward voice frame" Warning and after a Hangup ? |
| 16:18.58 | funxion | it seems somewhere along the line the leading 0 of the incoming callerid on a e1 pri is getting dropped. |
| 16:19.21 | lmadsen | shadebob: that's not a problem |
| 16:19.35 | lmadsen | it just means audio was trying to be passed after the hangup, which of course can't happen because the call hung up |
| 16:19.57 | shadebob | lmadsen: hi, Big problem because I have many dropped calls with this message... |
| 16:20.05 | CapRicORN^80 | any suggestion for me ? |
| 16:20.41 | lmadsen | shadebob: could be a NAT problem |
| 16:21.13 | funxion | is digium support open today? |
| 16:21.25 | shadebob | lmadsen : not a NAT problem |
| 16:21.25 | Qwell | funxion: I would imagine so |
| 16:21.32 | shadebob | lmadsen :here a sample : |
| 16:21.33 | *** join/#asterisk nirz (n=nir@bzq-79-178-72-175.red.bezeqint.net) |
| 16:21.37 | shadebob | [Dec 26 16:33:45] VERBOSE[24213] logger.c: -- Executing [022361632@101:4] Dial("SIP/101-081875c8", "Zap/2/022361632||tT") in new stack |
| 16:21.38 | shadebob | [Dec 26 16:33:45] DEBUG[24213] dsp.c: dsp busy pattern set to 0,0 |
| 16:21.40 | shadebob | [Dec 26 16:33:45] DEBUG[24213] chan_zap.c: Dialing '022361632' |
| 16:21.41 | shadebob | [Dec 26 16:33:45] DEBUG[24213] chan_zap.c: Deferring dialing... |
| 16:21.43 | shadebob | [Dec 26 16:33:45] VERBOSE[24213] logger.c: -- Called 2/022361632 |
| 16:21.44 | shadebob | [Dec 26 16:33:48] DEBUG[24213] chan_zap.c: Done dialing, but waiting for progress detection before doing more... |
| 16:21.46 | shadebob | [Dec 26 16:35:09] WARNING[24213] app_dial.c: Unable to forward voice frame |
| 16:21.48 | shadebob | [Dec 26 16:35:09] VERBOSE[24213] logger.c: -- Hungup 'Zap/2-1' |
| 16:21.49 | shadebob | [Dec 26 16:35:09] NOTICE[24213] cdr.c: CDR on channel 'Zap/2-1' not posted |
| 16:21.51 | shadebob | [Dec 26 16:35:09] VERBOSE[24213] logger.c: == Spawn extension (101, 022361632, 4) exited non-zero on 'SIP/101-081875c8' |
| 16:21.58 | lmadsen | wow |
| 16:22.05 | lmadsen | I can't believe you just did that |
| 16:22.09 | funxion | lol |
| 16:22.28 | *** mode/#asterisk [-v shadebob] by lmadsen |
| 16:22.50 | lmadsen | I just got the same thing in a priv msg -- that's 2 strikes |
| 16:23.00 | funxion | lol |
| 16:23.36 | shadebob | lmadsen : big big big problem for me, many hours of headache .... And IRC is my last solution.... :( |
| 16:24.00 | lmadsen | you still have to follow the netiquette of IRC |
| 16:24.01 | shadebob | lmadsen : and many customers (more than 40) with the SAME problem |
| 16:24.34 | shadebob | lmadsen : I know :( |
| 16:24.59 | lmadsen | you know? then why are you doing it? |
| 16:25.07 | lmadsen | (ignoring netiquette) |
| 16:25.57 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 16:25.58 | lmadsen | emergencies should be handled by a consultant |
| 16:26.09 | shido6 | amen |
| 16:26.43 | lmadsen | if you can't afford a consultant, then it's not really an emergency |
| 16:26.48 | cpm | indeed |
| 16:27.03 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 16:28.54 | *** join/#asterisk dkatz334 (i=121mhz@53.sub-70-198-130.myvzw.com) |
| 16:30.54 | russellb | funny to hear a bunch of consultants saying that :-p |
| 16:32.13 | CapRicORN^80 | when i run command safe_asterisk it say command not found |
| 16:32.33 | lmadsen | russellb: of course :D |
| 16:33.21 | [TK]D-Fender | CapRicORN^80, What user are your running as? |
| 16:33.34 | [TK]D-Fender | you* |
| 16:33.58 | CapRicORN^80 | root |
| 16:35.00 | [TK]D-Fender | CapRicORN^80, Was * running properly before? have you checked your path ev var? |
| 16:35.13 | CapRicORN^80 | yes |
| 16:35.35 | CapRicORN^80 | i just follow the installation on CentOs 5 |
| 16:35.40 | CapRicORN^80 | and every thing work fines |
| 16:35.51 | CapRicORN^80 | but safe_asterisk not working |
| 16:36.00 | [TK]D-Fender | CapRicORN^80, Go pastebin the CLI output of your attempt including the output of "which safe_asterisk", and some attempt to show us where you're looking for it. |
| 16:36.01 | [TK]D-Fender | ~pb |
| 16:36.02 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:36.02 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 16:36.17 | shadebob | lmadsen : no problem with consultant. If you can help me, just msg me in private and I will give my mail for a quote. |
| 16:36.31 | lmadsen | it's a holiday and I don't do zap :) |
| 16:36.39 | lmadsen | but maybe someone else does? |
| 16:37.06 | CapRicORN^80 | no safe_asterisk in (/usr/kerberos/sbin:..................) |
| 16:37.11 | funxion | shadebob where is your * box? |
| 16:37.24 | shadebob | funxion : Morocco |
| 16:37.39 | funxion | are any outboudn calls working? |
| 16:38.01 | funxion | what is it connected to? |
| 16:38.19 | funxion | is it pri or cas? |
| 16:38.27 | funxion | or is it analog |
| 16:39.25 | *** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net) |
| 16:39.27 | shadebob | funxion : This problem is in many boxes. We package a box with asterisk, We have some experiences with Asterisk... We have more than 220 installed box in Morocco and Algeria... And more than 40 with this problem... |
| 16:40.53 | shadebob | funxion : we don't understand why. So if any consultant want to send quote to solve this problem I'm OK |
| 16:41.04 | CapRicORN^80 | there is nothing to paste in pastebin |
| 16:41.18 | CapRicORN^80 | its simple . no safe_asterisk in (paths) |
| 16:41.46 | [TK]D-Fender | CapRicORN^80, How did you install *? |
| 16:42.03 | [TK]D-Fender | CapRicORN^80, And show us what version you're running |
| 16:42.17 | CapRicORN^80 | http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_centos.html |
| 16:42.31 | ManxPower | "Unable to forward voice frame" usually means the far end hung up. |
| 16:42.31 | CapRicORN^80 | [TK]D-Fender : check the above link |
| 16:42.57 | ManxPower | The "done dialing but waiting for progress detection" issue can be removed by setting callprogress=no and busydetect=no in zapata.conf |
| 16:43.44 | ManxPower | You should NEVER EVER set those options on a PRI |
| 16:44.02 | [TK]D-Fender | CapRicORN^80, What version of *? |
| 16:44.39 | ManxPower | Setting callprogress or busydetect will cause random disconnects. |
| 16:44.58 | shadebob | ManxPower : I encount this problem with SIP provider, ZAP from TDM400 and Xorcom... :( I don't understant why "Unable..." appear before ZAP hangup and phone hangup :( |
| 16:45.16 | CapRicORN^80 | asterisk 1.4.16.2 |
| 16:45.19 | [TK]D-Fender | CapRicORN^80, do "which asterisk" |
| 16:45.28 | ManxPower | shadebob: it is not an error message, it is expected to happen, it is not a problem. |
| 16:45.39 | CapRicORN^80 | no asterisk in (path of directory ) |
| 16:45.54 | *** join/#asterisk dkatz334 (i=121mhz@53.sub-70-198-130.myvzw.com) |
| 16:45.58 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust233.winn.cable.ntl.com) |
| 16:46.00 | ManxPower | CapRicORN^80: then asterisk is NOT installed or your paths are really really screwed up. |
| 16:46.22 | CapRicORN^80 | ah |
| 16:46.43 | CapRicORN^80 | i gave you the asteriskguro.com installation for centos |
| 16:46.49 | CapRicORN^80 | i guess thats not wrong |
| 16:47.03 | ManxPower | CapRicORN^80: something could have easly went wrong. |
| 16:47.26 | [TK]D-Fender | CapRicORN^80, Just follow the instructs in the readme & install docs in your source tarball.... |
| 16:47.37 | tzafrir_laptop | shadebob, please remind me: what device exactly? FXO? BRI? |
| 16:47.45 | funxion | shadebob what zones are u using in zaptel.conf |
| 16:48.13 | funxion | I had a similiar problem with a boxin europe |
| 16:48.27 | funxion | the far end wasnt recognizing the tones |
| 16:48.31 | shadebob | ManxPower: So this message "Unable...." is not linked with my hangup problem |
| 16:48.34 | ManxPower | Fortunatly shadebob already knows how to fix the problem. |
| 16:48.54 | funxion | I guess |
| 16:49.01 | ManxPower | shadebob: It is not linked to your hangup problem. You have callprogress=yes or busydetect=yes. THAT is what is causing your problem. |
| 16:49.04 | shadebob | funxion : zone fr but I have this problem with SIP provider and SIP gateway |
| 16:49.30 | ManxPower | I think this is the third time I've told you that. |
| 16:49.56 | ManxPower | CapRicORN^80: run "make install" again in the asterisk source directory. |
| 16:49.59 | shadebob | ManxPower: I will find some sample of hangup calls between SIP provider and Linksys SPA941... |
| 16:50.01 | ManxPower | look for error messages. |
| 16:50.09 | De_Mon | yep that was #3 all right! |
| 16:50.29 | shadebob | ManxPower: I have played with RTP packet size on my phones, change last firwmare, etc... but nothing |
| 16:51.02 | ManxPower | I think you will have to stop and start asterisk when you change the two ooptions I told you to change. |
| 16:51.13 | ManxPower | De_Mon: maybe he doesn't really want to solve his problem. |
| 16:53.23 | *** join/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 16:53.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:53.24 | meuserj | ok.. we need to block calls from a number on our asterisk system.. brief googling hasn't revealed a clear path to doing this.... can someone point me in the right direction? |
| 16:53.24 | De_Mon | ManxPower I think hes in panic mode and not actually reading the words you are typing |
| 16:53.24 | ManxPower | De_Mon: poor thing. |
| 16:53.24 | [TK]D-Fender | meuserj, "show function CALLERID" , "show application gotoif" <------------- |
| 16:53.24 | ManxPower | De_Mon: I'll stop typing then. |
| 16:53.33 | shadebob | ManxPower, De_Mon : No I'm in SSH mode ... Sorry |
| 16:53.42 | CapRicORN^80 | well i guess every thing was alright |
| 16:53.56 | CapRicORN^80 | i got all output right , the asterisk logo etc |
| 16:54.09 | ManxPower | I'll stop typing at least until shadebob confirms he made the changes I told him to make and that he has stopped and started Asterisk |
| 16:54.12 | shadebob | ManxPower : callprogress=no and busydetect=no on all the boxes |
| 16:54.41 | De_Mon | $5 says he didn't check and just knows what is "preconfigured" |
| 16:55.01 | ManxPower | shadebob: The only reason to have the message "Done dialing, but waiting for progress detection before doing more..." is because either callprogress or busydetect is set to yes. |
| 16:56.32 | shadebob | ManxPower: This "SPAM" is in a test box.... I check somes customers box and callprogress=no and busydetect=no |
| 16:57.21 | Nugget | anyone want to buy a TDM400P with 2 FXS and 1 FXO? |
| 16:57.40 | ManxPower | Nugget: finally upgraded to a PRI? |
| 16:57.55 | Nugget | I'm shutting down my home asterisk server. No real use for it now. |
| 16:58.55 | Corydon76-dig | You don't use it like I do, to remotely reset the cable modem? |
| 16:59.22 | Nugget | I've never had a cable modem. :) |
| 17:00.01 | Nugget | I'm going to drop the land line entirely, so it wouldn't be useful for that purpose anyway |
| 17:01.25 | rob0 | I lost my telco line at home. |
| 17:01.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:01.52 | rob0 | BellSouth wasn't taking the debit card payments we sent them ... oh well. |
| 17:02.36 | rob0 | I finally figured out that we were paying $30/month for nothing. |
| 17:03.28 | De_Mon | Nugget I'm interested we're using a clone for the landline right now ;) |
| 17:03.29 | rob0 | POTS service is not competitively priced for home & small business. |
| 17:03.49 | rob0 | ouch! |
| 17:03.51 | cpm | rob0, I can give you a steep discount for debit card payments |
| 17:04.02 | cpm | I'll bill you $25 a month for nothing. |
| 17:04.10 | rob0 | WOW! |
| 17:04.19 | cpm | it's nothing. |
| 17:04.20 | cpm | :) |
| 17:04.22 | rob0 | So I'll save $5 over BellSouth! |
| 17:04.25 | De_Mon | and then wipe out the account! |
| 17:04.25 | cpm | yup |
| 17:05.03 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.99.127) |
| 17:05.14 | rob0 | $60/year ... enough to take the family out for dinner once or twice. |
| 17:05.49 | Nugget | De_Mon: it's a couple years old, so it's almost certainly got an older firmware on it. No sure if you can flash it or whatnot. But $150 and it's yours. :) |
| 17:09.01 | De_Mon | checking current prices |
| 17:09.26 | [TK]D-Fender | rob0, yay McD's again! Their favourite! |
| 17:10.47 | ManxPower | rob0: Shell has been cashing the checks I send them for my gas card payments, but not crediting my account. This is month 3 of the saga. |
| 17:10.50 | rob0 | cpm: $17.5/month, my final offer. |
| 17:11.11 | cpm | Okay, done. |
| 17:11.31 | cpm | but yer a friend |
| 17:11.35 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 17:11.57 | rob0 | That puts my savings up to $150/year, thus, I can buy that card from Nugget if De_Mon doesn't. |
| 17:12.36 | [T]ank | i can dial extension 1009 from extension 1001. But then i try to transfer from 1009 to 1010 and get the following error in the CLI> Call from '1009' to extension '10' rejected because extension not found. |
| 17:12.52 | [T]ank | doing the transfer from the transfer button on a pollycom SIP 301 |
| 17:13.13 | [T]ank | here is the dial from my macro: exten => s,n,Dial(SIP/${ARG1}||tToO) |
| 17:13.31 | [T]ank | what else can I look to for the issue? |
| 17:13.55 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 17:14.40 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 17:14.57 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 17:15.25 | lirakis | im trying to store the unique id when a call comes in as a different variable, then stamp the userfield with that stored unique id, so that if a cdr forks (from queuing) i can correlate the call records |
| 17:15.44 | De_Mon | [T]ank what makes you think the extension exists? asterisk doesn't think it does. |
| 17:16.05 | [T]ank | I can dial the extension directly. I cannot transfer to it |
| 17:16.13 | lirakis | 1) is this a retarded way to do this... 2) if so whats a better way 3)if not its not retarded, how can i test if the stored variable has already been "set" |
| 17:17.06 | [T]ank | lirakis: use NoOp() to print the value of your variable |
| 17:17.37 | lirakis | [T]ank: and how can i test taht it is set? |
| 17:17.40 | rob0 | [T]ank, if calling directly uses "1010" then you're stripping digits that you need. |
| 17:17.47 | *** part/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 17:18.32 | dkatz334 | Anybody know how to make a hint that only shows when the channel is in a call? I have SIP phones that are showing InUse when they are dialing... quite annoying and lots of extra traffic |
| 17:18.35 | lirakis | [T]ank: for instance .. in a queue situation. the queue will send a agent bound call to the specified context.. in which, if the stored unique id is already set, i will use it and stamp the userfield with it... if not.. i want to store the unique id and stamp it |
| 17:18.50 | lirakis | [T]ank: so i need a if(var is set) type function |
| 17:19.12 | [T]ank | here is the output from cli when dialing direct to extension 1010: http://pastebin.ca/832807 |
| 17:20.19 | [T]ank | lirakis: in the dialplan you can just issue a NoOp($variable) and have it print out its value |
| 17:20.21 | *** join/#asterisk Havokmon (n=rick@64-198-2-66.ip.mcleodusa.net) |
| 17:20.23 | [TK]D-Fender | BBIAB |
| 17:20.29 | lirakis | [T]ank: dude.. are you listening? |
| 17:20.45 | lirakis | [T]ank: i dont need to physically see if its set |
| 17:21.01 | lirakis | [T]ank: i need asterisk itself to know if it is set.. . then to react based on whether or not it is |
| 17:21.31 | [T]ank | lirakis: then no. i am not listening, |
| 17:21.47 | ManxPower | GotoIf($["${UNIQUEID" = ""]?firstplace,secondplace) |
| 17:22.22 | lirakis | ManxPower: thank you |
| 17:22.48 | *** join/#asterisk lotho (n=lotho@static.69.46.46.78.clients.your-server.de) |
| 17:23.03 | ManxPower | lirakis: a simple google search, mailing list archive search, hanging out on the channel, or voip-info search, as well as "show applications" would have given you the answer. |
| 17:23.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:23.17 | ManxPower | Heck, I'll bet extensions.conf.sample shows and example. |
| 17:23.46 | lirakis | ManxPower: i did google... but i searched for "asterisk variables" and "asterisk test variables" |
| 17:24.40 | ManxPower | I leave it as an exercise to the reader to correct the obvious flaw in my example. |
| 17:25.18 | lirakis | ManxPower: fair enough ;) |
| 17:25.33 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 17:25.40 | Sci_05 | morning all |
| 17:26.50 | Havokmon | Hmmm. I had a working PRI into a PBX. Then I put an asterisk box with 2 T1 cards 'in between' the CO and the PBX, and setup the config. So why might only the PBX PRI come up, and the CO one say 'Provisioned,Down,Active' ? |
| 17:28.24 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 17:31.12 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 17:31.46 | *** part/#asterisk harpal (n=Harpal@124.125.79.212) |
| 17:46.48 | *** join/#asterisk RoyK (n=roy@72.84-48-10.nextgentel.com) |
| 17:50.50 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 17:52.19 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 17:54.04 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 18:02.22 | lirakis | is there a good way to correlate the records that are created during a call transfer in asterisk 1.2? |
| 18:03.07 | lirakis | since the channel vars are torn down.. i can seem to find a way to mark the userfield with an indicator to "group" a set of cdrs as "one call" |
| 18:03.17 | lirakis | or .. create a relation ship between them rather |
| 18:03.59 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 18:12.26 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 18:13.15 | De_Mon | ManxPower did shadebob ever figure out what going on? |
| 18:13.21 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:18.40 | tzafrir_laptop | De_Mon, no, he didn't |
| 18:20.04 | *** join/#asterisk Laureano (n=chatzill@200.59.172.38) |
| 18:26.47 | *** join/#asterisk techie (n=techie@adsl-76-214-31-16.dsl.lsan03.sbcglobal.net) |
| 18:26.58 | *** part/#asterisk techie (n=techie@adsl-76-214-31-16.dsl.lsan03.sbcglobal.net) |
| 18:28.44 | shadebob | De_Mon : I search ... I search ... |
| 18:37.01 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:38.05 | *** part/#asterisk Deeewayne (n=Deeewayn@ool-43507e21.dyn.optonline.net) |
| 18:40.31 | Kobaz | anyone familiar with getting h323 clients going with asteirsk and ooh323 (or something equivalent) |
| 18:41.34 | De_Mon | shadebob so you verified (as in looked) at those two settings and made sure they were correct, then restarted the test system, and still get the message "Done dialing, but waiting...bla"? |
| 18:42.57 | *** join/#asterisk mog (n=mog@nat/digium/x-98e3f982d803ea9e) |
| 18:42.57 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:43.21 | shadebob | De_Mon : I don't think it's a Zaptel problem because SIP provider/gateway have also dropped calls. It's a random problem, I still wait for more and more log, sip debug, etc from my customers box... |
| 18:43.44 | shadebob | De_Mon : I currently work on the rtp packet size of the linksys SPA921/941 |
| 18:44.02 | mog | morning all |
| 18:44.14 | mog | im looking for cheap iax2 origination |
| 18:44.24 | mog | and i need a 617 boston did |
| 18:45.02 | De_Mon | shadebob oh, just make sure you work on one problem at a time so people trying to help you don't go batty |
| 18:45.27 | De_Mon | it does sound like you've got 2 separate problems there |
| 18:47.40 | shadebob | De_Mon : Problem is dropped calls :) just that... I working with it for one month now... I will solve this issue and write this experience in voip-info.org... Maybe some other people encount the same problem.... |
| 18:48.33 | shadebob | De_Mon : like this od post http://www.broadbandreports.com/forum/r19418497-Best-settings-for-freepbxasterisk-Calls-randomly-dropping |
| 18:51.06 | shadebob | De_Mon : maybe a Mac Mini compatibility issue ... because all my boxes are bases on Mac Mini Core Solo |
| 18:57.41 | [TK]D-Fender | shadebob, Thats all just boring SIP, and they say 1 provider drops, the other doesn't. That means the provider sucks, not some hypothetical settings. |
| 18:58.17 | [TK]D-Fender | shadebob, And as that thread shows... FreePBX shmucks don't tend to know much about anything. |
| 18:58.57 | shadebob | [TK]D-Fender: an old post of 2005, just to illustrate my problem of dropped call. |
| 18:59.00 | De_Mon | shadebob are all these customers usign the same voip provider? I'd focus on the pstn |
| 18:59.30 | [TK]D-Fender | shadebob, Unfortunately doesn't contribute anything of value to debugging your issue |
| 19:00.00 | shadebob | De_Mon: no. Some Classic Internet SIP Provider, some Local Wimax SIP provide, some PSTN from TDM or Xorcom, some E1 from TE... |
| 19:00.48 | *** join/#asterisk karleeto (n=karl@c-98-193-182-202.hsd1.tn.comcast.net) |
| 19:01.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:01.07 | De_Mon | what version of asterisk are you using? I didn't see that mentioned anywhere (it is asterisk right not freepbx or something) |
| 19:01.19 | shadebob | I will solve this issue, sure... With time and ..... log... But amount of customers calls to my support callcenter give me headache.... |
| 19:01.42 | shadebob | 1.4.4 and 1.4.14 |
| 19:02.51 | De_Mon | are the logs comming from both versions or mostly 1.4.4? |
| 19:03.01 | shadebob | mostly 1.4.14 |
| 19:04.23 | De_Mon | what os/distro are those mac mini's running? |
| 19:04.39 | shadebob | De_Mon : etch |
| 19:05.06 | karleeto | ok, i've got an onboard nic i NEED to use (with trixbox), well, my nics driver wasnt introduced until 2.6.20, what should i do ? use my own 2.6.23 kernel ? can i use trixbox's /boot/config-XXXX file to config my new 2.6.23 kernel similarly to TB's kernel ? |
| 19:05.16 | karleeto | or sohuld i got about this a different way? |
| 19:05.40 | De_Mon | damn can you really put a pci card in those things? |
| 19:05.57 | De_Mon | debian etch hrm |
| 19:06.03 | De_Mon | ~trixbox |
| 19:06.04 | jbot | i heard trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 19:06.07 | shadebob | De_Mon : that's why I use Xorcom |
| 19:06.21 | shadebob | shadebob : TE are on HPML110 |
| 19:06.28 | karleeto | De_Mon: damn it. |
| 19:06.37 | shadebob | De_Mon: TDM also on HP ML110 |
| 19:06.44 | karleeto | De_Mon: this is more of linux kernel question |
| 19:07.01 | [TK]D-Fender | karleeto, try #linux :) |
| 19:07.20 | [TK]D-Fender | karleeto, * doesn't care much about your kernel |
| 19:07.33 | karleeto | [TK]D-Fender: i'm in there |
| 19:07.40 | karleeto | [TK]D-Fender: and in #centos ;) |
| 19:07.43 | *** join/#asterisk dave_mwi (n=dave_mwi@208.251.184.82) |
| 19:07.44 | [TK]D-Fender | karleeto, "Just do it" - Nike |
| 19:07.48 | karleeto | which is really where i should ask |
| 19:07.53 | karleeto | ;) |
| 19:08.05 | [TK]D-Fender | karleeto, then lift up your skirt, grab your balls and MAN UP! |
| 19:08.22 | [TK]D-Fender | karleeto, Now please don't utter that dirty word again today, ok? |
| 19:09.32 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583307.dsl.bell.ca) |
| 19:10.05 | dave_mwi | I am initiating an outbound call with a call file, and the first line in the call file dials an internal context. I do some pre-processing of other data in the call file before I actually decide that I want to place the outbound call. However, I now have 'two' executions going on from this one call file. Is there something about calling an internal context from a call file that spawns two processes? |
| 19:10.12 | karleeto | [TK]D-Fender: i'll try |
| 19:10.44 | De_Mon | I classified it as a "will dirtyword kernel config work for..." |
| 19:10.54 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 19:11.20 | De_Mon | dave_mwi yes, the source AND destination |
| 19:11.52 | dave_mwi | De_Mon: is there any way to tell the two apart? |
| 19:11.54 | *** join/#asterisk Winkie (n=urmom@general-ld-220.t-mobile.co.uk) |
| 19:12.38 | dave_mwi | De_Mon: I only need one of these to execute |
| 19:13.32 | De_Mon | dave_mwi well one side is defined in the 'Channel' line of the call file and the context/priority are where the channel is sent in the dialplan. |
| 19:14.29 | De_Mon | a 'Channel' can be a Local/extension@context (not priority) |
| 19:14.53 | tzafrir_laptop | karleeto, but also note that TrixBox have a strange habbit of not working with the latest CentOS kernel |
| 19:15.13 | De_Mon | what sort of call processing are you doing thats a really weird scenario |
| 19:15.52 | shadebob | De_Mon : I will make a comilation of SIP Debug, full Log when issue occur... I will post it on my blog and continue to work on it... I will keep #informaed informed on the progress |
| 19:16.04 | [TK]D-Fender | dave_mwi, pastebin what you're doing and we'll tell you what you need to change |
| 19:16.30 | dave_mwi | [TK]D-Fender: let me give you a background on this, then I'll pastebin |
| 19:17.16 | De_Mon | pastebin the background too |
| 19:17.31 | [TK]D-Fender | dave_mwi, all the background in the world won't tell us where you recurse inappropriately and only wastes time |
| 19:17.57 | De_Mon | these xorcom units look pretty fun |
| 19:18.30 | dave_mwi | [TK]D-Fender...ok |
| 19:19.33 | shadebob | De_Mon : very good hardware ... USB 2, zaptel module, configuration like a TDM in zaptel and zapata config files... When you haven't free PCI slot it's THE solution... |
| 19:21.24 | karleeto | tzafrir_laptop: good tip, thanks |
| 19:22.06 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 19:22.17 | *** join/#asterisk Kage` (n=Kage@pool-141-153-112-141.clrk.east.verizon.net) |
| 19:25.42 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 19:36.08 | dave_mwi | De_Mon and [TK]D-Fender: http://pastebin.ca/832909 |
| 19:37.33 | [TK]D-Fender | dave_mwi, immediate "gotcha" : exten => s,2,Answer |
| 19:38.58 | [TK]D-Fender | dave_mwi, * will drop the first leg of the that call-file into the target destination immediately upon being answered. And what are you doing on the 2nd line of your context (which you're RECURSING!)... you're ANSWERING the Local channel! |
| 19:39.18 | [TK]D-Fender | dave_mwi, So as soon as the Originated end is "called" you trigger the 2nd step |
| 19:40.02 | [TK]D-Fender | dave_mwi, and you seriously shouldn't be using the same context for both legs of the call..... |
| 19:41.05 | shadebob | Thanks all for your time and your help. I will keep you informed. Bye |
| 19:44.47 | dave_mwi | [TK]D-Fender: ok, I think I get that...thanks for being patient. I'm not sure I understand how to use different contexts for the different legs...or why they are even using the same context to beging with. |
| 19:45.02 | [TK]D-Fender | dave_mwi, well... its your Call file. |
| 19:45.28 | [TK]D-Fender | dave_mwi, you have : Channel: Local/s@timedcontext/n |
| 19:45.31 | dave_mwi | [TK]D-Fender: is it the First Line and the 'Context' Line? |
| 19:45.59 | [TK]D-Fender | dave_mwi, and then : Context: timedcontext , Extension:s , Priority:1 |
| 19:46.37 | [TK]D-Fender | dave_mwi, thats like telling * to "Call John... oh and when he answers bridge him to another outbound call to... JOHN!!!!" |
| 19:46.48 | dave_mwi | [TK]D-Fender: I see |
| 19:47.11 | [TK]D-Fender | dave_mwi, Talking to yourself is a bad thing (warning to all you Bluetooth headset abusers!) |
| 19:48.06 | dave_mwi | [TK]D-Fender: hehe...seriously, thanks. That makes sense to me now. Can I remove the Context, Extension, and Priority lines - I mean as far as call file syntax goes? |
| 19:48.44 | [TK]D-Fender | dave_mwi, I'm not fully sure exactly what it is you want. I was merely explaining what it was you were doing. Maybe now you can start over again :) |
| 19:49.09 | dave_mwi | [TK]D-Fender: yeah, I understand...thanks for explaining that though. |
| 19:50.17 | [TK]D-Fender | dave_mwi, np. So.. what do you actually want to do? |
| 19:50.59 | dave_mwi | [TK]D-Fender: you want the background? |
| 19:51.28 | [TK]D-Fender | dave_mwi, the ultra-condensed version I guess. Looks like you want a call to be sent out in a time-window |
| 19:52.44 | dave_mwi | [TK]D-Fender: ultra-condensed - call files have a standard format with predefined variables. We want some control in the context to decide whether or not to actually place the call - hence I don't want the call file to initiate the call. Dialing the internal context gave me some processing lines before actually making the call. |
| 19:53.11 | [TK]D-Fender | dave_mwi, whats generating the call-file? |
| 19:53.24 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 19:54.19 | dave_mwi | It's being written by some server side scripting...coming from an application that is generating the outbound call information. |
| 19:58.08 | De_Mon | dave_mwi what happens with the call does go out? |
| 19:58.32 | tzafrir_laptop | any idea what I need to put in avahi-daemon's config so local zero-conf-enabled SIP clients will know about asterisk's SIP service? |
| 19:59.04 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 19:59.11 | hmmhesays | bah I need a new phone |
| 19:59.58 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:00.09 | dave_mwi | The call goes out and the original context routes the call to a macro. The macro does it's processing and then sents a global variable. Then the other leg sees the global var and terminates. HOwever, I used to be able to tell the two legs apart...and that's broken now with the upgrade to a new version of *. But I need to change how I'm doing this anyway. I'm sure I was taking advantage of something that was not standard to tell the two apart. |
| 20:02.24 | [TK]D-Fender | dave_mwi, you're still trying to describe the macros. Leave that out. Describe what you WANT to have happen without any description of the code & mechanisms that you think should be involved in making that happen. |
| 20:02.45 | [TK]D-Fender | hmmhesays, Ask your doctor if consumerism is right for you! |
| 20:03.50 | dave_mwi | [TK]D-Fender: I want to have a specified set of variables that can go into a call file. Then I want to be able to process those variables and decide if I should place a call or not. These variables could be things like specifying a start and end of a call window, a macro that I want the call to be routed to, or anything. |
| 20:04.38 | [TK]D-Fender | dave_mwi, First why is this process throwing out calls you don't want to place? |
| 20:05.24 | [TK]D-Fender | dave_mwi, this whole idea is dangerously "arbitrary" |
| 20:05.36 | dave_mwi | It's not necessarily that - it could be the retries inside of * that take us out of the window. |
| 20:05.53 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 20:06.21 | dave_mwi | [TK]D-Fender: It's just another level of control for us that we like to have, and let's us be flexible with our call file structure... |
| 20:07.11 | redder86 | http://bugs.digium.com/view.php?id=11141 (??!!??) http://bugs.digium.com/view.php?id=11620 Any ideas on this compatibility problem with 2.6.24 yet? |
| 20:07.26 | Qwell | redder86: 2.6.24 hasn't even been released yet |
| 20:07.32 | redder86 | so |
| 20:07.47 | Qwell | so, why would we fix something that is just going to change again? |
| 20:07.50 | *** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 20:08.19 | FuriousGeorge | hey all |
| 20:08.21 | redder86 | it is in 2.6.24-rc5 .... what makes you think that this is gonna change before the release? |
| 20:08.38 | Qwell | redder86: what changed between rc4 and rc5? |
| 20:08.57 | dave_mwi | [TK]D-Fender: what works is if I specify the Context to be 'timed-nullcontext' from that original pastebin I sent you. This is so a$$-backwards now after looking at it 2 years later. But I really don't have time to fix it now :( |
| 20:08.58 | Qwell | if the answer is not "nothing changed", then you can see the problem |
| 20:09.44 | redder86 | Qwell: is this a trick question? stuff changes. But what makes you think that this Makefile include issue is going to go away on its own? |
| 20:10.01 | Qwell | because it's happened many times before |
| 20:10.05 | hmmhesays | I'm thinking about just picking up a cheapo razr |
| 20:10.15 | redder86 | Qwell: show me |
| 20:11.16 | [TK]D-Fender | hmmhesays, Who are you with now? |
| 20:12.11 | Qwell | redder86: I'm not a kernel developer. It's a long standing policy that we don't support unreleased kernels. |
| 20:12.20 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:12.21 | tzafrir_laptop | redder86, the fact that CFLAGS gets added to the kernel module's CFLAGS is a bug. Regardless of the kernel warning |
| 20:12.47 | hmmhesays | [TK]D-Fender: alltel |
| 20:12.51 | Qwell | tzafrir_laptop: I won't argue that point. I'm just explaining why it hasn't been looked at. |
| 20:12.52 | hmmhesays | best provider in fargo |
| 20:13.02 | Qwell | tzafrir_laptop: btw, I closed the one you reported, because it was a dupe |
| 20:13.10 | redder86 | tzafrir_laptop: any idea on a quick-fix to simply let me test 2.6.24-rc5 ? |
| 20:13.16 | russellb | hmmhesays: alltel uses asterisk? :) |
| 20:13.26 | [TK]D-Fender | hmmhesays, :/ |
| 20:13.29 | hmmhesays | russellb what? |
| 20:13.38 | *** join/#asterisk Tux889 (n=Tux889@mctnnbsa28w-142167102006.pppoe-dynamic.nb.aliant.net) |
| 20:13.39 | tzafrir_laptop | I remembered there was another one, but couldn't find it |
| 20:13.39 | hmmhesays | [TK]D-Fender: they have good service around here |
| 20:13.42 | russellb | you said you were with alltel ... thought that you meant you worked for them |
| 20:13.48 | russellb | guess you mean that's just your cell provider, heh. |
| 20:13.53 | hmmhesays | oh no, I need a new phone |
| 20:13.55 | Qwell | :p |
| 20:14.00 | hmmhesays | the moto e815 is a piece of sh1t |
| 20:14.07 | redder86 | Qwell: a bug is a bug ... so simply ignoring it until release seems a bit closed-minded |
| 20:14.28 | Qwell | it not building against 2.6.24 isn't a bug until it gets released |
| 20:14.29 | Corydon76-dig | redder86: but they could change the API yet again before release |
| 20:14.35 | shido6 | hmmhesays: ( you upgrade the firmware? ) |
| 20:14.58 | redder86 | Corydon76-dig: they may, indeed. |
| 20:15.14 | hmmhesays | shido6 I did not |
| 20:15.17 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 20:15.23 | Corydon76-dig | redder86: hence why we don't support unreleased code |
| 20:15.27 | Qwell | it isn't the first time, and most definitely won't be the last |
| 20:15.35 | [T]ank | what is the name of the command that if i enter a dtmf digit, it plays it back to me? |
| 20:15.48 | russellb | echo |
| 20:16.00 | redder86 | Qwell: when was the previous time? |
| 20:16.13 | Qwell | again - I'm not a kernel dev, I just know that they have many times |
| 20:16.27 | [T]ank | russellb: sorry.. is that a command for extensions.conf? I wasnt clear on what I was asking for |
| 20:16.29 | tzafrir_laptop | Qwell, no point in keeping it closed: you'll only get more dupes |
| 20:16.40 | russellb | redder86: it's a waste of time trying to support unreleased versions of the kernel ... and I don't want him wasting even more time looking up history that won't help anything |
| 20:16.42 | [T]ank | i want something like say digit, but I want it to tell me what button i pressed on the phone. |
| 20:16.44 | tzafrir_laptop | mind if I reopened it? |
| 20:16.47 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:16.59 | russellb | it's not getting messed with until it's a problem with a released version |
| 20:17.31 | russellb | reopening is fine if you have a patch sitting there you can point people to |
| 20:17.36 | redder86 | so a bug's not a bug until it's a bug with a released kernel? |
| 20:17.40 | russellb | but i don't want anything committed until it's released |
| 20:17.42 | hmmhesays | I don't think firmware is going to fix this phone |
| 20:17.44 | russellb | correct |
| 20:17.53 | De_Mon | dave_mwi pretend you don't know what a pbx is and have come in here trying to find out if "what you want to do" is possible. don't use any asterisk terms like macro and context and variable in your definition of "what you want to do". go! |
| 20:17.57 | tzafrir_laptop | The bug report I had suggested some workarounds, actually |
| 20:18.11 | Corydon76-dig | redder86: it's not a bug until the code is released |
| 20:18.36 | tzafrir_laptop | it is "released": in the kernel of current Ubuntu |
| 20:18.39 | [TK]D-Fender | hmmhesays, I beg to differ on the E815..... but what do you have now, and what do you want out of your phone? |
| 20:18.39 | redder86 | Well, I beg to differ. But I guess you'll get to see this fixed in a month or two or whenever. |
| 20:18.45 | De_Mon | so far I've got make a call (from -- to?) bla bla time window |
| 20:19.09 | hmmhesays | [TK]D-Fender I have an e815 and I can't get more than 4 hours of standby out of it |
| 20:19.10 | [TK]D-Fender | [T]ank, Go make a simple IVR for this. |
| 20:19.26 | redder86 | As far as I'm concerned if the kernel-dev people cut an "rc" that's a release ... someone there "cut" it. |
| 20:19.28 | [TK]D-Fender | [T]ank, "show application saydigit" |
| 20:19.50 | [TK]D-Fender | hmmhesays, then you're battery is screwed. I had mine for 2 years and it lasts a week easily. |
| 20:20.06 | russellb | ok, then i'll put it this way ... if it's a bug against an rc, i consider it a lower priority than the other 400 things on the bug tracker. if it's against an official release, then i would consider it more important |
| 20:20.08 | cpm | sounds like dead battery |
| 20:20.09 | hmmhesays | [TK]D-Fender: negative have had multiple batteries exhibit the same behavior |
| 20:20.10 | redder86 | and if some distro pulls in various patches and such to make their kernel somewhat different than the vanilla... well, that's a release too |
| 20:20.27 | [TK]D-Fender | hmmhesays, is your phone under warrantyA? |
| 20:20.28 | hmmhesays | its definately the phone itself |
| 20:20.36 | [TK]D-Fender | hmmhesays, could be you phone is screwed..... |
| 20:20.37 | hmmhesays | no, I'm almost two years out on it |
| 20:20.46 | redder86 | russellb: I understand priorities... just right now this is my priority. |
| 20:20.50 | [TK]D-Fender | hmmhesays, ok, but its not the model, its jsut yours.... |
| 20:20.57 | hmmhesays | the phone is screwed and alltel is giving away razr's now |
| 20:20.58 | tzafrir_laptop | russellb, fine, let's not make it a blocker. But it is an open issue |
| 20:21.08 | russellb | fair enough |
| 20:21.10 | [T]ank | [TK]D-Fender: isnt saydigit just for playing back digits already entered in with this command. I seem to remember that I could just use a command and when I called into the system and entered dtmf it would tell me the number I pressed. |
| 20:21.15 | [TK]D-Fender | hmmhesays, so what do you want out of a phone beyond an appropriate battery life? |
| 20:21.17 | [T]ank | i just cant remember how i did it. |
| 20:21.48 | hmmhesays | [TK]D-Fender not much, camera, mp3 player |
| 20:21.50 | [TK]D-Fender | [T]ank, No, there isn't. Make an IVR and say back the digit you entered into it and loop. This is 10 lines of dialplan max. |
| 20:22.03 | hmmhesays | decent web browser |
| 20:22.18 | [TK]D-Fender | hmmhesays, Sony 810i was decent when I was helping a friend shop for one. |
| 20:22.25 | [TK]D-Fender | hmmhesays, WEB is another matter. |
| 20:22.53 | hmmhesays | openwave blows goats |
| 20:22.59 | [TK]D-Fender | hmmhesays, I love my HTC Touch, but its the data plabn that makes it rock. the best choice is therefore incredibly localized... |
| 20:23.13 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 20:23.45 | hmmhesays | yeah |
| 20:24.30 | *** join/#asterisk luke-jr (n=luke-jr@wsip-70-167-147-10.om.om.cox.net) |
| 20:26.10 | [TK]D-Fender | hmmhesays, $37+tax = 250 DT, Unlim eve/wk @6pm, CID, VM, Unlim Data, no SAF |
| 20:26.49 | hmmhesays | what provider? |
| 20:26.57 | [TK]D-Fender | hmmhesays, Bell |
| 20:29.20 | hmmhesays | nice |
| 20:31.51 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 20:32.21 | hmmhesays | well I'm going to the store to browse I think |
| 20:33.06 | [TK]D-Fender | hmmhesays, shop the plans first. See which are limited to certain models. Then check what the phone offers vs your wants. |
| 20:34.16 | hmmhesays | [TK]D-Fender: of course |
| 20:34.32 | hmmhesays | it sucks that alltel is the best carrier around here. |
| 20:34.47 | hmmhesays | to get what I have now through sprint or verizon would be almost double the cost |
| 20:34.52 | Havokmon | Anyone try using Inter-Tel 8622 phones with Asterisk? I can't get the damn thing to dial properly, but it'll receive calls :/ |
| 20:35.25 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:37.15 | [TK]D-Fender | Havokmon, what do you get on sip debug for an attempted outbound call? |
| 20:37.46 | Havokmon | nada. It doesn't seem to ever 'leave' the phone :/ |
| 20:37.58 | Havokmon | It's a phone issue for sure. |
| 20:40.43 | [TK]D-Fender | Havokmon, well I can tell you we only hear mention of Intertel around here maybe half a dozen times a year. And and of those none with the feeling of "Oh and I got it all working great" |
| 20:40.57 | Havokmon | lol |
| 20:42.09 | Havokmon | Unfortunately it's what I happen to have around here, and I'm already being sneaky just trying bring in asterisk when corp has other ideas.. |
| 20:42.15 | *** join/#asterisk techie (n=techie@adsl-76-214-31-16.dsl.lsan03.sbcglobal.net) |
| 20:42.17 | *** part/#asterisk techie (n=techie@adsl-76-214-31-16.dsl.lsan03.sbcglobal.net) |
| 20:42.49 | [TK]D-Fender | Havokmon, I can relate. My head office nearly went that route.... |
| 20:43.00 | [TK]D-Fender | Havokmon, then they went with an Avaya IP Office instead :) |
| 20:44.34 | Havokmon | I have no idea what's going to happen. I have an old Fujitsu 9600, and I'm using asterisk to add features to it.. corp has/had Intertel, but with the sale of Intertel to Dell apparently they're flipping out about something and want to go Cisco |
| 20:45.37 | Havokmon | someone in upper management apparently thinks swapping phone systems is like swapping a dumb terminal |
| 20:46.38 | dave_mwi | De_Mon: Can I using asterisk be able to tell if a call should actually go out by looking at variables in a call file? |
| 20:47.10 | [TK]D-Fender | dave_mwi, wanna restructure that into an actual sentence? :) |
| 20:47.34 | De_Mon | dave_mwi whats a call file? stop using these words I don't know! |
| 20:48.01 | De_Mon | use application language not implimentation language |
| 20:48.32 | [TK]D-Fender | De_Mon, umm... a "call file" is a very specific Asterisk term :) |
| 20:48.33 | De_Mon | I need ASTERISK to CALL a number provided by an external application and DO SOMETHING unless SOMETHING ELSE happens |
| 20:48.45 | De_Mon | oops without the word asterisk |
| 20:49.40 | [TK]D-Fender | dave_mwi, and the variables you set in your call-file I believe only apply once the call is answered and is bridged to the dialplan into the context/ext/prio speified |
| 20:50.44 | De_Mon | you can set variables in the .calfile and use them to dial a number before the channel is answered, but I still don't know why a call file is even being used. say the call goes out, whats going to happen one side of the call is already in lala land |
| 20:50.47 | dave_mwi | I want to use asterisk to determine whether or not a call should be placed from a call file. Is this possible? |
| 20:51.16 | De_Mon | still no details and using implimentation language. You suck as a customer I quit |
| 20:51.47 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 20:51.58 | dave_mwi | De_Mon: I'm not a first time customer. You can't be that basic with me. |
| 20:52.13 | dave_mwi | De_Mon: and I'm a developer, not a business type. |
| 20:52.31 | dave_mwi | De_Mon: Can I call people with Asterisk? |
| 20:53.10 | russellb | asterisk has nothing to do with phone calls ... |
| 20:53.12 | dave_mwi | [TK]D-Fender: no, the variables apply before the bridge. The variable determine if the bridge will be made. |
| 20:53.14 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.99.127) |
| 20:53.33 | De_Mon | dave_mwi this came up because you wern't using the call file correctly. Its still not clear what the hell your trying to do but without details all we can do is say, it should work. |
| 20:54.18 | dave_mwi | De_Mon: I can't give you details without giving you specifics. You want something I can't give you based on your requirements. You suck as a vendor. I'm going somewhere else. ;) |
| 20:56.00 | dave_mwi | De_Mon: How's this: I want to make outbound phone calls with asterisk. I want asterisk to be able to decide if that outbound phone call should be made based upon a specified call time window. |
| 20:56.18 | dave_mwi | De_Mon: can I do that? |
| 20:56.48 | [TK]D-Fender | dave_mwi, Yes, go right ahead. |
| 20:57.05 | De_Mon | wheres that picture of a swing thats going through the tree instead of hanging off a branch its a great illistration of this conversation. |
| 20:57.29 | dave_mwi | De_Mon: Give me an example of what you expect my question to be like. |
| 20:57.43 | dave_mwi | Specifics, but no names specific to the technology. |
| 20:57.55 | De_Mon | I need ASTERISK to CALL a number provided by an external application and DO SOMETHING unless SOMETHING ELSE happens |
| 20:58.15 | dave_mwi | De_Mon: What kind of use would I get out of a question like that? |
| 20:58.17 | dave_mwi | 'yes' |
| 20:59.10 | dave_mwi | De_Mon: Can * make an outbound call any other way other than a call file. That's my first question and it should be pretty simple. |
| 20:59.21 | De_Mon | its how you describe a feature you want implimented without requiring callfiles or macros |
| 20:59.47 | [TK]D-Fender | dave_mwi, is the 1st leg of your call-file processing the way you want yet? |
| 21:00.10 | dave_mwi | [TK]D-Fender: It is. And so is the second. |
| 21:00.27 | [TK]D-Fender | dave_mwi, So everything is working? |
| 21:01.19 | dave_mwi | [TK]D-Fender: It is. It was the explanation of the call file that you and went through. Which I wouldn't have been able to get without giving any specifics as De_Mon wants. I'm just trying to do it his way now to see if I learn anything else new. It's an exercise in humility on my part. |
| 21:04.10 | [TK]D-Fender | dave_mwi, Your workding is very vague and sometimes backwards and based on what you might ask it could be that youwould refer to one implementations tools (like call-files), where this new scenario you want to solve would not even be well severved by using them. |
| 21:04.35 | [TK]D-Fender | dave_mwi, like the saying goes : if the only tool you have is a hammer, every problem begins to look like a nail. |
| 21:05.34 | dave_mwi | [TK]D-Fender: I understand that, but I would think that a question like: Is there any other way to place outbound calls with asterisk other can call files (which by the way I'm very interested in still hearing an answer about)? Would be a very straightforward question. |
| 21:07.40 | De_Mon | I just dug up the pastebin and got the missing piece of information where you are dialing the number and playing an audiofile (thru dial) |
| 21:08.04 | [TK]D-Fender | dave_mwi, allow me to rephrase your question : What other ways does * have to INITIATE an outbound call (not being prompted by something like a call coming in from a phone, or other link). |
| 21:08.28 | dave_mwi | [TK]D-Fender: ok, that works. |
| 21:08.30 | [TK]D-Fender | dave_mwi, And yes, there is anotehr way : Aside from call-files, there is the AMI "Originate" command which does jsut about the same thing. |
| 21:09.58 | [TK]D-Fender | dave_mwi, Now aside from the tool that will trigger the call to be placed, you have several conditions to apply to whether or not to even follow through. This is the part where perhaps you should think about WHEN this decision should have been made. |
| 21:10.03 | dave_mwi | [TK]D-Fender: Ok. how would I invoke such a command from an external application? |
| 21:10.38 | [TK]D-Fender | dave_mwi, Go read about AMI in the BOOK or WIKI. |
| 21:11.39 | [TK]D-Fender | dave_mwi, You clearly have a process creating this call-file already. now why would you make it *'s job to check the time when you could just as easily made the choice not to issue the call-file at all? |
| 21:12.22 | [TK]D-Fender | dave_mwi, Ask yourself when these decisions are best made and what processes should be responsible for the "bigger picture". |
| 21:12.27 | dave_mwi | [TK]D-Fender: I understand what you're getting at and we do that already - but this is just another level of control that was built in because some higher-ups wanted to have it. |
| 21:13.13 | dave_mwi | [TK]D-Fender: I'm looking at the AMI right now. We already use this to monitor some other things with Asterisk, so I'll look into using it for initiating outbound calls. |
| 21:15.45 | [TK]D-Fender | dave_mwi, It isn't very different really, just another means that can work over a network without file-access, etc/ |
| 21:15.50 | De_Mon | the thing missing piece to my puzzle was how you were doing anything with the call with both legs of the call going to the same place. To which the answer is a macro that plays audio before connecting to the calling channel |
| 21:16.13 | [TK]D-Fender | dave_mwi, and when higher-ups invent redundency like that, they should get out of the way and let coders do their work. |
| 21:16.47 | De_Mon | wouldn't it be funny if the time on one of those systems was off by an hour or two and no calls would go out |
| 21:16.59 | [TK]D-Fender | dave_mwi, just tell them "You wanted a check for time, you got it. Now don't tell me where in my code I have to PUT it, and don't tell me to repeat it 20 times for nothing" |
| 21:17.10 | [TK]D-Fender | De_Mon, Comedy gold. |
| 21:18.26 | De_Mon | dave_mwi were i doing this, the calling channel would do just that, and when the call connected (to the other context/extension/priority in the call file) play the audio. |
| 21:19.19 | De_Mon | and if both systems are checkign the time, get rid of it in asterisk. 0 macros two easy to understand call legs and less variables |
| 21:24.56 | [TK]D-Fender | dave_mwi, Oh and at a bare minimum you'll want to drag your code up to at LEAST full * 1.2 spec before you cause yourself more migration headaches |
| 21:25.27 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 21:28.45 | lirakis | anyone use sipfly for a service provider? .. ive got a res. provider now.. im looking for another outbound provider.. no did req. byod, preferably unlimited us & can. min. SipFly seems to be one of the few that you can just get channels./ports from at the retail level. |
| 21:29.11 | dave_mwi | I'm glad at least that I brought some comedic relief the room. |
| 21:29.13 | lirakis | fyi im already on voip-info's service providers page.. so no need to post that link |
| 21:29.17 | [TK]D-Fender | lirakis, Would you like fries with that, sir? |
| 21:29.54 | anonymouz666 | anyone know a populated SQL table that contains international country codes and area codes? |
| 21:29.58 | lirakis | <PROTECTED> |
| 21:30.26 | [TK]D-Fender | dave_mwi, We have mercestes for that already.... your's was more of an exercise in patience and in teaching an approach, not just a solution. |
| 21:30.31 | lirakis | id go to my current provider and just ask for another channel... but id rather have "redundancy" accross another carrier |
| 21:31.01 | [TK]D-Fender | lirakis, This is the part where you hopefully begin to realize you are a demographic the market-place doesn't care about :) |
| 21:31.51 | lirakis | [TK]D-Fender: .. right.. well ive been well aware of that for some time... i void warranties by looking at things |
| 21:32.34 | lirakis | okay.. well ill look more tomorrow |
| 21:32.36 | lirakis | later all |
| 21:32.40 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:35.38 | ManxPower | Most service providers have per-min, no monthly fee. Where the hell was lirakis looking? |
| 21:36.02 | ManxPower | (assuming you don't need a DID) |
| 21:36.46 | [TK]D-Fender | ManxPower, "im looking for another outbound provider.. no did req. byod, preferably unlimited us & can. min." <-- |
| 21:37.16 | Maliuta | peenytel do a flat-rate landline (to a heap of countries) with per minute mobile rates and a DID included |
| 21:37.18 | ManxPower | [TK]D-Fender: I said where, not what. |
| 21:37.28 | ManxPower | Teliax, Nufone off the top of my head. |
| 21:37.49 | ManxPower | of and vitelity |
| 21:37.51 | Maliuta | [TK]D-Fender: pennytel have $0.08 untimed to the US and Canada |
| 21:38.10 | Maliuta | [TK]D-Fender: and Australia, UK .... |
| 21:38.26 | [TK]D-Fender | ManxPower, I think you've missed this a few times. He doesn't want per minute charges. |
| 21:38.39 | dave_mwi | [TK]D-Fender: yes, I know you were trying to teach me an approach. And I know what you're saying - there's no sense in fixing something that's fundamentally broken already. But I have what's before me now, and I have to fix it now. I'd actually much rather fix it and have it break again. Because then I would have a basis for rebuilding it - because they keep wanting me to fix something that is fundamentally broken. |
| 21:38.46 | jwh | £0.025 to Australia |
| 21:38.51 | ManxPower | [TK]D-Fender: oh, so he basically wants all his calls to be free? What a moron. |
| 21:38.55 | jwh | and mm, £0.01 to US |
| 21:39.18 | jwh | possibly less, cba to open my rate sheets |
| 21:39.18 | [TK]D-Fender | ManxPower, no, not free, just a fixed rate for the channel, no extra DID charge, just pure flat-rate termination. |
| 21:39.26 | jwh | (me pays the same as customers) |
| 21:39.42 | ManxPower | Lets see. Teliax has that at least. A softcap of 2,500 mins, of course. |
| 21:39.43 | [TK]D-Fender | ManxPower, AKA $20 to call county X with 1 channel with no DID for callback, etc.... |
| 21:40.18 | ManxPower | *sniff* Smells like a telemarketer to me. |
| 21:40.58 | Maliuta | [TK]D-Fender: so not even a per call costing |
| 21:41.03 | [TK]D-Fender | dave_mwi, thats fine. You say you've already fixed it following the points we've highlighted already so I guess you're on you way AND a bit wiser at the same time. |
| 21:41.32 | [TK]D-Fender | Maliuta, And to think I gained that understanding by merely reading what he said! Its insanity! |
| 21:42.10 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 21:50.12 | *** join/#asterisk Tux889 (n=Tux889@mctnnbsa28w-142167102006.pppoe-dynamic.nb.aliant.net) |
| 21:50.12 | *** join/#asterisk The-Bat (i=for-kart@221.134.197.160) |
| 21:56.56 | Yourname` | This is really weird. |
| 21:56.57 | Yourname` | Dec 26 16:57:17 ERROR[3016]: app_dial.c:1712 retrydial_exec: RetryDial requires the numerical argument <sleep> |
| 21:57.11 | Yourname` | exten=>_NXXNXXXXXX,n,RetryDial(/var/lib/asterisk/sounds/connecting,0,3,SIP/${EXTEN}@outbound,,tT) |
| 21:57.37 | Yourname` | What am I doing wrong? I made sure the 0 is a zero. |
| 21:58.20 | Yourname` | Ver 1.2.20 |
| 22:00.10 | Yourname` | Hmm, it doesn't take 0 for a value. Now, I don't get the error.. but it doesn't work like it should either. |
| 22:09.10 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 22:09.13 | teknoprep | hey all |
| 22:09.15 | teknoprep | having a problem |
| 22:09.25 | teknoprep | i have tryed many different types of dtmfmodes |
| 22:09.33 | Yourname` | Why does the first part of a playback file not get played at times? |
| 22:09.44 | teknoprep | but either have have nothing when i press a # or i have it play a tone like 5 times really fast |
| 22:10.40 | ManxPower | Yourname`: usually because of a reinvite happening |
| 22:11.15 | Yourname` | On a retydial? |
| 22:11.19 | Yourname` | retry* |
| 22:11.29 | file | .0 |
| 22:11.31 | ManxPower | "Yourname`: Why does the first part of a playback file not get played at times?" Answer: Because a reinvite is happening |
| 22:12.15 | Yourname` | ManxPower: Isn't it better to have canreinvite=yes in NAT situations? |
| 22:12.45 | ManxPower | Yourname`: What crazy universe do you live in? NAT is not compatible with reinvites in most cases |
| 22:13.19 | teknoprep | well on your provider you can do that |
| 22:13.29 | teknoprep | if you setup 1:1 nat with alot of open ports |
| 22:13.32 | ManxPower | teknoprep: if the phone DTMF mode is different from Asterisk's DTMF mode, you will get weird things happening. Also remember that INBAND DTMF only works with ulaw/alaw |
| 22:13.46 | teknoprep | ManxPower yeah i know |
| 22:13.53 | teknoprep | what about my provider info? |
| 22:13.54 | caio1982 | ManxPower: any technical reason on why the playback is not delayed until the reinvite is gone? |
| 22:14.08 | teknoprep | what should i have for dtmfmode= on the sip channel to my provider |
| 22:14.19 | ManxPower | teknoprep: I have no idea what DTMF mode your provider uses, but don't expect anything to work until you know what DTMF mode your provider uses. |
| 22:14.34 | teknoprep | rfc2833 |
| 22:14.53 | ManxPower | caio1982: the audio can be missed while the reinvite is happening. I don't know why. |
| 22:14.59 | ManxPower | teknoprep: and you are SURE of that? |
| 22:15.07 | teknoprep | yes |
| 22:15.11 | teknoprep | http://www.bandwidth.com/wiki/article/Configuring_Asterisk_for_bandwidth.com |
| 22:15.24 | ManxPower | Maybe you did something very, very stupid like t or T on the Dial line. |
| 22:15.38 | teknoprep | let me check |
| 22:15.55 | teknoprep | nope |
| 22:15.59 | teknoprep | just on inbound calls |
| 22:16.03 | ManxPower | teknoprep: providers are frequently wrong. |
| 22:16.23 | teknoprep | i still believe its internal |
| 22:16.33 | ManxPower | teknoprep: define "internal" i.e. phones? |
| 22:16.37 | teknoprep | yeah |
| 22:16.59 | ManxPower | so we go back to my original statement, if Asterisk and your phones disagree on the DTMF mode, it's not going to work |
| 22:17.01 | teknoprep | i have a linksys spa962 and 12 cisco 7940's |
| 22:17.14 | teknoprep | if i get the 7940's to work the spa962 stops working |
| 22:17.16 | teknoprep | i don't get it |
| 22:19.02 | Yourname` | Let me try, ManxPower.. one sec |
| 22:21.15 | Yourname` | Hmm, didn't work. Said "ting" instead of "connecting", lol |
| 22:37.01 | *** join/#asterisk Tux889 (n=Tux889@mctnnbsa28w-142167102006.pppoe-dynamic.nb.aliant.net) |
| 22:45.55 | De_Mon | Yourname` you also need to Answer() the channel and Wait(1) a second before playback to ensure everything makes it into the channel |
| 22:46.01 | De_Mon | Wait(.5) might work too |
| 22:47.15 | De_Mon | dave_mwi it was mostly my fault for not actually reading the pastebin and expecting you to tell me what it was supposed to be doing |
| 22:47.47 | De_Mon | hopefuly we both learned something from that exchange ;) |
| 22:56.01 | *** join/#asterisk RoyK (n=roy@ip-113-4-149-91.dialup.ice.no) |
| 23:01.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:01.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:02.24 | hmmhesays | [TK]D-Fender you around? |
| 23:02.40 | [TK]D-Fender | hmmhesays, uyp |
| 23:04.06 | hmmhesays | I'm trying to disable evdo on my d815 |
| 23:04.10 | hmmhesays | *e815 |
| 23:04.47 | hmmhesays | you ever do this? |
| 23:05.58 | *** join/#asterisk marlow (n=marlow@loke.sca.airwire.ie) |
| 23:05.58 | [TK]D-Fender | hmmhesays, well I've disabled the browser, would that do it? |
| 23:08.00 | hmmhesays | nevermind I just got it to work |
| 23:11.27 | hmmhesays | holy crap you could fsck your phone up in here |
| 23:12.47 | *** part/#asterisk dave_mwi (n=dave_mwi@208.251.184.82) |
| 23:15.28 | *** join/#asterisk MrFollies (n=Miranda@60-242-243-193.static.tpgi.com.au) |
| 23:17.01 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 23:18.09 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 23:20.52 | *** part/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 23:20.59 | MrFollies | Are there any FreeBSD asterisk users around today? I have some quesions about upgrading from 1.2.9 to 1.4.13 |
| 23:22.45 | tzafrir_laptop | MrFollies, I'm not one, but I suppose you btter just ask |
| 23:24.05 | MrFollies | Yeah I guess :) |
| 23:24.52 | MrFollies | I have been using 1.2.9 for some time now. I'd like to upgrade to 1.4.13 (the latest in the freeBSD ports). First off I was wondering if I need to migrate my config. |
| 23:25.57 | MrFollies | Secondly (and FreeBSD specific) I was wondering if it's possible to build asterisk and test without installing. That is easy for the actual package, but dependencies worry me... |
| 23:27.59 | De_Mon | MrFollies you will need to migrate your configs, see the tarball docs for specifics |
| 23:29.37 | MrFollies | Ok. I assume if I just upgrade to 1.2.24 (the latest 1.2 for FreeBSD) then the configs can stay? |
| 23:30.19 | *** join/#asterisk RoyKa (n=roy@ip-86-1-149-91.dialup.ice.no) |
| 23:30.29 | ManxPower | MrFollies: read the UPGRADE.txt file for BOTH 1.2 and 1.4 if you are going to upgrade to 1.4 |
| 23:31.06 | De_Mon | (things depriciated in 1.2 are gone in 1.4 for example) |
| 23:32.17 | MrFollies | Yeah. |
| 23:32.59 | MrFollies | I am trying to have a rollback procedure. My * is production and I do not have a test server. |
| 23:34.50 | hmmhesays | heh 13k does sound a lot better than evrc on my cell |
| 23:37.20 | *** part/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 23:39.09 | [TK]D-Fender | hmmhesays, ? |
| 23:42.34 | hmmhesays | [TK]D-Fender: codecs on the e815 |
| 23:42.55 | [TK]D-Fender | hmmhesays, never knew you could set them.. GSM or CMDA? |
| 23:43.05 | hmmhesays | cdma |
| 23:43.13 | [TK]D-Fender | hmmhesays, where do you drill to? |
| 23:43.32 | hmmhesays | menu button 073887* |
| 23:43.59 | *** join/#asterisk bmcghee`home (n=brent@209.250.225.2) |
| 23:44.03 | [TK]D-Fender | and then a code? |
| 23:44.07 | hmmhesays | 000000 |
| 23:44.10 | hmmhesays | six 0's |
| 23:44.20 | [TK]D-Fender | hmmhesays, mine seems locked |
| 23:44.30 | hmmhesays | what did it say? |
| 23:44.36 | [TK]D-Fender | hmmhesays, says invalid |
| 23:44.46 | bmcghee`home | asterisk wont accept the inbound call from the number, it says its not inservice, and on hold music starts then stops instantly |
| 23:44.49 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 23:45.15 | hmmhesays | after you entered the security code? |
| 23:45.25 | [TK]D-Fender | hmmhesays, yes, says the code is incorrect |
| 23:46.34 | hmmhesays | hmm I saw an alternate code when I was researching this |
| 23:46.35 | hmmhesays | hold on |
| 23:47.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:48.08 | hmmhesays | 236197 |
| 23:48.35 | [TK]D-Fender | yup |
| 23:48.36 | [TK]D-Fender | in |
| 23:49.25 | [TK]D-Fender | hmmhesays, I stee stuff, but can't set the vocoder. |
| 23:49.35 | [TK]D-Fender | hmmhesays, ANOTHER PW. and none work |
| 23:50.12 | hmmhesays | mine lets me change everything in that menu |
| 23:50.54 | hmmhesays | but you can see the vocoder menu? |
| 23:51.16 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 23:51.27 | [TK]D-Fender | hmmhesays, yes, the option to enter it, but every option in the main is PW'd |
| 23:55.46 | hmmhesays | might have to search on that one |