| 00:01.05 | p3nguin | Anyone here use Whistle and know how to register a second account? I can't figure out how to create a second account using the same client. |
| 00:01.49 | p3nguin | I don't need to use both accounts on the single client at the same time, but I do need to create the account. |
| 00:11.01 | *** join/#asterisk Moe__ (~MJames@c-71-62-116-235.hsd1.va.comcast.net) |
| 00:13.48 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 00:30.58 | p3nguin | Terms of Service does not permit it, so nevermind. |
| 01:11.22 | *** join/#asterisk tomaw_ (tom@freenode/staff/tomaw) |
| 01:30.28 | WIMPy | Wasn't there a dahdiras or something to terminate data calls? But I guess that's only for X.75 or something. |
| 01:52.16 | g_r_eek | can i ask a question about freepbx in here? i try to use the endpoint conf. mannager, i set up yealink phones inside there but i see my tftp directory is empty , is there another script that i need to do to populate it with the conf files? |
| 02:02.59 | WIMPy | That's definitely something for #freepbx |
| 02:03.15 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 02:19.44 | showme | kage: I don't really have any idea who might want to do this, in theory it sounds like a pretty simple piece of software, but I'm just not a programmer, and to find someone willing to write it, would probably be someone that also has the need for it. |
| 02:21.05 | WIMPy | Not at all. It's a very complex thing to do a modem. That's why you hardly find any non-commercial attempts. |
| 02:22.43 | coppice | that depends on whether you want to keep it legal, and whether you really want a 64 bit vesion |
| 02:22.57 | coppice | although doing a SIP anything is a PITA |
| 02:24.51 | showme | If you're going to support v.92 and everything, sure, I'd agree with you, but for a v.32 (9600) it should be fairly simple. Back 10 years ago, most of the pci modems you bought were really just a sound card interfaced to the phone line and the "modem" was pure software, a "sip modem" should be as simple as just the software piece of that and a bit of network code, no hardware needed. |
| 02:25.56 | coppice | showme: now describe just how simple a V.32 modem is |
| 02:26.26 | WIMPy | Thst has been commecial. And I have no idea, how well they worked. |
| 02:26.29 | showme | I really wonder if some source code for those older modems could be found, if it wouldn't be fairly easy to adapt. |
| 02:26.54 | showme | v.32 is very well documented, and unlike newer stuff like v.90 and v.92, its not tied up with patents and legal junk. |
| 02:26.55 | WIMPy | showme: Do it and become famous :-) |
| 02:27.23 | showme | if I were a programmer, I would, I'm strictly a networking guy. |
| 02:27.51 | showme | Not sure how you become famous with a piece of software that only about 15 people in the world would use... |
| 02:28.35 | coppice | if you were a programmer you wouldn't have a clue where to start on a V.32 modem, unless you were also a DSP expert |
| 02:28.57 | WIMPy | You should get an old NAS, if you're luck you find one for a tenner on ebay. |
| 02:29.16 | WIMPy | +y |
| 02:29.38 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 02:29.40 | showme | I might give you that, but if you could find the source code for one of the old pci modems, most of that code would be there, just add networking code and you're done.. |
| 02:30.06 | showme | although I'm not sure if there were a bunch of v.32 software modems, they didn't really seem to get too popular until the 14.4 or even the 28.8 days.. |
| 02:30.37 | coppice | V.32 was the 14.4k modem |
| 02:31.21 | showme | and for everything faster, software modems dominated the pci (internal) market, but that would be a bunch of code that probably wouldn't work. |
| 02:31.37 | showme | actually, v.32bis was 14.4, v.32 never addrressed anything faster than 9600 |
| 02:31.48 | coppice | The V.17 modem in spandsp is basically half a V.32 modem. You would need to add an echo canceller and an elaborate startup procedure. |
| 02:32.11 | coppice | V.32bis is just the updated V.32 |
| 02:32.44 | coppice | and V.17 is the 14.4 variant |
| 02:33.21 | showme | more technically, v.32bis was an extention to v.32, not just an "update", that would infer any 9600 modem could be updated with new software, definately not the case. |
| 02:34.22 | coppice | if you look at the softmodem packages for Linux, like slmodem, you can actually fudge those to work with things like Asterisk and Freeswitch with a little DSP shim in the way |
| 02:34.44 | showme | v.17 is a fax standard for 14.4, faxing is a half duplex activity |
| 02:35.01 | coppice | showme: so you believe version 1 of anything can do 100% of what version 2 can? |
| 02:35.17 | coppice | as I just said, V.17 is *half* a V.32bis modem |
| 02:35.54 | showme | no, but in the case of v.32 and v.32bis, they really are different things, I argued 20 years ago that I didn't think it was appropriate to use the term v.32bis.. |
| 02:36.27 | coppice | V.32bis is just a V.32 modem stretched |
| 02:36.32 | showme | faxing and modem communication, although similar, really are different beasts.. |
| 02:37.13 | florz | showme: coppice is the author of spandsp, you know ... |
| 02:39.37 | showme | then he'd be the ideal guy to write a v.32 sip modem.. |
| 02:40.58 | coppice | V.32 just isn't interesting. V.34 half duplex for FAX will be interesting in 3 or 4 years when the patents expire |
| 02:42.14 | showme | I don't have a need for faxing, but in my search for a sip modem, I did run across a couple sip fax modems... I thought one I saw was open source, but I'd have to dig again to find it.. |
| 02:44.34 | showme | v.32 has a very practical application though, tons of equipment (cisco for example) that has a serial console port that at default runs at 9600, thats one of the only reasons I keep an analog line in my home office, so the 3 times a year I need to dialup to something like that, I still can. If I could do that over my voip, that would be a killer good thing for me, save me about $300/year, and just make life simpler. |
| 02:45.36 | coppice | wouldn't a TCP/IP<->serial gateway be more useful for that? |
| 02:46.33 | showme | the reason I'd dialup to a router like that is if the outside IP link is down, so no, once IP dies, so does my gateway. |
| 02:46.57 | coppice | in that case you still need the analogue line |
| 02:47.05 | showme | I'm getting in to check the T3 or whatever, becaues I can't get IP to the unit. |
| 02:47.32 | showme | Yeah, every where I've deployed such a unit, I require the customer to keep an analog line for that very purpose... |
| 02:47.45 | showme | I want the sip modem for my end, not theirs |
| 02:49.13 | showme | plus every now and then I come across an old bbs or something that doesn't seem to have telnet or ssh.. |
| 02:50.03 | showme | our city government even has a dialup (bbs style) for getting some city info, although "most" of that has finally migrated to the web. |
| 02:50.30 | WIMPy | showme: Where in the world do you live? |
| 02:50.56 | showme | I'm in the USA, a town in Missouri |
| 02:53.02 | WIMPy | They knew BBSs? Cool. |
| 02:53.09 | showme | the biggest thing on the city dialup thats not on the web is a complete phone listing for every city employee |
| 02:53.26 | showme | Yeah, the whole bbs thing was up in the early 80's before internet was even a thought |
| 02:53.34 | showme | its just never been shut down. |
| 02:54.05 | showme | its probably hitting 40 years old |
| 02:54.16 | xxiao | plan to buy an ATA then setup asterisk at home, any recommendation on voip service provider? ooma(?), vonage? |
| 02:55.07 | showme | vonage doesn't allow you to connect directly to them via sip, only preprgrammed devices, |
| 02:55.07 | xxiao | both provides ATA i think, but i'd like to use vendor-agnostic ATA with asterisk instead |
| 02:55.07 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 02:55.29 | coppice | the first BBSes appeared around 1980, so its more like 30 than 40 years |
| 02:55.48 | xxiao | i.e. a voip service provide that has PSTN gateway worldwide and i can register/use without getting devices from them |
| 02:56.09 | showme | xxiao: it really depends on your usage, how much time do you use the phone, a lot of long distance or international? |
| 02:56.37 | showme | coppice: your right, bad math on my end, but like I said, early 80's.. |
| 02:56.41 | xxiao | showme: about 200-300 minutes international call each month, same for national long distance |
| 02:57.18 | xxiao | i need it to block calls at certain time to avoid the ring waking my kids at nap/sleep |
| 02:57.32 | WIMPy | Many countries or mainly one? |
| 02:57.49 | coppice | the word "community" is thrown around wildly these days, but early BBSes had a real sense of community |
| 02:58.05 | xxiao | i'm using onesuite for international calls |
| 02:58.14 | xxiao | WIMPy: half europe, half asia |
| 02:58.26 | coppice | like my kids :-) |
| 02:58.37 | WIMPy | Then get an ITSP in europe and one in asia. |
| 02:58.57 | showme | I used to have a 4 line bbs, it was definitely a community, we had a couple picnics a year and everything. |
| 02:59.15 | WIMPy | Those were the days... |
| 03:00.10 | xxiao | showme: nowadays it's twitter/facebook/sms and people hate to see each other |
| 03:01.08 | xxiao | WIMPy: thanks. the "popular" voip service providers are all supplying their equipments while i don't want to be locked in by that |
| 03:01.39 | xxiao | guess i'm just looking for a company has lots of PSTN gateway and SIP proxy/registra servers |
| 03:02.05 | WIMPy | I don;t know any that sell VOIP but aren't open. |
| 03:02.42 | WIMPy | There are lots that sell phone lines that actuially are voip and only work with their IADs, but they don't sell (it as) VOIP. |
| 03:03.17 | showme | xxiao: I don't do much international, but your best bet may be to (your in the usa, right?) get one "main" provider, then look at providers in asia and europe that you can do low volume there too, its easy to have a ton of providers all on one single asterisk box. |
| 03:03.18 | xxiao | WIMPy: ? you normally got a device(ATA) from them with one or two year contract, i want to use my own ATA instead |
| 03:03.40 | showme | if you're doing your own asterisk box, you don't need an ATA at all |
| 03:04.01 | WIMPy | xxiao: Not here (EU). |
| 03:04.33 | showme | your asterisk box talks directly to your voip provider, no hardware inbetween (that is unless you're using a voip provider that locks down like vonage). |
| 03:04.42 | xxiao | just curious, does EU's celluar contract work like US, i.e. one or two year contract? |
| 03:05.01 | xxiao | I now in Asia they don't lock your cellphone to operators/contracts at all |
| 03:05.05 | xxiao | s/now/know |
| 03:05.06 | WIMPy | You can get anything and more :-) |
| 03:05.56 | xxiao | showme: cool, that's what I'm trying to set up, want to get openwrt+asterisk on my little wireless router for family use |
| 03:06.20 | xxiao | only need buy a good ATA, best, with Wifi |
| 03:07.23 | WIMPy | You don't want to do voip over wifi, usually. |
| 03:07.57 | showme | unless you're wanting to use traditional land line type phones, you don't need any ata at all, your asterisk box talks to your (multiple if wanted) and you use sip phones in your house and they talk to your asterisk box. |
| 03:09.05 | xxiao | i have 3 analog phones now that would like to continue to use...besides SIP phones are expensive still |
| 03:09.33 | showme | oops: your asterisk box talks to your voip provider (multiple if wanted) and you use sip phones (as many as you want) in your house and they talk to your asterisk box. |
| 03:10.11 | *** join/#asterisk marlowe (~marlowe@ip68-100-147-177.dc.dc.cox.net) |
| 03:10.49 | xxiao | if i use ATA who service the ring tone? asterisk, the voip-provider, or the ATA? |
| 03:10.51 | showme | they've come way down, depending on features you need, you can actually find simple phones in the $30 range, you don't need those $400 cisco ip phones |
| 03:11.25 | xxiao | showme: let me google for a good sip phone then, but then i need run ethernet cable in the house if wifi is not optimal for echo etc |
| 03:11.32 | showme | the voip signals the asterisk box, the asterisk box signals the ata, and the ata generates dial tone, ring voltage, etc. |
| 03:12.32 | showme | another option (not cheap, but not really expensive either), is panasonic makes a multi-line multi-handset sip cordless (dect) system. |
| 03:13.49 | WIMPy | Yes, those things are an option. But I'd make sure it's CAT-iq. |
| 03:14.16 | showme | you can get the base as either just the dect box (needs power and ethernet, then dect to the phones), or you can get a base that includes one traditinal phone built in then add as many cordless handsets as you want (I think it maxes out at 8?) |
| 03:14.55 | WIMPy | I think the most bases support 6 handsets. |
| 03:16.39 | showme | your right, I just looked, it supports 6 handsets (plus the base phone if you get that model), they support adding up to 8 sip accounts to them.. |
| 03:17.17 | xxiao | it's possible i can hook ATA directly to the network(or SIP home) without installing asterisk, that should also work at the start |
| 03:17.20 | *** join/#asterisk cesar_CR (~cesar@celord.ice.co.cr) |
| 03:17.33 | xxiao | s/SIP home/SIP phone/ |
| 03:17.41 | WIMPy | sure |
| 03:18.05 | xxiao | that's how vonage/ooma etc works i think |
| 03:18.26 | showme | that the ONLY way vonage works, they don't allow direct sip access |
| 03:18.46 | *** join/#asterisk cesar_CR (~cesar@celord.ice.co.cr) |
| 03:19.15 | showme | so if you wanted vonage on your asterisk box, you have to get an FXO gateway to hook up to the vonage ATA |
| 03:19.49 | *** join/#asterisk celord (~celord@celord.ice.co.cr) |
| 03:19.57 | xxiao | has a classmate producing lots of DECT using DSP-groups solution for years.... |
| 03:20.02 | xxiao | s/has/have/ |
| 03:20.08 | xxiao | can't type today |
| 03:21.25 | xxiao | showme: why? i thought vonage's ATA has ethernet and FXS, that's about it? |
| 03:21.34 | xxiao | why do I need a FXO gateway |
| 03:21.39 | WIMPy | If you want to go voip, don't use the old dect, use cat-iq. |
| 03:22.14 | showme | Yeah, but vonage views it as closed, so there is no way to directly hook an asterisk box up to vonage. |
| 03:22.38 | xxiao | showme: no wonder vonage can't fly high |
| 03:22.39 | showme | they won't give you the sip info, and the ATA is locked down to where you can't even log into it for any real purpose |
| 03:22.47 | xxiao | no vonage for me |
| 03:22.58 | coppice | The DSP-Group is a company with a weird history |
| 03:23.06 | xxiao | coppice: why? |
| 03:23.18 | showme | The panasonic phones I'm thinking of are cat-iq |
| 03:23.29 | xxiao | they're doing cortex-A8 with DECT block in it, will possibly run android as told |
| 03:23.48 | xxiao | it will be a tablet + CAT-iq i think, 7" LCD |
| 03:23.52 | xxiao | or even bigger |
| 03:24.17 | showme | panasonic kx-tgp500 (pure cordless) and kx-tgp550 (cordless with one corded phone) |
| 03:25.21 | coppice | xxiao: they sort of became audiocodes, without really becoming audiocodes, and then reappeared as DSP Group again as a separate company |
| 03:25.43 | xxiao | coppice: i feel they're doing well |
| 03:25.57 | coppice | xxiao: and much of what was the DSP Group is now called CEVA |
| 03:26.15 | coppice | CEVA does well |
| 03:26.47 | coppice | A8 + DECT sounds like an odd mix |
| 03:26.55 | WIMPy | Why? |
| 03:27.14 | coppice | a high performance core + a low performance comms channel |
| 03:27.35 | xxiao | just revisit dsp-group's site, saw "DECT Android Phone" there |
| 03:27.52 | xxiao | CEVA, not sure how it is doing though. DSP-group seems booming |
| 03:28.16 | WIMPy | With todays software I wouldn't call an A8 high performance. |
| 03:28.55 | coppice | ceva gets a cut from a large percentage of the cell phones sold |
| 03:29.24 | xxiao | coppice: what does that mean? |
| 03:29.38 | coppice | ceva is the arm of DSP cores |
| 03:30.42 | coppice | for some reason "there's an arm in every cellphone" gets a lot more publicity than "there's a ceva in a high percentage of cellphones" |
| 03:31.19 | xxiao | you mean ceva owns some DSP IP in cellphone, or inside ARM chip? |
| 03:31.35 | xxiao | say, the NEON, or Mali(which is not really a DSP per se) |
| 03:32.43 | coppice | ceva licences DSP cores in teh same way ARM licences general purpose CPU cores, and they are very popular |
| 03:32.59 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 03:34.05 | xxiao | i c, though i don't what exactly ceva owns inside arm(or anything), only know TI DSP, freescale DSP, and ARM's Neon DSP |
| 03:34.32 | coppice | ceva owns nothing inside arm. |
| 03:35.21 | xxiao | then where |
| 03:36.23 | xxiao | on the A8+DECT, the A8 is also acting as a tablet to some extent, so people can touch the screen to do more things |
| 03:36.25 | coppice | I just said. they licence DSP cores. If you see a chip that mentions it has an oak or teak core, it uses a core licenced from ceva. most cellular chips either have a TI core or a ceva one |
| 03:36.39 | xxiao | i c |
| 03:36.52 | xxiao | good to know something new |
| 03:42.29 | stram | hi, i installed asterisk 1.8.4.2 from source on a linux machine, it seems to be segfaulting out of the box: http://pastebin.com/xKmu46BM |
| 03:42.42 | stram | any tips? i can't seem to get friggin GDB installed to take a look at the core dump |
| 03:45.06 | coppice | xxiao: some of those DSP Group chips have a TeakLite core in them |
| 03:49.13 | coppice | xxiao: that Cortex + DECT shipset is really Cortex + 802.11 + DECT. That makes a lot more sense |
| 04:04.01 | xxiao | coppice: right |
| 04:04.21 | xxiao | hope they can have that A8 in production soon |
| 04:20.16 | *** join/#asterisk showme (~jerry@24.94.180.175) |
| 04:25.33 | *** join/#asterisk ltd (~z@zx.io) |
| 04:37.42 | WIMPy | stram: Without gdb you kann only search bu hand. Disable autoload in modules.conf and load the modules one by one until you find the one that crashes. |
| 04:38.03 | WIMPy | You can use a shell script however. |
| 04:39.38 | coppice | xxiao: there is an amazing flood of A8 and A9 chips appearing from people you've never heard of |
| 04:45.56 | *** join/#asterisk celord (~celord@celord.ice.co.cr) |
| 04:57.04 | WiretapSeven | stram, make sure you're only loading ONE voicemail module |
| 04:57.40 | xxiao | coppice: for A8 yes, for A9 not so |
| 04:58.07 | xxiao | so far only tegra2 is selling in volume |
| 04:59.35 | xxiao | i have a pandaboard and is less interested in it as you can not get the chips,anyway, all devices should have a light-arsterisk built-in |
| 05:17.49 | *** join/#asterisk showme (~showme@24.94.180.175) |
| 05:26.44 | coppice | xxiao: there must be at least 10 people shipping dual A9s now |
| 05:27.30 | coppice | xxiao: what do you mean when you say you can't get the chips in the pandaboard? |
| 05:28.52 | stram | WiretapSeven: thisi s completely out of the box, but ok i'll try that |
| 05:29.18 | stram | wait, WiretapSeven, it crashes even when i rm rf'd /etc/asterisk |
| 05:29.35 | WiretapSeven | stram, ensure your RAM is good, and ensure you compiled fro the right architecture |
| 05:30.57 | stram | WiretapSeven: its a xen domU, think that's causing issues? |
| 05:31.53 | WiretapSeven | virtualisation and asterisk don't get along so well, very dependant on the type of virtualisation though |
| 05:48.13 | *** join/#asterisk fauxalliance (~fauxallia@142.163.158.222) |
| 05:49.29 | xxiao | coppice: well many claim they have A9 but only Tegra is selling them in volume. omap4 is still quite hard to get unless you are a key customer to TI |
| 05:49.59 | xxiao | AMlogic is the one I know is shipping in volume, single core A9 though |
| 05:50.21 | xxiao | hope i missed some major A9 players, I'd happy to find that out |
| 05:56.16 | *** part/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 06:00.24 | coppice | xxiao: the snapdragon, samsung's own dual A9, TI and others ship in volume. I don't know any big sellers with the Tegra 2. Its in things like the LG dual core phone, but that seems to be failing |
| 06:00.48 | coppice | TI are on backlog for the dual A9s ships, because they *are* selling |
| 06:02.04 | coppice | Freescale seem to be selling quite a few A9s into Chinese tablet makers, and they have single, dual and quad core A9s available |
| 06:02.51 | WIMPy | o.O Quad A9, available? |
| 06:04.18 | coppice | I somehow doubt the quad one will be a big seller, but someone told me he was actually playing with them |
| 06:05.15 | WIMPy | Would be nice for eco servers. |
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| 09:37.33 | *** join/#asterisk PoWeRKiLL (~powerkill@109.65.28.127) |
| 09:37.37 | PoWeRKiLL | hello |
| 09:38.13 | PoWeRKiLL | someone know what version of asterisk with wich version of spandsp is the more reliable for fax receiving ? |
| 09:39.40 | irroot | dont use 1.4 its not easy to get set up |
| 09:39.58 | irroot | 1.6+ and spandsp 0.0.6 |
| 09:40.13 | irroot | use 1.8 if you looking at T.38 gateway |
| 09:40.58 | PoWeRKiLL | T.38 gateway mean asterisk receive the fax from sip to tiff ? |
| 09:42.04 | irroot | nope to/from a T.38 endpoint to a TDM endpoing |
| 09:42.07 | irroot | point |
| 09:42.51 | PoWeRKiLL | so with 1.6 and spandsp 0.0.6 asterisk can act as a T38 endpoint |
| 09:42.53 | PoWeRKiLL | ? |
| 09:44.14 | irroot | yes endpoint only |
| 09:44.42 | irroot | the gateway allows 2 different endpoints to send/recieve [T.30/T.38] |
| 09:45.52 | PoWeRKiLL | I'm using 1.6.2.18 and spandsp-0.0.6pre18 and I have problem with receiving fax more than 1 pages |
| 09:46.13 | coppice | what happens with more than one page? |
| 09:46.39 | PoWeRKiLL | fax are getting cut |
| 09:47.07 | coppice | cut in what way? betwen pages? in the middle of a page? |
| 09:47.09 | irroot | make sure timing is right faxes require timing to be critical |
| 09:53.55 | *** join/#asterisk wonderworld (~ww@port-92-201-31-72.dynamic.qsc.de) |
| 09:58.49 | PoWeRKiLL | timing is right |
| 09:59.00 | PoWeRKiLL | i have a digium card inside |
| 09:59.18 | *** join/#asterisk vfabi (~fabi@host-static-188-237-243-70.moldtelecom.md) |
| 09:59.22 | coppice | what makes you think the timing is right? |
| 10:00.00 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 10:00.07 | PoWeRKiLL | i use the conference and other stuff |
| 10:00.11 | irroot | timing set in asterisk.conf and you are using res_timing_dahdi.so ?? just having a card does not make timing right |
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| 10:05.45 | PoWeRKiLL | Is there a utility to run to test timing ? |
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| 10:21.27 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 10:21.32 | [sr] | howdy friends |
| 10:22.19 | [sr] | asterisk CDR doesn't have the ability of showing me the info: time between rings and the answer? and also if a call was not answered? |
| 10:27.15 | PoWeRKiLL | coppice I still get WARNING[19623]: app_fax.c:173 span_message: WARNING T.30 Page did not end cleanly |
| 10:27.43 | PoWeRKiLL | sr yes it's the diff between duration and billsec |
| 10:28.02 | PoWeRKiLL | for answered and not answered you have the column disposition |
| 10:50.20 | coppice | PoWeRKiLL: what kind of digium card are you using? if its an E1 or T1 what is your clock source configuration? |
| 10:57.12 | *** join/#asterisk Denial (~Denial@drgi.co.uk) |
| 11:07.07 | [sr] | PoWeRKiLL: hum, ok this is not really asterisk related but freepbx |
| 11:07.10 | PoWeRKiLL | coppice it's E1 card but I don't have any pri on it |
| 11:07.15 | [sr] | my call logs doesn't show me nothing :( |
| 11:07.32 | PoWeRKiLL | sr I'm not using freepbx |
| 11:07.55 | [sr] | PoWeRKiLL: what do you use to have that information? |
| 11:08.18 | PoWeRKiLL | wich one WARNING[19623]: app_fax.c:173 span_message: WARNING T.30 Page did not end cleanly ??? |
| 11:09.20 | irroot | where is the fax endpoint on DAHDI ?? or sip or the like |
| 11:10.15 | PoWeRKiLL | the scenario is like this TELES GW -> Opensips -> Asterisk 1.6 Fax Endpoint (with Dahdi and digium hardware on it) |
| 11:13.19 | irroot | you using T.30 on a IP link this is bound to be a problem |
| 11:13.30 | irroot | T30 SHOULD be used on TDM |
| 11:13.42 | irroot | T.38 on NON TDM |
| 11:14.16 | irroot | even if the timing starts out fine UDP / IP / .... does not facilitate it stays that way |
| 11:14.50 | PoWeRKiLL | so what is the solution ? |
| 11:20.44 | irroot | use T.38 |
| 11:21.11 | *** join/#asterisk davlefou (~david@41.227.51.176) |
| 11:21.19 | irroot | a ATA on a local ethernet with A/U LAW can be reliable to a point |
| 11:21.36 | irroot | read the spandsp home page re faxing over IP |
| 11:31.49 | irroot | http://www.soft-switch.org/spandsp_faq/ar01s03.html |
| 11:31.59 | irroot | Can I FAX reliably through a VoIP channel? |
| 11:33.24 | irroot | http://www.soft-switch.org/foip.html |
| 11:33.32 | irroot | Faxing over IP networks |
| 11:42.11 | *** join/#asterisk hetii (~hetii@87.99.51.172) |
| 11:42.15 | hetii | Hello :) |
| 11:43.51 | hetii | I have access to sip provider that support t38 protocol to sending fax. I use asterisk 1.8 + freepbx. What i need else to be able to send/recive faxes ? |
| 11:44.23 | [sr] | PoWeRKiLL: i see, i have some problems with cdr with unexisting files |
| 11:45.17 | hetii | what i can image is t38modem --> hylafax --> iaxmodem --> asterisk but i`m not sure if i need all of that stuff ? Is there some other more easy solution ? |
| 11:45.24 | irroot | hetti if it stays all sip a good ATA that does T.38 hooked up to asterisk |
| 11:46.04 | irroot | hylafax <-> t38modem <-> gnugk <-> ooh323 <-> asterisk <-> SIP |
| 11:46.11 | irroot | is the setup i use |
| 11:46.40 | irroot | then i also use T.38 GW to replace the last sip with T38<->T.30<->DAHDI |
| 11:47.05 | hetii | ok thats a bit twisted :) |
| 11:49.44 | irroot | it works well i promise :P |
| 11:50.15 | hetii | hmm but why you use gnugk and ooh323 when hylafax can comunicate over iaxmodem with * i thought that t38modem is only required on this case by sip provider that support t38? |
| 11:53.19 | coppice | sadly, if you need to use H.323 and T.38 for FAXing, we haven't provided great options :-) |
| 11:54.05 | coppice | but if you only use T.38, T.38modem + hylafax is the best current option |
| 11:54.13 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 11:54.16 | hetii | on my case H.323 is not required |
| 11:54.54 | coppice | then why does you chain look so weird? what's wrong with just asterisk? |
| 11:56.35 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc7-pool12-2-0-cust219.15-1.cable.virginmedia.com) |
| 11:58.11 | hetii | i don`t know how its look now on * 1.8, on the past when i want have fax sending/reciving solution i needed hylafax +iaxmodem+* and some pstn line But know when i know that my sip provider support t.38 i want to use it. So the question is what i need to run it. As i see on freepbx there are option T38 Pass-Through so something else are required as well |
| 11:59.40 | coppice | install spandsp. install asterisk. set the config files. FAX away |
| 12:01.44 | PoWeRKiLL | asterisk 1.6 with spandsp can act as a T38 endpoint or not ? |
| 12:02.08 | PoWeRKiLL | coppice can you help to check why I got this error WARNING[19623]: app_fax.c:173 span_message: WARNING T.30 Page did not end cleanly ??? |
| 12:02.12 | hetii | I have isdn card on my * box but i don`t plan to use it to sending fax over it |
| 12:02.15 | coppice | yes it can. only gateway is missing from 1.6 |
| 12:05.21 | hetii | but spandsp is only a librarie, as i suppose i need also something that will handle faxes like hylafax or whatever else that will allow send faxes to mail etc... |
| 12:07.04 | coppice | spandsp does everything hylafax does. you just need fax to mail and mail to fax scripts. various versions of which are around. keep a low profile doing email to fax or fax to email, or you may get lawyer's letters from J2 |
| 12:08.05 | hetii | J2 ? |
| 12:08.21 | PoWeRKiLL | hetii J2 = eFax |
| 12:08.38 | PoWeRKiLL | coppice how can I debug the page truncate problem ? |
| 12:12.33 | hetii | ok i need to read more about spandsp but all off that is strange a bit, for example i cannot understand why i should letters form J2 when i use spandsp to send few fax :/ |
| 12:12.37 | hetii | or recive ;/ |
| 12:14.12 | coppice | if you only send a few there won't be enough in it for them to bother. if you send or receive a lot you may get letters demanding patent royalties, or that you get out the business |
| 12:27.51 | *** join/#asterisk davlefou (~david@41.225.195.104) |
| 12:29.16 | *** join/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 12:33.52 | gavimobile | I get this error when trying to install asterisk configure: error: *** termcap support not found (on modern systems, this typically means the ncurses development package is missing) |
| 12:37.59 | tzafrir_laptop | gavimobile, what distro is it? |
| 12:42.20 | gavimobile | tzafrir_laptop: centos5 new install baby |
| 12:42.44 | tzafrir_laptop | yum install ncurses-devel |
| 12:43.19 | gavimobile | tzafrir_laptop: and then try to ./configure again |
| 12:46.36 | *** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 12:46.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:53.01 | *** join/#asterisk marlowe (~marlowe@ip68-100-147-177.dc.dc.cox.net) |
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| 12:59.17 | [sr] | PoWeRKiLL: problem solved, i didn't have the -addons installed after the upgrade from 16 to 18 : |
| 12:59.18 | [sr] | :) |
| 13:00.54 | khussein78 | problem with connecting asterisk to cisco 1700v router |
| 13:01.01 | khussein78 | logs is here http://pastebin.com/4UgCYAjY |
| 13:01.32 | khussein78 | how can i debug this |
| 13:04.33 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 13:14.33 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 13:19.33 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 13:19.37 | [sr] | howdy again :) |
| 13:19.58 | [sr] | what's the best option to block an extension from going to the outside? |
| 13:20.34 | [sr] | sample, if it dials 234XXXXXX it'll go to the default outgoing route and goes outside |
| 13:24.03 | *** join/#asterisk boazb (~boaz@bzq-82-80-219-90.red.bezeqint.net) |
| 13:25.31 | hetii | you can check callerid and setsome other route eg. to hangup |
| 13:39.33 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
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| 13:46.16 | [sr] | hetii: hum |
| 13:47.16 | hetii | you write your own configs or use freebpx ? |
| 13:48.53 | *** join/#asterisk anonymouz666 (~anonymouz@189.36.178.12) |
| 13:49.05 | anonymouz666 | good morning! |
| 13:49.41 | hetii | or afternoon :) |
| 13:52.38 | anonymouz666 | hetii: you are too fast :) |
| 13:54.26 | *** join/#asterisk LemensTS (~matthew@adsl-70-130-178-219.dsl.stlsmo.swbell.net) |
| 13:55.35 | LemensTS | Im using the hpec echo canceller, in /etc/dahdi/system.conf it has echocanceller=mg2,1 ..... is mg2 hpec? |
| 13:56.53 | anonymouz666 | nops |
| 13:56.56 | anonymouz666 | mg2 is mg2. |
| 13:57.31 | anonymouz666 | it's the default dahdi echocan I guess. |
| 13:57.45 | anonymouz666 | In the beggining I was using mg2 but I switched to oslec. |
| 13:57.55 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-249-008.pools.arcor-ip.net) |
| 13:57.56 | LemensTS | Is oslec better then hpec? |
| 13:57.58 | irroot | oslec rox |
| 13:58.04 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:58.07 | anonymouz666 | hello irroot |
| 13:58.16 | anonymouz666 | are you the master of t.38 gateway? :) |
| 13:58.24 | anonymouz666 | LemensTS: I don't think so. |
| 13:58.45 | irroot | lo anon master mmm not sure but i have been getting working |
| 13:58.52 | irroot | mastered me perhaps it has |
| 13:59.40 | irroot | you using it ?? |
| 13:59.49 | anonymouz666 | not yet, but I will soon |
| 14:00.11 | irroot | cool please get hold of me its timing critical as all T.30 |
| 14:00.32 | irroot | IMHO T.30 must be on TDM |
| 14:01.00 | irroot | the OOH323 updates allow it to work with t38modem / hylafax / gnugk |
| 14:01.12 | *** join/#asterisk nex_necis (~xix@host187-16-dynamic.57-82-r.retail.telecomitalia.it) |
| 14:01.12 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 14:01.53 | anonymouz666 | good to know, I have been using asterisk since 2004 never had to touch in h323 channels :-) I am lucky |
| 14:03.03 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 14:03.17 | irroot | used all 3 versions of h323 |
| 14:03.34 | irroot | the chan_h323 is worst ooh323 best FYI |
| 14:08.35 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 14:11.16 | anonymouz666 | nice to see a setup with 8 e1's in the same machine and asterisk running rock-solid without a restart. |
| 14:11.49 | tzafrir_laptop | irroot, whis is the third? |
| 14:12.06 | irroot | the old chan_oh323 |
| 14:12.21 | irroot | it was run by i belive greek company in access |
| 14:12.49 | irroot | started with asterisk 1.[01] |
| 14:13.03 | irroot | i kept it in my distro till 1.4 |
| 14:13.31 | irroot | switched to patched chan_h323 @ 1.6 to support h323plus |
| 14:13.48 | irroot | but could not figgure out why it would load unload and crash on reload |
| 14:14.11 | irroot | then used ooh323 |
| 14:15.10 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
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| 14:19.29 | *** part/#asterisk jayvee (~jayvee@azaroth.sunriseroad.net) |
| 14:20.53 | LemensTS | http://pastebin.com/20mQRHvz <---anyone have an idea on this hpec error digium support is not open today |
| 14:24.26 | PoWeRKiLL | how to go from t30 to t38 ? |
| 14:25.31 | irroot | powerkill many ATA's do T.38 and your ITSP needs to support it too |
| 14:26.37 | irroot | lemensts you change network card ?? i believe the licences are bound to the hardware |
| 14:29.54 | *** join/#asterisk vfabi (~fabi@host-static-188-237-243-55.moldtelecom.md) |
| 14:38.19 | LemensTS | irroot: nope haven't changed it |
| 14:38.51 | irroot | just checking when you signed up you needed the ethernet detials right ?? |
| 14:39.06 | *** join/#asterisk i_eat_children (~chatzilla@pool-74-110-3-57.bflony.east.verizon.net) |
| 14:39.27 | i_eat_children | good morning |
| 14:40.27 | i_eat_children | im getting this error in console, i thought it was bug in 1.6.1 but it seems to be even more severe (causing higher CPU usage) in 1.6.2 |
| 14:40.28 | i_eat_children | http://pastebin.com/vxGM2n5j |
| 14:42.03 | *** join/#asterisk davlefou (~david@41.225.127.125) |
| 14:42.26 | LemensTS | irroot: i do not remmeber it had been a long time. i will try to reregister maybe that wil fix it |
| 14:44.28 | *** join/#asterisk t_dot_zilla (~chatzilla@pool-74-110-3-57.bflony.east.verizon.net) |
| 14:44.34 | t_dot_zilla | im getting this error in console, i thought it was bug in 1.6.1 but it seems to be even more severe (causing higher CPU usage) in 1.6.2 |
| 14:44.38 | t_dot_zilla | http://pastebin.com/vxGM2n5j |
| 14:45.51 | *** join/#asterisk jkroon (~jkroon@dsl-241-240-125.telkomadsl.co.za) |
| 14:48.46 | LemensTS | echocancel=1024 will do more than 256 right, it just uses more cpu? |
| 14:50.26 | t_dot_zilla | i dont know... i dont think anyone is here... |
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| 15:25.38 | t_dot_zilla | http://nerdvittles.com/?p=743#comments |
| 15:26.00 | t_dot_zilla | or rather http://nerdvittles.com/?p=743 |
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| 15:51.45 | xxiao | gosh, i had a sip2sip account that worked out of the box, but ekiga.net account never worked |
| 15:51.57 | nex_necis | hi, i need to add /quit |
| 15:52.13 | xxiao | try to make calls between these two accounts, guess i should use two sip2sip instead |
| 15:54.54 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:01.16 | xxiao | two calls between sip2sip worked well, now replacing sip2sip with my own asterisk setup |
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| 16:53.37 | *** join/#asterisk admin0 (~admin0@cm16.theta48.maxonline.com.sg) |
| 16:53.55 | admin0 | hi .. does asterisk send all AAA records to radius or just auth only ? |
| 17:14.24 | *** join/#asterisk fauxalliance (~fauxallia@142.162.192.5) |
| 17:16.09 | admin0 | found my answer . thanks for listening |
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| 19:06.13 | gavimobile | remind me please where I need to put the language pack |
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| 19:43.37 | xxiao | set up arsterisknow on a local IP, set up 2 extensions, but softphone can not register with it |
| 19:43.56 | xxiao | softpohne registered with public sip proxy just fine though |
| 19:45.09 | xxiao | is there a way for me to turn on some debug on asterisk and see what's wrong? |
| 19:46.27 | xxiao | softphone reports "no response timeout" |
| 19:49.31 | seraphie | xxiao: on the CLI "sip set debug on" may help |
| 19:56.43 | xxiao | can u please define "CLI" in this case, a command line at where? |
| 19:56.52 | xxiao | do I need restart asterisk? |
| 19:57.03 | xxiao | new to asterisk, started yesterday |
| 19:57.31 | seraphie | go to commandline and type "asterisk -r" |
| 19:57.40 | seraphie | this will connect you to asterisk without restarting asterisk |
| 19:57.45 | seraphie | and give you an asterisk command prompt |
| 19:58.02 | seraphie | this is called the CLI "commandline interface" |
| 19:58.36 | carrar | try #asterisknow |
| 19:58.39 | xxiao | seraphie: thanks, just did it |
| 20:05.07 | xxiao | same procedure tried on trixbox and it worked, strange |
| 20:08.56 | gavimobile | does asterisk need something special to receive faxes to a regular incoming trunk |
| 20:10.22 | xxiao | alright, did 'service asterisk restart" and it now works with asterisknow |
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| 21:23.21 | gavimobile | am I running asterisk as root? http://pastebin.com/sBH19Eh7 |
| 21:24.07 | carrar | doesn't appear so |
| 21:24.23 | carrar | Running as user 'asterisk' |
| 21:30.59 | gavimobile | carrar: ok, thanks |
| 21:43.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 21:46.01 | xxiao | under asterisk, i need use extension@my-IP to login then make calls, under freeswitch, i need |
| 21:46.25 | xxiao | my-identity@my-ip to login, then use my-extension@my-ip to make calls, |
| 21:47.36 | xxiao | looks like freeswtich allows one identity to have multiple extensions, i only saw "extension" under asterisk |
| 21:48.03 | xxiao | does asterisk carry the same concept as "identity" or "username", e.g. username@sip2sip.info |
| 21:48.35 | xxiao | i thought they're identical, i.e. username _is_ extension, just the latter is digits and the former is easier to remember? |
| 21:51.36 | WIMPy | Extensions can lead to peers. Peers can have user names. |
| 21:53.04 | leifmadsen | extensions are not associated with a particular endpoint -- an extension is simply something that gets matched in the dialplan |
| 21:55.00 | *** join/#asterisk Korolev (~HellsBell@c-98-211-170-229.hsd1.fl.comcast.net) |
| 22:13.18 | xxiao | weekend project of asterisk/freeswitch is done, both are great stuff, look forward to dive in |
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