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| 00:04.07 | *** part/#asterisk ROARJ (n=joseph_2@201.216.146.69) |
| 00:05.14 | CanWood | outtolunc - deeperror, thank you both. I'm off to play |
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| 00:23.48 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 00:24.14 | C4away | anyone know if the bonk-bonk and ba-dunk join/leave sounds in meetme are built-in or if they can be overridden? |
| 00:25.35 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 00:26.34 | Strom_C | C4away: they're just wav files |
| 00:26.55 | C4away | I couldn't find them in the /var/lib/asterisk/sounds directory ... but i don't know what they would be called |
| 00:27.03 | C4away | they don't start with "conf-" as far as I can tell |
| 00:29.04 | mookid | What is meetme? :) |
| 00:29.14 | C4away | app_meetme.so |
| 00:29.30 | C4away | a conference bridge app |
| 00:31.42 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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| 00:38.07 | *** part/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 00:41.16 | outtolunc | fyi: the enter/leave audio is actually in enter.h and leave.h (in the apps dir) |
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| 01:17.24 | ManxPower | the sound files are listed, along with their text in, oddly enough, /path/to/src/asterisk somewhere |
| 01:18.54 | Corydon76-dig | ManxPower: not anymore |
| 01:19.15 | Corydon76-dig | ManxPower: the text file is included in the sound file distribution, though |
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| 01:24.13 | C4away | ack |
| 01:24.18 | C4away | that is what I was afraid of |
| 01:25.52 | C4away | so, there is no simple way of replacing the sound without formatting the audio data as a byte array? |
| 01:26.37 | C4away | and then rebuilding, at a minimum, enter.h and leave.h? |
| 01:27.35 | C4away | wait, I'm only finding those files in the source, they are integrated into app_meetme then? |
| 01:28.03 | *** join/#asterisk Gwayne (n=Gwayne@bb116-14-75-1.singnet.com.sg) |
| 01:29.55 | C4away | that is way too much work for a simple prank |
| 01:30.48 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:30.57 | TJNII | Oh I disagree |
| 01:31.14 | TJNII | because, while it is a lot of work, how long will it take them to find and fix it? |
| 01:31.26 | C4away | hahah |
| 01:31.58 | C4away | I thought it would be funny to swap the bonk-bonk with the doink-doink from Law & Order |
| 01:32.21 | TJNII | Especially if you _really_ mess with them by changing it everywhere on the drive, and put in a dns rule so when they try and re-download the source they download it from you. |
| 01:32.32 | C4away | hahaha |
| 01:32.55 | C4away | ok, hmm, so raw != pcm wav |
| 01:33.05 | C4away | I'll have to re-transcode the file to raw then |
| 01:35.32 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 01:39.33 | C4away | damn, too many options, do I want pcm or a/ulaw, mono/stereo, little or big endian, singned or unsigned... not so raw afterall then huh |
| 01:40.06 | Strom_C | alaw and ulaw are both pcm |
| 01:40.25 | C4away | sound forge gives me all these options when saving as .RAW |
| 01:40.39 | C4away | I'm going with 16-bit mono signed little endian |
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| 01:54.29 | C4away | anyone have any suggestions for converting a .raw audio file into a hex-formatted byte array? |
| 01:54.42 | C4away | for example: 0xba, 0xba, 0xb0, 0xa6, 0xa9, 0xb8, 0xfe, 0x46, 0x42, 0x46, |
| 01:54.53 | NovceGuru | :\ |
| 01:55.29 | C4away | I'm thinking perl or python |
| 01:58.20 | Strom_C | you're probably on the right track with that |
| 01:58.21 | Strom_C | gogogo |
| 01:58.29 | C4away | lol |
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| 01:58.50 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:01.56 | C4away | bin2h |
| 02:02.07 | C4away | much simpler |
| 02:09.06 | *** join/#asterisk rsc-232 (n=mrdigita@216-164-0-122.c3-0.drf-ubr2.atw-drf.pa.cable.rcn.com) |
| 02:09.09 | rsc-232 | hey al |
| 02:09.10 | rsc-232 | all |
| 02:09.41 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 02:12.40 | C4away | it works! |
| 02:12.41 | C4away | hahaha |
| 02:39.39 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 02:39.39 | *** mode/#asterisk [+o denon] by ChanServ |
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| 03:20.31 | *** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167059025.pppoe-dynamic.nb.aliant.net) |
| 03:20.40 | C4away | who runs the voip-info.org site? |
| 03:21.11 | Strom_C | C4away: why? |
| 03:21.43 | file | Strom_C: alexhopper is a fan of your payphone listing |
| 03:21.50 | Strom_C | oh? |
| 03:21.54 | alexhopper | Ohhh yeah! |
| 03:22.03 | alexhopper | I love to prank call the Belagio :p |
| 03:22.12 | Strom_C | oh god |
| 03:22.14 | Strom_C | don't be a fucktard |
| 03:22.17 | C4away | in the past 20 minutes they updated the css for the site and now the left bar is in the middle of the screen |
| 03:22.22 | alexhopper | lol! |
| 03:22.31 | alexhopper | I haven't done it in a long time, actually |
| 03:22.44 | Strom_C | yeah, blah blah blah. |
| 03:22.45 | C4away | I used to get drunk and call 1-800-call-att |
| 03:22.50 | alexhopper | It's neat to call around and see if anyone answers the different places though |
| 03:22.57 | C4away | see how long the operators would stay on the phone and chat |
| 03:23.07 | alexhopper | lol, C4away |
| 03:23.22 | C4away | had one old guy who was once in the navy talk about the deck cannons for 45 minutes |
| 03:23.46 | alexhopper | Is that one of the call centers where they can't terminate the calls? There's some like that in Canada, they will not hang up... |
| 03:24.09 | C4away | see, the thing is, if you try to have phone sex with them they hang up quick, but if you just chat about everyday bullshit they don't get upset |
| 03:24.18 | C4away | learned that one the hard way |
| 03:24.20 | C4away | j/k |
| 03:24.32 | C4away | no, they would hang up on us all the time |
| 03:24.43 | alexhopper | haha - nice. |
| 03:24.49 | C4away | you'd have to call about 10 times to get someone who was willing to talk for more than about 5 minutes |
| 03:25.23 | C4away | anyway, the voip-info.org website is pretty much unusable now |
| 03:25.38 | Strom_C | it's been pretty much unusable content-wise for years |
| 03:25.45 | C4away | heh, well, kinda |
| 03:25.48 | Strom_C | the layout is just shifting to match the content |
| 03:25.53 | C4away | hahaha |
| 03:27.32 | *** join/#asterisk rabelais (n=blank@unaffiliated/rabelais) |
| 03:28.00 | C4away | hmm, trying to find an example of setting a non-default MOH class from the dialplan |
| 03:28.58 | C4away | for an entire context |
| 03:30.26 | *** join/#asterisk PepOSX (n=angeldav@190.72.151.25) |
| 03:31.03 | C4away | ah SetMusicOnHold(buttrock) |
| 03:38.08 | jaytee | SetMusicOnHold(rickrolled) |
| 03:38.17 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 03:38.37 | C4away | haha |
| 03:39.17 | C4away | for some reason it is not finding the new class I set up =( |
| 03:39.34 | C4away | reloaded, even restarted, asterisk |
| 03:39.58 | C4away | [Jul 8 21:36:30] WARNING[5886]: res_musiconhold.c:660 get_mohbyname: Music on Hold class 'cfbi' not found |
| 03:42.03 | [TK]D-Fender | C4away: pastebin is your friend. |
| 03:42.40 | C4away | for one line? |
| 03:42.43 | C4away | that's excessive |
| 03:43.04 | [TK]D-Fender | C4away: ....it isn't 1 line, It'd better be at least 3... |
| 03:43.40 | C4away | for a single-line error? |
| 03:43.49 | C4away | should I repeat it three times in the pastbin post? |
| 03:44.33 | C4away | oooh you mean my musiconhold.conf ? |
| 03:44.41 | C4away | lol, sorry I was quite confused for a second |
| 03:45.00 | C4away | I thought you were giving me hell for pasting the error |
| 03:46.50 | C4away | http://pastebin.ca/1066252 |
| 03:47.02 | C4away | that's is everything in musiconhold.conf that is not preceeded by the ";" character |
| 03:47.26 | *** join/#asterisk JCJC (n=JCJC@netblock-72-25-115-165.dslextreme.com) |
| 03:47.30 | *** join/#asterisk hi365_m (n=hi365@213.151.61.31) |
| 03:47.56 | C4away | ok, now that is everything, I deleted all the default comments and examples |
| 03:48.10 | *** kick/#asterisk [alexhopper!n=file@asterisk/developer-and-muffin-lover/file] by file (file) |
| 03:48.35 | [TK]D-Fender | C4away: I'd check your permissions. |
| 03:48.59 | [TK]D-Fender | C4away: and that the files are indeed in that place. |
| 03:50.04 | C4away | would it say that same error for files missing / permissions errors? |
| 03:50.29 | C4away | in that case I would think that a [none] class with directory=/dev/null would fail over to [default] if that were the case |
| 03:51.02 | C4away | which is the commonly demonstrated method of implementing a [none] class |
| 03:51.14 | C4away | in any case, I am checking as we speak |
| 03:51.27 | C4away | uh, hmm |
| 03:51.49 | C4away | yes, the directory is empty and coreftp is sitting there with "permission denied" for all the transfers for the .wav files |
| 03:52.11 | C4away | shouldn't root be able to write to any directory it wants using sftp? |
| 03:52.46 | C4away | ok, hmm, chmod'd it 777 and chown'd it root and still no go |
| 03:53.01 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001225e00576.cpe.net.cable.rogers.com) |
| 03:53.07 | Hadi- | hello everyone |
| 03:53.13 | Hadi- | I have a quick question... |
| 03:53.48 | Hadi- | I have purchased the g729 codec from Digium |
| 03:54.00 | Hadi- | and I'm just wondering what file will be best for my server... |
| 03:54.18 | Hadi- | the server is a DELL 1850 2x Xenon 3.0 |
| 03:55.05 | ltd-- | What options do you have? |
| 03:55.19 | Hadi- | http://downloads.digium.com/pub/asterisk/g729/asterisk-1.2/ |
| 03:55.34 | ltd-- | are you running amd64 or i386? |
| 03:55.40 | Hadi- | i386 |
| 03:55.53 | Hadi- | its a dual xenon |
| 03:55.54 | Hadi- | intel |
| 03:56.26 | ltd-- | you probably want v32_i386 then. |
| 03:56.43 | lanning | if you are running a 64 bit kernel, with 64 bit asterisk, then 64 bit codec |
| 03:56.47 | C4away | hmm, still getting permission denied even in /usr/src/ |
| 03:56.53 | C4away | I'll just put them up on the web and wget them |
| 03:57.10 | [TK]D-Fender | Hadi-: And that Xeon. Xenon is one of the inert or "noble" gasses on the periodic table. Its far too late for that kind of of chemical humour :p |
| 03:57.10 | lanning | if you are running a 32 bit kernel, with 32 bit asterisk, then the 32 bit codec. |
| 03:57.12 | Hadi- | yes |
| 03:57.28 | Hadi- | but there are also several |
| 03:57.37 | *** join/#asterisk jeffgus (n=jeffgus@216.86.199.4) |
| 03:57.39 | [TK]D-Fender | Hadi-: What is your distro? |
| 03:57.57 | ltd-- | More importantly, what does /proc/cpuinfo say |
| 03:58.09 | Hadi- | CentOS 5 |
| 03:58.55 | lanning | probably this one: codec_g729a_v32_i686.tar.gz |
| 03:58.57 | Hadi- | http://www.pastebin.ca/1066267 |
| 03:59.34 | Hadi- | there is the /proc/cpuinfo ;) |
| 03:59.44 | ltd-- | i686 or i386 will be fine |
| 03:59.51 | ltd-- | i686 will probably get you some extra optimisations |
| 04:00.00 | *** join/#asterisk moy (n=moyhu@189.169.83.78) |
| 04:00.47 | Hadi- | I see.. |
| 04:02.17 | unpaidbill | sweet, new pocketsphinx release today! 50% smaller and up to 18% faster! |
| 04:02.25 | unpaidbill | good news! |
| 04:02.58 | bijit | hi how can i add two ips to eth0? |
| 04:03.06 | bijit | sorry |
| 04:03.26 | unpaidbill | ifconfig eth0:1 ipaddress |
| 04:03.26 | Hadi- | we were using this one before |
| 04:03.27 | Hadi- | codec_g729a_v32_nocona |
| 04:03.40 | Hadi- | that explains why the other party would never hear the caller |
| 04:03.51 | Hadi- | when the call was going through as g729 :) |
| 04:04.07 | bijit | ty unpaidbill |
| 04:06.19 | C4away | heh unpaidbill is the reason for callers not hearing me when they call |
| 04:06.39 | unpaidbill | i can be a real bitch sometimes |
| 04:06.50 | C4away | lol |
| 04:10.05 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:13.37 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 04:16.22 | C4away | hmm, I had to put the new class above [default] in musiconhold.conf and everything works |
| 04:18.07 | C4away | hmm, is there not a way to set the music on hold class for an entire context? |
| 04:18.43 | [TK]D-Fender | C4away: MoH is not by context, its per device |
| 04:18.57 | C4away | I must put exten => 1234567890,1,SetMusicOnHold(foo) before every DID in extensions.conf then |
| 04:21.26 | [TK]D-Fender | C4away: ..... |
| 04:21.41 | [TK]D-Fender | C4away: for each DEVICE. You should generally not be doing this in your DIALPLAN. |
| 04:22.04 | C4away | hmm |
| 04:22.07 | C4away | good point |
| 04:22.21 | C4away | unless I want to override one specific DID for "support" and another for "Sales" or something |
| 04:22.37 | [TK]D-Fender | C4away: Yes |
| 04:22.45 | C4away | but in this case it is for the entire contex which is, conveniently, one user in sip.conf |
| 04:24.44 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 04:24.48 | [TK]D-Fender | C4away: So set the global on the user & override just that once for that single DID./ |
| 04:25.39 | C4away | yep. seems to be working |
| 04:25.40 | C4away | thanks |
| 04:25.48 | C4away | sometimes a sanity check is needed |
| 04:26.07 | C4away | I didn't think that I needed to add SetMusicOnHold for every extension in extensions.conf |
| 04:27.55 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
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| 04:39.05 | nr4q | ~centos52bug |
| 04:39.08 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 04:51.16 | *** join/#asterisk adorah (n=Michael@87.69.57.141.cable.012.net.il) |
| 04:57.14 | *** join/#asterisk MCooper (n=kmoore@64.110.169.173) |
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| 05:11.29 | MCooper | Ok... I am beating my head against a wall, and its hurting.. maybe someone here can help.... |
| 05:13.05 | MCooper | I have a digium TE410P board, euroisdn trunk between it and a Cisco CCME router... The issue is, I am getting a link light with alarm on the Cisco, and a flashing red light on the digium board... Any Ideas.. I will send pizza via paypal..... |
| 05:13.36 | MCooper | running zttools shows span one RED... |
| 05:13.51 | MCooper | (I believe it is something stupid and simple...) |
| 05:13.59 | [hC] | i can probably help |
| 05:14.14 | MCooper | [hC], Thanks... |
| 05:14.24 | [hC] | are you using a t1 crossover cable between the two devices? (i presume they are both acting in CPE mode?) |
| 05:14.38 | MCooper | Yes.. there is a crossover... |
| 05:14.48 | [hC] | so you basically arent getting link |
| 05:14.56 | MCooper | Yes... correct... |
| 05:15.09 | [hC] | i would check to make sure both devices are operating in CPE mode (if you are intending to do that, of course) then check your cable |
| 05:15.09 | MCooper | I am getting link and error on the Cisco... |
| 05:15.18 | [hC] | I'm not extremely familiar with the cisco |
| 05:15.45 | MCooper | Ok... I will check that out. |
| 05:15.52 | [hC] | i would start with the simplest of things |
| 05:15.59 | [hC] | check the cable, make sure you are using the correct pinout for example |
| 05:16.01 | MCooper | The CIsco I will verify when that person comes in... |
| 05:16.24 | [hC] | make sure that your setup is supposed to be CPE/CPE not CPE/NET (where one side acts as a network and one acts as a CPE) |
| 05:16.34 | [hC] | if they are both supposed to be CPE, the switchtype shouldnt matter for link |
| 05:16.53 | [hC] | as long as you are in pri_cpe mode on the asterisk side, it will negotiate as CPE |
| 05:17.05 | [hC] | a RED link from the digium card indicates total link loss |
| 05:18.14 | [hC] | im not positive if you should get a link without the zaptel modules loaded... im tempted to say no, and that your zaptel cnfig needs to be right, your module needs to be loaded and ztcfg needs to have been run |
| 05:18.22 | MCooper | OK... I did a cable check.. Good call... |
| 05:18.43 | MCooper | I was able to figure out that one side was bad.. got a good cable there now. |
| 05:19.46 | [hC] | oh good! |
| 05:19.49 | [hC] | do you have link now? |
| 05:20.49 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 05:22.32 | MCooper | I am working on it.. right now... |
| 05:22.40 | [hC] | Cool, good luck! |
| 05:25.54 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 05:25.56 | MCooper | [hC], Thank you.. where do I send the pizza.... |
| 05:26.01 | MCooper | I have green lights.... :) |
| 05:26.38 | [hC] | Congrats! well, i just ate :P |
| 05:26.57 | MCooper | I owe you one.. but thanks... I was getting a flat forehead... :) |
| 05:27.42 | [hC] | no worries man, another time! Ive had my fair share of help in here, pay it forward |
| 05:28.30 | MCooper | You bet... thanks... (Doing the dance... living the dream..) |
| 05:28.36 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 05:32.04 | *** join/#asterisk SanityIO_ (n=SanityIO@77.242.106.77) |
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| 05:51.58 | bkruse | MCooper: I am, however, hungry :) |
| 05:52.34 | MCooper | bkruse, Ok... |
| 05:52.44 | MCooper | I just noticed that you were hungry... |
| 05:52.48 | bkruse | :P |
| 05:55.18 | bkruse | wants a street bike |
| 05:55.26 | bkruse | I think I am going to buy one next week... |
| 05:56.11 | MCooper | Now I have to figure out why I cannot get the ISND layer 2 to link with the Cisco.... |
| 05:56.35 | MCooper | We have layer 1 link... but I am still not seeing connection to the ISDN line on the Cisco |
| 05:59.26 | MCooper | I have the Cisco side flapping... going from awaiting-establishment to TEI-Assigned... |
| 05:59.32 | MCooper | any ideas guys? |
| 05:59.44 | MCooper | next free pizza.... |
| 06:00.13 | *** join/#asterisk dasuberdavid (n=443ed318@67.159.55.26) |
| 06:01.26 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
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| 06:10.21 | *** part/#asterisk Gwayne (n=Gwayne@bb116-14-75-1.singnet.com.sg) |
| 06:11.45 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 06:14.40 | zeeesh | normal calling card scenario when user make call to some destination and then he made another call he don't disconnect access number call he just presses # for dialing another number .. how to possible with asterisk .. which asterisk feature can help for this purpose? |
| 06:14.52 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 06:15.34 | MCooper | Anyone interested in helping work out a Cisco issue? |
| 06:18.38 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:25.17 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
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| 07:00.17 | MCooper | Does anyone have an idea why the TE410P would not talk to a Cisco E1 card? |
| 07:03.31 | *** part/#asterisk redax (i=redax@82.141.129.7) |
| 07:03.35 | chwijaya | Hi, i'm Henry and I need help on asterisk codec. Does asterisk support codec g.723.1? I made a .wav file and compressed it using lbccodec into a .g723 file, but when I put it in asterisk sound directory, I can't play it through a softphone. Any idea? thanks. |
| 07:05.43 | chwijaya | Help me please, PM me.. thanks. |
| 07:19.55 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 07:24.27 | *** join/#asterisk bipser (n=bipser@u34-10.dsl.vianetworks.de) |
| 07:24.39 | bipser | hi everybody |
| 07:25.35 | unpaidbill | hi |
| 07:26.55 | MCooper | Interesting... Anyone get a TE410P working with a Cisco 3745 E1 card? |
| 07:27.32 | bipser | has anyone discovered quality problems when playing voiceprompts in the vm system since hardy heron? |
| 07:27.46 | unpaidbill | i not so recently had a TE110P card working with a cisco T1 card, not sure of the model |
| 07:28.16 | MCooper | Were there any secrets or gotchas? |
| 07:28.24 | unpaidbill | not that i remember |
| 07:28.37 | unpaidbill | i set the signaling on the cisco and the zaptel.conf |
| 07:28.49 | unpaidbill | i dont remember much more unfortuantely |
| 07:30.39 | MCooper | what did you set them to - do you remember.. Right now I am getting layer 1 link, but no sync at layer two.. according to the cisco monitor |
| 07:31.32 | unpaidbill | i think i used em wink for signaling, i didnt set it up as pri |
| 07:32.10 | unpaidbill | but it turned out to be sucky so i just took the router out of the mix and went straight from the TSU into my TE110P |
| 07:33.28 | bipser | what could be the solution for it, if my voicemail-system doesn't play the promps right. the quality is like you talk over 2 cans connected through a long line... |
| 07:34.30 | unpaidbill | are you using a crappy quality codec? |
| 07:35.08 | unpaidbill | use alaw or ulaw and see if it's better |
| 07:35.44 | unpaidbill | or give more information on your set up |
| 07:35.59 | unpaidbill | and it wouldnt hurt to post your sip.conf/extensions.conf/any other relevant conf on pastebin |
| 07:36.50 | bipser | well, in sip.conf i only allowed alaw and ulaw |
| 07:37.15 | unpaidbill | and posting sip debug logs wouldnt hurt either |
| 07:40.50 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
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| 08:44.40 | tompaw | hi |
| 08:45.00 | tompaw | how do I accept ALL incoming SIP connections and bound them to one dialplan? |
| 08:45.23 | Falle | I have a problem with a Grandstream 2020 that i need some help with. It registers fine and recives MWI but outgoing calls get a 403. If I register another kind of phone to the same account it works great. Any ideas on what i could be doing wrong? |
| 08:45.25 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 08:45.32 | tompaw | right now, at my client I receive Failed to authenticate on INVITE to '"asterisk" <sip:Unknown@91.121.75.74> |
| 08:51.14 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 08:51.25 | tompaw | also, right now my server replies with: SIP/2.0 407 Proxy Authentication Required |
| 08:56.38 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:56.45 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:57.17 | *** part/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com) |
| 08:58.45 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:03.41 | MCooper | anyone have an E1 setup from a Cisco Router to Asterisk with a TE420P card? |
| 09:04.13 | MCooper | I am having issues getting the card to sync with the Cisco... |
| 09:05.09 | pputman | MCooper, Can you paste your /etc/zaptel.conf and /etc/asterisk/zapata.conf please? |
| 09:05.30 | pputman | pastebin, rather, sorry |
| 09:05.31 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:06.24 | MCooper | I can... I have not used pastebin... |
| 09:06.36 | pputman | ~pastebin |
| 09:06.36 | jbot | [~pastebin] 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 |
| 09:14.10 | pputman | http://arstechnica.com/reviews/os/open-moko-software.ars sweet. |
| 09:14.24 | MCooper | http://pastebin.com/m2142591d |
| 09:15.44 | MCooper | http://pastebin.com/m1570a78c |
| 09:15.58 | MCooper | pputman, There you go... both the zaptel and zapata conf files... |
| 09:16.32 | MCooper | The issue that I am having - the layer2 portion of the ISDN Link is not coming up.. (Cisco calls it layer 2) |
| 09:17.10 | pputman | which span are you having problems with? |
| 09:17.18 | *** join/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg) |
| 09:17.22 | anebi | hi |
| 09:17.33 | MCooper | Span 1 right now... |
| 09:17.44 | anebi | we have installed asterisk on centos 5 and i want to ask |
| 09:18.00 | anebi | is asterisk user need to have bash login permissions? |
| 09:18.11 | anebi | or i can set it /sbin/nologin |
| 09:18.47 | pputman | MCooper, Is the cisco side set to cpe or net? and to receive or provide timing? |
| 09:18.57 | MCooper | Net |
| 09:19.45 | pputman | MCooper, well that's one problem then, you have both sides set to pri_net. You need one side set to cpe, the other side to net. You also have the te420 set to provide timing, so you will need to configure the cisco side to receive it. |
| 09:19.56 | MCooper | isdn switch_type primary_net5 |
| 09:20.27 | MCooper | how would I set the 420 to be CPE and get timing? |
| 09:21.23 | pputman | MCooper, You will change your zaptel.conf to span=1,1,0,ccs,hdb3,crc4 instead of 1,0,0 if you want it to receive timing. |
| 09:21.39 | pputman | And you need to change your zapata.conf to signalling=pri_cpe instead of pri_net |
| 09:21.59 | MCooper | Ok.. let me give that a try and see what happens... |
| 09:22.16 | MCooper | pputman.. I hope this works... My forehead is hurting. |
| 09:22.19 | pputman | But you have your switchtype on the 420 set as euroisdn. |
| 09:22.41 | pputman | I've never heard of primary_net5, so I don't think zaptel supports it, can you change the switchtype on the cisco side to be euroisdn? |
| 09:23.33 | MCooper | Sure... let me do that also... |
| 09:23.40 | MCooper | Very cool |
| 09:25.17 | MCooper | primary_net5 is the setting in Cisco for Asia, Europe, UK and Assuies |
| 09:25.36 | MCooper | could set it for DPNSS |
| 09:26.56 | pputman | maybe that's what cisco calls euroisdn |
| 09:26.59 | pputman | I dunno |
| 09:28.16 | MCooper | yeah.. I think it is. |
| 09:28.48 | pputman | Did level 2 come up for you? |
| 09:28.58 | MCooper | nO |
| 09:29.10 | MCooper | No... :( |
| 09:29.27 | pputman | what version of zaptel? |
| 09:30.22 | MCooper | SVN-Branch-1.4.r4395 |
| 09:31.01 | MCooper | Is there a better revision to use? |
| 09:31.27 | TheH | Hey guys ! Any one likes to help me with a ISDN issue http://forums.digium.com/viewtopic.php?p=73905#73905 |
| 09:31.40 | pputman | the svn is a development area, I would use zaptel-1.4.11 just in case. |
| 09:32.22 | MCooper | Let me grab it... and I will give that a tray |
| 09:32.26 | MCooper | try |
| 09:33.11 | pputman | MCooper, You could also have a defective card. I would email support@digium.com and open up a case. |
| 09:33.42 | MCooper | I have three cards... same behavior from all three... |
| 09:34.00 | pputman | MCooper, what about the cisco device? Possible failure? |
| 09:34.03 | MCooper | (could I have three bad cards - what is the odds.. but considering I am in Baghdad....) |
| 09:34.09 | MCooper | Tested that also... |
| 09:34.33 | MCooper | we can get a good link off of several other items. |
| 09:35.21 | pputman | MCooper, In that case I would definitely email support. |
| 09:36.33 | MCooper | I agree... let me try this new driver.. and see what happens |
| 09:36.42 | MCooper | You might have hit on the issue |
| 09:37.13 | MCooper | It should not take that long... |
| 09:38.59 | MCooper | There was an error.. I am applying the patch and hope that fixes the error |
| 09:39.07 | *** part/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg) |
| 09:43.33 | zeeesh | <PROTECTED> |
| 09:43.47 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 09:43.57 | creativx | zeeesh: DISA |
| 09:43.57 | MCooper | pputman, and it blows up on the build with a build error... :( |
| 09:44.12 | MCooper | This is getting rather nasty... |
| 09:44.16 | pputman | can you pastebin the error? |
| 09:46.03 | MCooper | http://pastebin.com/m45f247ae |
| 09:46.14 | MCooper | complaining about bool defines... |
| 09:47.17 | lesouvage | I use the A parameter in a dial statement to have a announcement played when the called party pick up the phone. The sip provider sends a 183 sip message and establish the rtp connection. This is interpreted as an "answer" dialstatus and the announcement starts before the phone is picked up. It is the same with the G and M dial parameters. Is this sloppy of the SIP provider or something... |
| 09:47.18 | lesouvage | ...Asterisk like. |
| 09:47.53 | pputman | MCooper, now that's odd, I'm not sure what's happening there. If you do a make clean, and then a ./configure, make, and make install does it go away? |
| 09:48.07 | zeeesh | <creativx>: thnx bro :):) |
| 09:48.13 | MCooper | No.. I already tried that.. twice... |
| 09:48.20 | lesouvage | Dial(SIP/31xxxxxxxxx/${OUTBOUND},40,A(/var/lib/asterisk/sounds/aankondiging)) |
| 09:55.59 | MCooper | pputman, Found the issue... resolved it.,. |
| 09:56.21 | pputman | which was? |
| 09:57.23 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 09:58.49 | *** join/#asterisk tompaw (n=tompaw@slave12.tesserakt.eu) |
| 10:02.15 | TheH | is sad because of http://forums.digium.com/viewtopic.php?t=23689 |
| 10:02.45 | *** join/#asterisk voltagex (n=voltagex@123-243-213-146.static.tpgi.com.au) |
| 10:03.31 | voltagex | hi, I'm unsure what I've done wrong, I have a DID, when I dial into it, the phone rings, but when I test it, the person who picks up the phone only hears an echo of themselves |
| 10:04.25 | Strom_M | the answer is cocks. |
| 10:05.38 | voltagex | oh dear, strom is in freenode as well |
| 10:05.43 | voltagex | ;) |
| 10:06.45 | pputman | hi Strom_M |
| 10:06.57 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:07.28 | MaliutaLap | voltagex: that's about all the answer you'll get without giving us more information, stuff like the dialplan, sip.conf and details on the terminating device |
| 10:09.28 | voltagex | MaliutaLap: yep, before I do that I'm checking the simple stuff, and I think I've solved it.... |
| 10:11.11 | *** join/#asterisk Segnale007 (n=Segnale0@host15-242-dynamic.9-79-r.retail.telecomitalia.it) |
| 10:12.44 | MaliutaLap | voltagex: so "cocks" was enough for you to fix it? |
| 10:14.56 | *** join/#asterisk _khan (i=_Khan@202.133.65.159) |
| 10:25.26 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:26.17 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 10:26.28 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 10:27.04 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000517.dsl.bell.ca) |
| 10:38.51 | _gm | will asterisk get support stun in near future? |
| 10:39.02 | _gm | stund support* |
| 10:43.53 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 10:54.35 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 10:55.38 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 10:59.16 | Datax | _gm: does asterisk currently support client stun ? |
| 10:59.28 | _gm | nops |
| 10:59.52 | Datax | didn't think so :p |
| 11:07.40 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 11:09.35 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 11:10.03 | *** join/#asterisk Segnale007 (n=Segnale0@host136-253-dynamic.42-79-r.retail.telecomitalia.it) |
| 11:10.21 | Delvar | hi anyone around who can delete a crappy trace from my bug on bugs.digium.com? |
| 11:10.46 | *** join/#asterisk oilinki (n=oil@ppp-124-120-6-247.revip2.asianet.co.th) |
| 11:12.18 | tzafrir_laptop | Delvar, which bug? |
| 11:12.36 | Delvar | 7844 |
| 11:12.38 | tzafrir_laptop | And what specific trace in it? |
| 11:12.50 | Delvar | the file name is t38_ht496.log |
| 11:13.11 | Delvar | file_id=12266 |
| 11:13.49 | tzafrir_laptop | Any reason you want to remove it now? |
| 11:19.25 | *** join/#asterisk zydoon (n=zydoon@41.225.159.197) |
| 11:19.30 | zydoon | hi |
| 11:20.05 | zydoon | I have a TDM2422b on production server |
| 11:20.08 | zydoon | in tunisia |
| 11:20.30 | zydoon | I had to change the source code of zaptel to make it work on our "european" telco network |
| 11:20.52 | zydoon | I had to do this : |
| 11:20.55 | zydoon | kernel/wctdm.c |
| 11:20.55 | zydoon | 82,84c82,84 |
| 11:20.55 | zydoon | < {19,6,"RING_V_OFF",0x0000}, |
| 11:20.55 | zydoon | < {20,7,"RING_OSC",0x7EF0}, |
| 11:20.55 | zydoon | < {21,8,"RING_X",0x0160}, |
| 11:20.56 | zydoon | --- |
| 11:20.58 | zydoon | > {19,6,"RING_V_OFF",0x42AB}, |
| 11:21.00 | zydoon | > {20,7,"RING_OSC",0x79BC}, |
| 11:21.02 | zydoon | > {21,8,"RING_X",0x047C}, |
| 11:21.08 | *** join/#asterisk matrix1233 (n=Administ@41.227.2.251) |
| 11:21.18 | zydoon | do I still need it everytime I need to upgrade ? |
| 11:21.22 | matrix1233 | hello |
| 11:21.28 | zydoon | I cannot test it on a production server |
| 11:21.43 | zydoon | hello ... someone can help |
| 11:22.30 | matrix1233 | any one can help me .. i wanna to install the H263 in my asterisk to can use video for my user |
| 11:22.38 | matrix1233 | any help :) |
| 11:22.54 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:23.27 | zydoon | might be a start: http://www.voip-info.org/wiki/view/Asterisk+video |
| 11:23.32 | zydoon | for h264 |
| 11:24.24 | zydoon | hello any zaptel help ? |
| 11:24.25 | matrix1233 | thx zydoon |
| 11:24.38 | zydoon | welcome :) |
| 11:24.48 | matrix1233 | but i cant found the right step to do that |
| 11:24.52 | matrix1233 | :) |
| 11:26.29 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 11:27.43 | matrix1233 | any anadher suggest fom h263 |
| 11:27.44 | matrix1233 | ? |
| 11:29.00 | iratik | Is there a way to make playback somehow inteface with festival so that when a sound file is not found, festival is triggered on the first argument passed to playback? Like Playback("Westbend, Indiana and Columbus, Ohio") ... instead of returning -1 because no such sound file exists..passes cmd to Festival? |
| 11:30.01 | matrix1233 | any one can help me .. i wanna to install the H263 in my asterisk to can use video for my user |
| 11:30.47 | matrix1233 | ? |
| 11:37.09 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 11:37.21 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 11:38.07 | MatBoy | my outgoing sip trunk works very ok, but incomming, i get a 'disconnected connection tone' is my inbound route not ok / |
| 11:39.35 | matrix1233 | any one have installer the vido in asterisk ? |
| 11:40.32 | *** join/#asterisk Uatec (n=uatec@adsl.ntsols.com) |
| 11:40.35 | Uatec | Hello there |
| 11:40.52 | Uatec | is there a command that i can use which will show all my sip peers and the user agent they were last recorded to have used? |
| 11:44.14 | MatBoy | mhh, weird that my inbound sip does not work |
| 11:44.56 | iratik | Does anyone have an example of the correct AGI cmd to execute a system command? I'm a little lost on the escaping |
| 11:45.32 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.132) |
| 11:47.50 | iratik | EXEC System "echo \"Atlanta, Georgia\" | text2wav ..." ? |
| 11:48.02 | *** part/#asterisk zydoon (n=zydoon@41.225.159.197) |
| 11:52.19 | matrix1233 | asterisk and video |
| 11:52.20 | matrix1233 | ? |
| 11:52.24 | matrix1233 | any suggest ? |
| 11:52.40 | matrix1233 | no one have installed the video on asterisk ? |
| 11:53.05 | MatBoy | mhh, call is comming in but 'connection not available' tone |
| 11:54.44 | MatBoy | anyone any suggestion / |
| 11:55.24 | *** join/#asterisk igascream (n=igascrea@80.179.192.178.cable.012.net.il) |
| 11:56.06 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:56.06 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 11:56.26 | lmadsen | morning y'all |
| 11:56.31 | MatBoy | hi 1 |
| 11:56.36 | MatBoy | good afternoon |
| 11:56.47 | MatBoy | sorry, my function buttons died ;0 |
| 11:56.49 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:56.56 | MatBoy | so a quastionmark is / |
| 11:57.02 | MatBoy | 8questionmark |
| 11:57.16 | MatBoy | puzzled: do you use the voip-out of xs4all / |
| 11:58.36 | puzzled | MatBoy: I have one of their 087 numbers registered on my * box if that's what you mean |
| 11:59.18 | MatBoy | puzzled: ah nice, i am not able to get the incomming calls working, i get an 'afgesloten' tone, but the call is comming in on the server |
| 11:59.48 | puzzled | lemme check my config |
| 11:59.58 | MatBoy | would be nice |
| 12:00.05 | MatBoy | i can call ;0 thatÅ nice already |
| 12:00.27 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:01.13 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
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| 12:06.35 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:06.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:06.41 | puzzled | MatBoy: http://pastebin.com/d4ef84352 |
| 12:07.43 | MatBoy | puzzled: ok, thanks, so in elastix, so freepbx also, the user part in a trunk can be left empty / |
| 12:07.55 | igascream | hi all have some problem... I have a fax on one line with asterisk so when I try to send the fax asterisk answer the call . Is it possible to make asterisk not to do it ??? |
| 12:08.31 | puzzled | MatBoy: what do you meanwith "user part"? |
| 12:09.03 | MatBoy | puzzled: when you add a trunk, you have the 'trunk part' and a user part |
| 12:09.21 | puzzled | ah right. I'm not familiar with elastix and freepbx |
| 12:09.55 | MatBoy | ah ok ;0 |
| 12:10.01 | igascream | Can I make asterisk devide between incoming and outbounding calls? |
| 12:10.19 | puzzled | igascream: yes use different contexts |
| 12:10.55 | igascream | puzzled: what do you mean? |
| 12:11.21 | puzzled | igascream: do yourself a favor and read the asterisk book if you do not know what contexts are |
| 12:11.24 | puzzled | ~tfot |
| 12:11.24 | 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 |
| 12:11.38 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:11.53 | puzzled | igascream: buy it or download it the links above |
| 12:12.08 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:12.30 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583112.dsl.bell.ca) |
| 12:13.12 | MatBoy | puzzled: mhh, still incomming but 'afgesloten' tone |
| 12:14.04 | igascream | puzzled: I know what the contexts is but I don't understand how can it help me? |
| 12:14.17 | puzzled | MatBoy: think I had that too until I slapped the dsl modem and disabled the sip helper stuff |
| 12:14.33 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:15.19 | puzzled | igascream: you don't understand contexts if you don't see how using separate contexts for inbound/outbound can divide calls |
| 12:15.51 | MatBoy | puzzled: mhh, but my modem is in bridge to an zyxel |
| 12:15.56 | MatBoy | *a zyxel |
| 12:16.02 | MatBoy | fixed his keyboard ;) |
| 12:16.21 | *** part/#asterisk real_epicac (n=IceChat7@136.240.13.217.in-addr.dgcsystems.net) |
| 12:16.29 | puzzled | MatBoy: and the zyxel is a router I assume? |
| 12:16.39 | MatBoy | puzzled: yap |
| 12:16.44 | MatBoy | everything forwarded and open |
| 12:17.17 | puzzled | the config I pastebin'ed works for me. I don't know elastix or freepbx so can't help you with that |
| 12:19.15 | igascream | puzzled: I think you don't understand the problem I have a telephone on the same line with asterisk so when I try to call using this phone asterisk answers the call thinking that is incoming call |
| 12:19.36 | MatBoy | puzzled: elastix is actually an extention ON asterisk, so should be quite the same |
| 12:20.12 | *** part/#asterisk bipser (n=bipser@u34-10.dsl.vianetworks.de) |
| 12:21.30 | puzzled | igascream: ah right. I have no idea but this has been asked before on the mailinglist so maybe search the list archives |
| 12:22.09 | *** join/#asterisk syslogd (n=syslogd@pD955F330.dip.t-dialin.net) |
| 12:22.22 | [TK]D-Fender | igascream: You have a phone sharing the same line as and FXO interface on your * box? |
| 12:23.29 | syslogd | Can I use DeTeWe TA 33 USB for Asterisk? |
| 12:23.55 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 12:24.24 | igascream | [TK]D-Fender: yes I use phone for outbounding calls and asterisk for incomming |
| 12:25.04 | [TK]D-Fender | igascream: so * thinks you have an incoming call when you place an outgoing call from that separate phone? Is it analog? |
| 12:25.29 | *** join/#asterisk CVirus (n=Burzum@196.218.41.31) |
| 12:25.34 | [TK]D-Fender | syslogd: No. |
| 12:25.43 | igascream | [TK]D-Fender: yes analog |
| 12:26.03 | [TK]D-Fender | igascream: pastebin your zapata.conf and all files linked to it. |
| 12:26.11 | [TK]D-Fender | ~pb |
| 12:26.11 | 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 |
| 12:29.41 | syslogd | [TK]D-Fender: Ok, thanks. Is there a cheap USB adapter that you can recommend? |
| 12:30.02 | syslogd | [TK]D-Fender: Unfortunately I do not have a PCI slot left so that I have to go with USB. |
| 12:30.15 | [TK]D-Fender | syslogd: there is virtually NO USB devices compatible with *. |
| 12:30.25 | _khan | Asterisk sense the call is answered but actually it is ringing. Problem is on analogue line. Any help please.... |
| 12:30.26 | igascream | [TK]D-Fender: http://pastebin.com/d76f4357c |
| 12:30.26 | [TK]D-Fender | syslogd: Get an ATA or gateway then. |
| 12:31.10 | [TK]D-Fender | igascream: the ENTIRE zapata and everything linked to it <- |
| 12:31.22 | *** join/#asterisk ManxPower (n=manxpowe@195.sub-75-201-209.myvzw.com) |
| 12:31.44 | igascream | [TK]D-Fender: thats all I have in zapata |
| 12:32.12 | [TK]D-Fender | igascream: Impossible. You don't even have a channel delcaration in there. |
| 12:32.26 | [TK]D-Fender | declaration* |
| 12:33.45 | iratik | Trying to execute a system command via AGI and having some trouble ... maybe i'm not escaping something correctly Here is the AGI log and a display of thet problem ... can anyone take a look to see if there is something that sticks out as wrong? http://www.pastie.org/230633 |
| 12:33.58 | _khan | Call is answered by asterisk when it is ringing. on PSTN line. |
| 12:34.03 | [TK]D-Fender | _khan: You would need to add "callprogress=yes" to your zapata.conf for * to follow progress tones from your telco to know that the line is "ringing". This often causes random disconnects however. |
| 12:34.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 12:35.23 | igascream | [TK]D-Fender: http://pastebin.com/m13bd489f |
| 12:35.40 | igascream | [TK]D-Fender: thats all I have |
| 12:35.43 | _khan | <[TK]D-Fender> I already added callprogress=yes but sometimes it shows ringing sometimes not, but after second ring asterisk sense that call is being answered while it is still in ringing |
| 12:35.51 | [TK]D-Fender | iratik: Thats retarded. You're already in AGI outside of * yet you are trying to tell * to run a system command? do it DIRECT from your AGI! |
| 12:36.11 | matrix1233 | vido on asterisk ?? |
| 12:36.16 | matrix1233 | video ? |
| 12:36.21 | [TK]D-Fender | _khan: Then either you did not set the correct tonezone for your region or * is having trouble following it. |
| 12:36.24 | matrix1233 | any suggest |
| 12:36.36 | iratik | [TK]D-Fender: text2wave is not on the server running the AGI script... festival is only installed on the asterisk server |
| 12:36.45 | [TK]D-Fender | matrix1233: Ekiga or eyeBeam softphones <- |
| 12:37.09 | _khan | [TK]D-Fender where to set the tonezone?? |
| 12:37.19 | [TK]D-Fender | _khan: zaptel.conf |
| 12:37.25 | *** join/#asterisk Segnale007 (n=Segnale0@host136-253-dynamic.42-79-r.retail.telecomitalia.it) |
| 12:37.30 | *** join/#asterisk StooJ (n=stooj@johnston37.plus.com) |
| 12:37.35 | iratik | [TK]D-Fender: I suppose I could create a local script on the server running the AGI... to speak to a php script on the asterisk box to run the same system command .... but that seems convoluted |
| 12:38.11 | [TK]D-Fender | iratik: or copy the file over, etc... |
| 12:38.28 | [TK]D-Fender | iratik: either way I think its probably a spacing issue for the parameters |
| 12:38.49 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 12:38.53 | [TK]D-Fender | igascream: What built your configs? |
| 12:39.36 | igascream | [TK]D-Fender: what do you mean? |
| 12:39.48 | iratik | [TK]D-Fender: There is only one parameter being passed to system here? or is asterisk interpreting that pipe as the parameter delimiter |
| 12:39.52 | [TK]D-Fender | igascream: did you build all your configs by hand yourself? |
| 12:40.04 | igascream | yes |
| 12:40.09 | [TK]D-Fender | iratik: I'm wondering first about the whitespace. |
| 12:40.25 | syslogd | [TK]D-Fender: I think I will get an ATA. What do you think of a Wi-Fi ATA such as WVTR-141 (http://www.sparklan.com/product_details.php?prod_id=9)? Are there any better/cheaper devices? |
| 12:40.26 | ManxPower | _khan: callprogress will randomly disconnect your calls. |
| 12:40.29 | [TK]D-Fender | igascream: then you've done something wrong. You don't have a channel declaration in there. |
| 12:40.41 | iratik | Spacing in parameters is only an issue when there are more than one parameter... for example Festival('hello world', 'any') does not work... but Festival('hello world','any') does work |
| 12:41.14 | [TK]D-Fender | syslogd: Are you looking to connect this to a PHONE? Or a LINE? |
| 12:41.27 | iratik | System("echo 'hello world' | text2wave -o out.ulaw") ... thats just one parameter... so spacing shouldn't be a problem |
| 12:41.32 | [TK]D-Fender | iratik: formatted that way in AGI? |
| 12:41.38 | ManxPower | iratik: no. |
| 12:41.46 | [TK]D-Fender | iratik: remember thats AGI spacing, not as you'd have it in dialplan |
| 12:42.11 | iratik | [TK]D-Fender: what do you think... I don't know the translation between what i see on AGI debug and what the equivalent command would be in the dialplan |
| 12:42.15 | ManxPower | iratik: your best bet is to make a tiny shell script and System(/usr/local/bin/happy.sh) |
| 12:42.30 | iratik | even when i turn verbosit on to see the system command execute ... it doesn't give me the literal command |
| 12:42.48 | ManxPower | iratik: It's virtually impossible to quote things in Asterisk. |
| 12:43.56 | syslogd | [TK]D-Fender: I want to connect a analog telephone. |
| 12:43.56 | [TK]D-Fender | iratik: what I'd suggest : place a local system call to issue that command via SSH <- |
| 12:44.31 | igascream | [TK]D-Fender: how can I know which channel type I have to declare? |
| 12:44.50 | iratik | understood. .. I'll make a php script on the asterisk box that takes a text string and a timestamp and writes it to /var/lib/asterisk/sounds ... then my AGI script can play the tmp file with that filename ... ! Because Festival 'hello world'|'any' is just returning 0 when i press the keys... I'm starting to think that passing multiple arguments to anything through AGI EXEC ... is going to be difficult |
| 12:45.09 | [TK]D-Fender | syslogd: I highly recommend you avoid trying to penny pinch your purchase You may very well end up with a real piece of junk. Linksys PA2T's are rather inexpensive. |
| 12:45.42 | [TK]D-Fender | igascream: You aren't showing me your entire config related to your zaptel channels.... |
| 12:46.23 | igascream | [TK]D-Fender: what else related to it? |
| 12:46.43 | *** join/#asterisk zapp-branigan (n=malebolg@9.218.216.87.static.jazztel.es) |
| 12:46.47 | [TK]D-Fender | igascream: these are your configs. You'd better get a clue as to what you are doing. |
| 12:46.49 | MaliutaLap | is very happy with his TMD400P |
| 12:47.24 | ManxPower | igascream: your configs, as shown to us, are not valid in any way. |
| 12:47.34 | zapp-branigan | hello asterisk 1.6 have g729 codec from digium ? i can't find this in the web page |
| 12:47.48 | MaliutaLap | igascream: it's rather simple, you know if you have an fxo or and fxs is a particular slot |
| 12:47.51 | ManxPower | zapp-branigan: You mean 1.6BETA, right? |
| 12:47.56 | zapp-branigan | yes |
| 12:48.11 | ManxPower | zapp-branigan: have you looked for it on Digium' |
| 12:48.15 | ManxPower | s site? |
| 12:48.21 | igascream | MaliutaLap: only fxo |
| 12:48.59 | MaliutaLap | igascream: that determines how you declare a channel in zapata.conf and zaptel.conf |
| 12:49.08 | zapp-branigan | http://downloads.digium.com/pub/asterisk/g729/ |
| 12:49.27 | ManxPower | igascream: put the output of "ztcfg -vvv" in pastebin.ca as well as /etc/zaptel.conf |
| 12:49.41 | zapp-branigan | http://downloads.digium.com/pub/telephony/codec_g729/ |
| 12:49.52 | zapp-branigan | here is a asterisk-B.1/ |
| 12:49.58 | zapp-branigan | but not work |
| 12:50.03 | ManxPower | zapp-branigan: looks like it's not supported yet. |
| 12:50.20 | zapp-branigan | ok thanks |
| 12:50.22 | ManxPower | zapp-branigan: don't be so suprized -- 1.6 is not supposed to be used in production until it's releases. |
| 12:50.43 | zapp-branigan | ok thanks |
| 12:51.07 | ManxPower | igascream: I'm doing this for free so don't be slow |
| 12:51.34 | zapp-branigan | and there is any way to install g723.1 in the asterisk |
| 12:51.45 | [TK]D-Fender | ManxPower: http://pastebin.com/m13bd489f <-- what he gave when I asked the same |
| 12:51.55 | [TK]D-Fender | ManxPower: for zapata. |
| 12:52.13 | [TK]D-Fender | ManxPower: Claiming that that is all there is. |
| 12:52.29 | ManxPower | [TK]D-Fender: but I was looking for zaptel.conf, not zapata.conf |
| 12:52.48 | ManxPower | [TK]D-Fender: he's either lying or confused -- either way I want to catch it. |
| 12:52.57 | [TK]D-Fender | ManxPower: I know... |
| 12:53.28 | ManxPower | zapp-branigan: no, G723.1 is not supported in Asterisk. You could create a G723.1 codec, but you'll still need a license, which runs about US$10,000 |
| 12:53.39 | [TK]D-Fender | ManxPower: He has an phone in PARALLEL with his FXO port and says when he calls OUT on it, * picks up. I'd bet on "immediate=yes", or a flakey card/telco (he's IT) |
| 12:53.51 | ManxPower | [TK]D-Fender: igascream obviously doesn't want help, as he's not trying |
| 12:54.02 | zapp-branigan | :) ok thanks |
| 12:57.39 | ManxPower | [TK]D-Fender: I suspect a debounce issue. |
| 12:58.13 | [TK]D-Fender | ManxPower: like a telco-flash on answer/accept? |
| 12:58.23 | ManxPower | WaitForRing would pretty much eliminate his issue, but since he doesn't want help.... |
| 12:58.55 | ManxPower | [TK]D-Fender: no, just the very slight voltage change when he picks up the phone triggering Asterisk |
| 12:59.34 | *** join/#asterisk zydoon (n=zydoon@41.225.159.197) |
| 12:59.43 | [TK]D-Fender | ManxPower: Ah, so it takes the first dip rather that a full-wave, etc to indicate the start? |
| 12:59.55 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 13:00.52 | ManxPower | [TK]D-Fender: that is what I suspect. This is not the first time I've heard of similar problems. |
| 13:01.12 | [TK]D-Fender | ManxPower: interesting... learn something new every day... |
| 13:01.55 | *** join/#asterisk Peri (n=redanti@surf99.net.rss.rogers.com) |
| 13:02.52 | ManxPower | [TK]D-Fender: most people don't put phones on the same line outside of Asterisk |
| 13:04.04 | [TK]D-Fender | ManxPower: He seems to be trying to use * as a fax machine. |
| 13:04.25 | [TK]D-Fender | ManxPower: (say from another PB) |
| 13:04.28 | [TK]D-Fender | saw* |
| 13:04.41 | MaliutaLap | having a phone on the same in house line as * would kind of defeat the purpose |
| 13:05.47 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:06.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.22 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:06.24 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 13:06.24 | *** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 13:06.49 | ManxPower | [TK]D-Fender: I won't be helping him further -- he has wasted too much of my time. |
| 13:07.00 | shido6 | ouch |
| 13:07.31 | ManxPower | shido6: If someone wants my help for free they had better damn well stick around. |
| 13:07.41 | shido6 | heh |
| 13:07.49 | shido6 | loved you and left u? |
| 13:08.04 | ManxPower | If I was being paid for this I'd be happy to do nothing while waiting. |
| 13:08.25 | shido6 | oh crap |
| 13:09.08 | *** part/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 13:09.38 | [TK]D-Fender | ManxPower: My favourite : "Oh, the system is at work and I can't access it remotely. I just thought you'd be psychic and be able to tell me exactly how to fix my problem based on the the vaguest recollection of a meaningless error message I have left to give you. Guess I'll come back later" |
| 13:10.20 | *** part/#asterisk zydoon (n=zydoon@41.225.159.197) |
| 13:13.46 | defswork | anyone recommend a desktop handset that has 20+ BLF |
| 13:15.41 | syslogd | [TK]D-Fender: Thanks for all your help. I think I'll simply use Ekiga until the prices are lower. |
| 13:16.55 | [TK]D-Fender | defswork: Polycom IP 6XX + 2 sidecars |
| 13:17.44 | defswork | [TK]D-Fender: this if for all desktops - not just a reception - so sidecars take up too much real estate |
| 13:18.03 | defswork | I think I need to convince them that they don't need to see all extension status on all phones |
| 13:18.14 | iratik | proposes an ISO standard for string escaping schemas ... a bit that identifies an escape level where there would be standardized 12 escape levels ... this way I don't have to double and quadruple escape strings to pass through multiple systems! |
| 13:18.17 | [TK]D-Fender | defswork: that IS psycho |
| 13:18.32 | defswork | yeah but they have it on their current system |
| 13:18.35 | [TK]D-Fender | defswork: Give them a web-panel instead |
| 13:18.37 | defswork | so they assume they need it |
| 13:18.44 | defswork | yeah I've told them about that |
| 13:18.59 | defswork | they were initially ok - then said - but what if i'm not logged into my pc :o |
| 13:19.11 | defswork | they dial and get engaged! |
| 13:19.13 | defswork | then* |
| 13:19.35 | defswork | their current system is over 10 years old |
| 13:19.39 | defswork | but has more lights!! |
| 13:20.21 | [TK]D-Fender | defswork: Next runner up : Aastra 57i |
| 13:20.47 | [TK]D-Fender | defswork: they'll have to scroll for it, but might work. I know you can ahve a few pages of soft keys, not sure on the exact # through |
| 13:20.53 | defswork | yeah - now they've fixed all the bugs |
| 13:21.37 | kaii | Aastra desk phones suck.. |
| 13:21.50 | defswork | kaii: I've had no problems apart from with a 55i |
| 13:22.03 | defswork | which after 4 firmware releases is now apparently ok |
| 13:22.41 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:23.03 | [TK]D-Fender | defswork: I hate the feel of them, but BLF seems to be the only important thing for those schmucks. |
| 13:23.38 | defswork | I tried to convince them to go to voicemail so they could just blind xfer all calls :) |
| 13:23.50 | defswork | they didn't want that |
| 13:24.13 | TheH | Would anybody be so kind to look at this trace and tell me what might be wrong ? http://pastebin.com/m1b664a97 |
| 13:24.18 | defswork | anyone used Doro phones ? I've just noticed them on voipon.co.uk |
| 13:24.23 | defswork | seems cheap |
| 13:26.52 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:29.07 | puzzled | defswork: that Doro IP840c phone looks like a Snom 360 |
| 13:29.25 | ManxPower | theH P[ 1] Â --> channel:0 mode:TE cause:16 ocause:16 rad: cad: |
| 13:29.34 | ManxPower | Can you guess what cause 16 is? |
| 13:30.37 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-7deb2e0b03c31f6e) |
| 13:30.41 | TheH | Manx: Cause 16 is "normal" hangup |
| 13:30.48 | *** join/#asterisk kippi (n=kippi@untrust-gct.equinoxit.net) |
| 13:30.50 | kippi | hey |
| 13:31.00 | TheH | Manx: Normal call clearing right |
| 13:31.00 | ManxPower | theH: Exactly. |
| 13:31.14 | ManxPower | theH: sounds like you have some other issue. |
| 13:31.39 | TheH | ManX: And that is the problem , how come it does normal call clearing ? Even with incoming calls it rings once and then it drops |
| 13:32.25 | ManxPower | theH: No idea. So few people use mISDN..... |
| 13:32.44 | TheH | is there anything else i can try which will work with B410P / ISDN2e |
| 13:33.10 | ManxPower | theH: I have no idea. |
| 13:33.14 | TheH | :( |
| 13:33.19 | kippi | I believe DBPut has been replaced in version 1.4, is there any documents on what the replacement is in 1.4 |
| 13:33.22 | ManxPower | theH: your Dial like looks OK to me. |
| 13:33.31 | ManxPower | kippi: you mean like the info in upgrade.txt ? |
| 13:33.32 | puzzled | TheH: which misdn version are you using? |
| 13:34.50 | TheH | puzzled: The latest one was installed using zaptel /make b410 which is version : misdn 1_1_7 |
| 13:35.08 | [TK]D-Fender | kippi: "core show function DB". |
| 13:35.13 | TheH | puzzled : although it still uses the misdn-init instead of the mISDN |
| 13:35.29 | [TK]D-Fender | kippi: And before asking like that you should always scan the complete function & application lists. |
| 13:36.08 | kippi | [TK]D-Fender: sorry, will look next time |
| 13:36.17 | puzzled | TheH: 1.1.7 is not the latest. 1.1.8 is. Try that one and see if it works: http://www.misdn.org/downloads/releases/ |
| 13:36.29 | TheH | ok |
| 13:36.39 | puzzled | TheH: alternatively you can try the new misdn development stuff: http://www.linux-call-router.de/download/lcr-1.0/ |
| 13:36.40 | TheH | so should i rebuild asterisk again |
| 13:36.48 | ManxPower | What I want to know is what Digium was thinking when they released a non-zaptel card. |
| 13:36.53 | kippi | thanks for the help |
| 13:36.54 | puzzled | TheH: no. only chan_misdn |
| 13:37.01 | TheH | puzzled: ok thanks |
| 13:37.03 | tzafrir_laptop | puzzled, what drivers are supported with the development branch? |
| 13:37.19 | *** join/#asterisk twisla (n=twisla@kasteel.twis.la) |
| 13:37.25 | puzzled | tzafrir_laptop: sorry don't understand your question |
| 13:37.47 | twisla | rtp allocatin is done incrementally inside the configured range, or it is randomized ? |
| 13:38.09 | ManxPower | twisla: why do you care? |
| 13:38.26 | twisla | testing some QoS issue |
| 13:39.00 | ManxPower | twisla: I suspect you either have to look at the source code or just try it and see. |
| 13:39.31 | ManxPower | just remember the SOURCE port will be randomlized in many situations. |
| 13:39.55 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 13:40.06 | twisla | yeah, I know, will look at the source, thanks |
| 13:40.08 | ManxPower | Your QoS rules should require either the source OR the dest containing the RTP port range, |
| 13:40.13 | puzzled | TheH: also this mailinglist may be more helpful with misdn issues: https://www.isdn4linux.de/mailman/listinfo/isdn4linux |
| 13:40.57 | *** join/#asterisk afink (n=chatzill@72-164-59-242.dia.static.qwest.net) |
| 13:41.28 | puzzled | ManxPower: I agree that it would have been better if Digium had released their ISDN bri card with supported drivers. But afaik those are coming |
| 13:41.37 | TheH | puzzled: Thanks , also make install will just overwrite the chan_misdn or do i need to manual copy the module over |
| 13:41.56 | puzzled | TheH: not sure so to be safe just copy it over |
| 13:42.26 | TheH | ok |
| 13:42.39 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 13:42.49 | hsv-al | . |
| 13:43.12 | hsv-al | everyone coffee'd up for another day at our monitors on the internet? :) |
| 13:43.12 | MaliutaLap | some people piss me off, the complete lack of detail in product descriptions |
| 13:43.23 | afink | Hello, is there a simple way to include all of the local, longdistance, international contexts for dialing out of the default context? I have tried to simply #include local and #include longdistance but they don't seem to do any pattern matching. thanks |
| 13:43.26 | MaliutaLap | "it's a DECT handset" "it does VoIP" |
| 13:43.55 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 13:44.03 | ManxPower | afink: #include is basically the same as copying and pasting the info into the main file. include => deals with contexts |
| 13:45.14 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:45.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:45.15 | afink | My apologies, that is actually what I have include => local |
| 13:45.48 | ManxPower | afink: A single mistake like that could cost you tens of thousands of dollars if your mistake would let people call thru your system and have you pay for it. |
| 13:45.58 | MaliutaLap | afink: you might want to think about pastebining something for us to look at |
| 13:46.42 | MaliutaLap | ManxPower: I have given up on try to reason with people who ask for dangerous/stupid things |
| 13:47.10 | ManxPower | MaliutaLap: A good policy. |
| 13:47.14 | MaliutaLap | ManxPower: if that's what they want, I'll help them get there. any damage is resulting from their stupidity |
| 13:48.01 | MaliutaLap | I have been asked to do too many stupid things in my time, I only fight the good fight when it's potentially my ass on the line |
| 13:48.16 | ManxPower | MaliutaLap: I expect more and more news stories about unsecured PBXs setup by newbies being compromised and having $10,000 phone bills because someone figured out a PBX was not secured. |
| 13:48.18 | TheH | ManX: Just did the upgrade but still the same problem :( this is my trace of a incoming call which rings my softphone once and then hangsup with a BT message saying the number is out of order http://pastebin.com/m6a567656 |
| 13:49.03 | afink | Guys, typo in IRC not in asterisk... |
| 13:49.12 | ManxPower | [Jul  9 14:42:50] WARNING[29505]: chan_sip.c:2921 create_addr: No such host: x47                                        22 |
| 13:49.15 | ManxPower | there is your error |
| 13:49.18 | afink | late night, early morning |
| 13:49.42 | ManxPower | theH: you do not have a [x4722] in sip.conf |
| 13:49.42 | Peri | ManxPower: I expect to not hear about all the people who's boxes were compromised by running those horrible insecure PBXs as root |
| 13:50.03 | [TK]D-Fender | ManxPower: the page is split there. |
| 13:50.20 | [TK]D-Fender | TheHstop using stupid GUI's to access that info and use a real SSH client. |
| 13:50.22 | TheH | Manx: That should not matter cause its a x4722 entry we use for roaming sip phones (wireless mobiles) |
| 13:50.22 | ManxPower | theH: next time you do a pastebin include the dialplan CLI stuff, not just the mISDN debug |
| 13:50.25 | *** part/#asterisk syslogd (n=syslogd@pD955F330.dip.t-dialin.net) |
| 13:50.34 | MaliutaLap | afink: you still haven't given us any information that might allow us to help |
| 13:50.38 | [TK]D-Fender | TheH: and alwayr pastebint he ENTIRE call. |
| 13:50.38 | TheH | TK: Putty is not a reall ssh client ? |
| 13:50.42 | ManxPower | theH: It does matter |
| 13:51.06 | [TK]D-Fender | TheH: Your pastebin shouldn't be split like that |
| 13:51.18 | ManxPower | theH: put the correct [x4722] in sip.conf and make the phone register |
| 13:51.26 | [TK]D-Fender | TheH: and you aren't showing things like the full dial attempt or any configs to back it up |
| 13:51.59 | afink | MaliutaLap: There is really nothing in my default context except include => longdistance, atm I am just trying to make an outgoing long distance call. The TRUNK variable is defined as Zap/G2 |
| 13:52.48 | MaliutaLap | afink: so your entire dialplan simply consist of includes? |
| 13:52.49 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 13:52.50 | *** join/#asterisk th0m (n=th0m@pub.a-d-m.fr) |
| 13:52.55 | th0m | hi |
| 13:53.23 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:53.35 | MaliutaLap | afink: you haven't specified anything in other contexts, like [local] and [longdistance] |
| 13:53.56 | afink | MaliutaLap: pastebin on the way |
| 13:54.31 | TheH | Manx: x4722 added and registered , but still the same problem: » Full pastebin http://pastebin.com/m1a683dcc |
| 13:54.59 | *** part/#asterisk th0m (n=th0m@pub.a-d-m.fr) |
| 13:55.10 | *** join/#asterisk J4zen (n=Jeroen@a82-95-153-17.adsl.xs4all.nl) |
| 13:56.24 | [TK]D-Fender | TheH: please turn off the misdn debug, and include SIP DEBUG. |
| 13:57.01 | [TK]D-Fender | TheH: and turn down core debug as well. |
| 13:58.02 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-163-161.dsl.teksavvy.com) |
| 13:59.25 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:00.20 | TheH | [TK]D-Fender : I know you tried to help fender , and i apperciate it but i dont thing your knowledge is anyware close to helping me , i can draw you pretty pictures and you will see something here and there but at the end of the day its a misdn/asterisk issue and not a sip peer issue |
| 14:02.22 | [TK]D-Fender | TheH: right now I don't see a proper call going through, or anything showing me that there should be more dialplan to execute, which would lead to a natural disconnect. It'd be nice to see this for a sanity check... |
| 14:02.47 | *** join/#asterisk Netsnipe (n=alau@wikipedia/Netsnipe) |
| 14:03.28 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 14:03.49 | Netsnipe | hi everyone |
| 14:04.10 | Netsnipe | are sln and sln16 the same format? |
| 14:04.49 | Netsnipe | i.e. does format_sln.so play the .sln16 files in asterisk-core-sounds-en-sln16-current.tar.gz? |
| 14:05.03 | ManxPower | theH: TURN OFF the debug stuff, it is hiding what needs to be seen |
| 14:05.46 | *** join/#asterisk HonestWorker (n=Wothanaz@201.87.225.101) |
| 14:05.59 | HonestWorker | Good morning, gentlemen. How are you doing? |
| 14:06.46 | gr0mit | what makes you assume we are all men? |
| 14:06.46 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 14:07.11 | HonestWorker | We have a nice and neat asterisk setup at my work place and I was willing to configure a WRTP54G to work as an ata for an analog phone. My question is, therefore, not directly related to asterisk. |
| 14:07.26 | HonestWorker | gr0mit, I didnt. I am sorry. Let me extend my greetings to all the ladies as well |
| 14:07.33 | gr0mit | fair enough. |
| 14:07.42 | gr0mit | is a bloke though! |
| 14:07.54 | [TK]D-Fender | HonestWorker: And some of us like it rough! |
| 14:07.58 | phpboy | Anybody here used AgentCallBackLogin() ? |
| 14:08.07 | HonestWorker | My analog phone won't ring. I don't know what kind of configuration is related to the sing signal to be send to the phone. |
| 14:08.15 | ManxPower | theH: pastebin the output of "sip show peers" |
| 14:08.24 | *** part/#asterisk twisla (n=twisla@kasteel.twis.la) |
| 14:09.38 | [TK]D-Fender | HonestWorker: And why should it ring? |
| 14:10.14 | HonestWorker | [TK]D-Fender, I beg your pardon? |
| 14:10.15 | ManxPower | phpboy: I doubt it. I believe it was removed in 1.6 and deprecated in 1.4 |
| 14:10.38 | Netsnipe | HonestWorker: what card are you using? |
| 14:10.39 | HonestWorker | I mean, we need an auditive sound to perceive that someone is calling our line. |
| 14:10.43 | [TK]D-Fender | HonestWorker: You're saying it won't ring. Show us your attempt to call it. |
| 14:10.53 | ManxPower | [TK]D-Fender: he's not using Asterisk. |
| 14:11.04 | ManxPower | HonestWorker: I doubt anyone here will be able to help you. |
| 14:11.05 | phpboy | ManxPower: I'm using 1.2 I see so in the config options. is there is better way to do this? |
| 14:11.06 | HonestWorker | Do you mean the asterisk log entry ? |
| 14:11.16 | phpboy | achieve the same result? |
| 14:11.20 | HonestWorker | I am using asterisk. Its my gateway |
| 14:11.22 | [TK]D-Fender | HonestWorker: What do you mean calling your "line"? that device is used to let you plug in a PHONE, not a LINE |
| 14:11.25 | ManxPower | phpboy: I cannot help you. |
| 14:11.32 | phpboy | :( |
| 14:11.45 | ManxPower | (9:07:12 AM) HonestWorker: We have a nice and neat asterisk setup at my work place and I was willing to configure a WRTP54G to work as an ata for an analog phone. My question is, therefore, not directly related to asterisk. |
| 14:11.45 | HonestWorker | However, my question is related to a WRTP54G, as I have announced earlier, it is not directly related to asterisk |
| 14:12.08 | ManxPower | HonestWorker: why is your WRT not connected to Asterisk? |
| 14:12.16 | [TK]D-Fender | HonestWorker: And that unit appears to be locked to vonage. |
| 14:12.18 | HonestWorker | It is. |
| 14:12.42 | HonestWorker | It is connected and it runs perfectly as far as it concerns initiating calls |
| 14:12.42 | ManxPower | theH: You have 5 mins to respond before I go back to paying work. |
| 14:12.50 | [TK]D-Fender | HonestWorker: You need to provide a MUCH better description of what it is you are doing and what is connected to awhat, in and which manner. |
| 14:13.01 | ManxPower | HonestWorker: Is english not you native language? |
| 14:13.03 | Peri | [TK]D-Fender you can actually purchase them from Linksys without them being vonage locked if you have a purchasing license |
| 14:13.13 | HonestWorker | The problem is no ring signal appears to be sent to the analog phone I have attached to one of its 'lines' |
| 14:13.22 | HonestWorker | Ok |
| 14:13.30 | ManxPower | HonestWorker: you need to put the cli output of a failed call to pastebin. |
| 14:13.36 | HonestWorker | We have an asterisk pbx connected to a E1 |
| 14:13.42 | Peri | HonestWorker: is the mta registering? |
| 14:13.43 | [TK]D-Fender | HonestWorker: Along with SIP DEBUG <- |
| 14:13.53 | TheH | Manx: I am collecting all configs 1 second. |
| 14:14.03 | HonestWorker | Yes, I can see the registration at asterisk |
| 14:14.09 | HonestWorker | I can initiate calls and receive calls |
| 14:14.16 | HonestWorker | I just cant hear a ring signal |
| 14:14.23 | Netsnipe | you mean a dial tone? |
| 14:14.28 | ManxPower | HonestWorker: stop talking and provide the requested niformation |
| 14:14.29 | HonestWorker | Its not related to asterisk. It has to do with the WRTP4G |
| 14:14.43 | ManxPower | HonestWorker: Then why are you here? |
| 14:14.50 | Netsnipe | define "ring signal" |
| 14:14.52 | HonestWorker | ManxPower, calm down. |
| 14:15.03 | Peri | HonestWorker: perhaps you should check their support, not many of us are overly familiar with their products |
| 14:15.05 | ManxPower | HonestWorker: it is either connected to asterisk or it's not. If it's connected to Asterisk then it's related to Asterisk |
| 14:15.12 | HonestWorker | The audible sound an analog telephone generates upon receiving a phone call |
| 14:15.14 | Peri | bbiab |
| 14:15.36 | ManxPower | HonestWorker: that is not your problem. your problem is that the call is not getting to the ATA |
| 14:15.45 | [TK]D-Fender | HonestWorker: show us the CALL <- |
| 14:16.14 | HonestWorker | ManxPower, I beg to disagree. The asterisk part has to do with the WRTP54 registering and acting as an extension. The interface between the WRTP54 and the analog phone has nothing to do with asterisk |
| 14:16.22 | ManxPower | theH: exactly how long does it take to paste the output of "sip show peers" |
| 14:16.35 | ManxPower | HonestWorker: I cannot help you further. |
| 14:16.43 | HonestWorker | The call is getting to the ata because if I remove the 'headset' from the telephone base, I can talk to the caller |
| 14:16.52 | Netsnipe | do I need to rename all the .sln16 files that come with asterisk-core-sounds-en-sln16-current.tar.gz to .sln? |
| 14:16.52 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 14:17.11 | HonestWorker | I will provide you all the information needed. I am sorry for the trouble. I am a beginner. |
| 14:17.26 | ManxPower | HonestWorker: perhaps [TK]D-Fender can help you. |
| 14:17.35 | Netsnipe | I only see a format_sln.so module and no format_sln16.so |
| 14:17.36 | HonestWorker | I bought Asterisk : The future of Telephony and I am reading it. I have made it to chapter 3 yet. |
| 14:18.02 | ManxPower | Netsnipe: why don't you just TRY playing one of the sound file. |
| 14:18.03 | [TK]D-Fender | HonestWorker: maybe you turned the ringer off on the phone. |
| 14:18.14 | Netsnipe | ManxPower: because I'm offsite = P |
| 14:18.18 | TheH | ManX: http://pastebin.com/m4a5473bf (tracewithout debug of misdn) |
| 14:18.29 | [TK]D-Fender | Netsnipe: ...... |
| 14:18.36 | [TK]D-Fender | ManxPower: My favourite : "Oh, the system is at work and I can't access it remotely. I just thought you'd be psychic and be able to tell me exactly how to fix my problem based on the the vaguest recollection of a meaningless error message I have left to give you. Guess I'll come back later" |
| 14:18.40 | [TK]D-Fender | Netsnipe: ^^^^ |
| 14:18.48 | ManxPower | theH: Where. Is. The. Output. Of. "sip show peers"?????????????????????????? |
| 14:18.59 | Netsnipe | ManxPower: I've ssh'ed into the asterisk box and it's currently unplugged from the phone system at the office |
| 14:19.01 | TheH | Manx: Sip show peers http://pastebin.com/m33c0374a |
| 14:19.34 | ManxPower | theH: it looks like your problem is solved. |
| 14:19.53 | TheH | manx: Explain |
| 14:20.07 | ManxPower | theH: the last pastebin contained no errors |
| 14:20.16 | TheH | sip show peers ? |
| 14:20.19 | TheH | or the dial ? |
| 14:21.03 | ManxPower | Do you see this " (Unspecified)" in that "sip show peers? That means DEVICE NOT REGISTERED CANNOT CALL DEVICE |
| 14:21.11 | ManxPower | theH: in the dial; |
| 14:21.22 | ManxPower | both devices are ringing according to the CLI output |
| 14:21.22 | Netsnipe | [TK]D-Fender: I'm not troubleshooting, just curious whether .sln16 files are the same as .sln |
| 14:21.42 | russellb | they are not |
| 14:21.44 | TheH | Manx: In the dial you see it is ringing but it rings only 1 and than just stops and you get the beep beep beep problem |
| 14:21.51 | russellb | .sln == 8 kHz, .sln16 == 16 kHz |
| 14:22.00 | russellb | both linear format, though |
| 14:22.06 | ManxPower | theH: for how long does it ring? |
| 14:22.18 | ManxPower | number of seconds, be specific, time it. |
| 14:22.22 | Netsnipe | russellb: so I need to pull format_sln16.so from svn? |
| 14:22.25 | TheH | TheH: Those unspecified sips are for our remote offices as a failover solution. We are talking about a 6 server system spread over 5 countries with +500 users in total |
| 14:22.41 | russellb | Netsnipe: i don't know, i was just answering that specific question |
| 14:22.46 | Netsnipe | russellb: cause it's certainly not in menuselect |
| 14:22.46 | TheH | Manx: it rings 1 second and then dies |
| 14:22.53 | ManxPower | theH: could it be about 20 seconds |
| 14:23.01 | Netsnipe | russellb: thanks for the pointer |
| 14:23.11 | TheH | Manx: 1second max. |
| 14:23.14 | Netsnipe | I guess I'll just stick with the default .gsm files |
| 14:23.46 | ManxPower | theH: put Noop(HANGUPCAUSE is ${HANGUPCAUSE}) as the priority after the Dial. |
| 14:23.53 | ManxPower | I'm sorry, but I must now leave. |
| 14:24.06 | ManxPower | you should find a way to provide the requested information faster |
| 14:24.13 | *** join/#asterisk PepOSX (n=angeldav@190.72.151.25) |
| 14:24.22 | TheH | Manx: no problem thanks for your help , let the chat know if i found a solutin |
| 14:25.26 | Netsnipe | russellb: yeah, looks like sln16 support is a v1.6 feature only |
| 14:26.02 | russellb | nods |
| 14:26.04 | mvanbaak | it is |
| 14:26.43 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 14:27.06 | afink | MaliutaLap: Here is the relevant part. http://pastebin.com/m747d153e |
| 14:28.15 | MaliutaLap | afink: none of the includes or extens are valid |
| 14:28.35 | MaliutaLap | afink: you should replace the "=" with "=>" |
| 14:28.42 | MaliutaLap | afink: that might help |
| 14:29.50 | iratik | Anyone know how to find the lib directory for festival? |
| 14:30.13 | MaliutaLap | afink: and you're not attempting to include the trunkdial context anywhere |
| 14:30.33 | MaliutaLap | afink: after that we need to start looking at what context you're dropping things into |
| 14:31.53 | MaliutaLap | iratik: with a debian install you could use dpkg -L, rpm has a similar option |
| 14:31.54 | *** join/#asterisk chigambamukoko (n=junk@71.55.10.211) |
| 14:32.15 | MaliutaLap | iratik: or you could always make use of tools like locate or find |
| 14:32.21 | iratik | too many files for find |
| 14:32.26 | iratik | non debian system doesn't have locate |
| 14:32.30 | iratik | but after some heavily googling |
| 14:32.37 | MaliutaLap | locate is not a debian thing |
| 14:32.42 | iratik | you type festival.. then type libdir |
| 14:32.55 | *** join/#asterisk Dr-Linux|home (n=Nothing@117.20.21.66) |
| 14:32.56 | iratik | MaliutaLap: well... i tried getting locate from yum... nada |
| 14:32.57 | Dr-Linux|home | hey guys |
| 14:33.06 | Dr-Linux|home | please someone look here: http://phpfi.com/330536 |
| 14:33.09 | Dr-Linux|home | my question is on top |
| 14:33.13 | MaliutaLap | iratik: you'd need rpm, not yum |
| 14:33.27 | MaliutaLap | iratik: there's a difference |
| 14:33.39 | Dr-Linux|home | maybe someone dialplan guru give me better way to do that |
| 14:33.49 | MaliutaLap | iratik: and your system should have locate on it, you proably need to run updatdb |
| 14:33.52 | iratik | MaliuataLap: whats the difference... i've always thought that rpm~=deb ... and yum~=apt-get |
| 14:34.08 | d-k-t | iratik, slocate |
| 14:34.15 | d-k-t | yum install slocate |
| 14:34.35 | iratik | MaliuataLap: no.. really ... no locate, no slocate ... found slocate in yum though |
| 14:34.42 | iratik | i would have never known to search under slocate though |
| 14:34.48 | MaliutaLap | iratik: rpm installs the packages and has the options for telling you what is installed and what is in the installed packages |
| 14:34.58 | iratik | like dpkg right? |
| 14:35.22 | iratik | got locate working |
| 14:35.32 | MaliutaLap | iratik: rpm is equiv of dpkg, and yum is the equiv of apt-get. apt-get calls dpkg to do the individual package installs |
| 14:35.48 | MaliutaLap | iratik: you need to read the rpm man page |
| 14:36.36 | MaliutaLap | iratik: rpm can tell you what files are installed by a given package, although I prefer debian systems I still do admin rpm based systems too |
| 14:37.00 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:37.04 | iratik | MaliuataLap: thats only if you didn't install from source though |
| 14:37.13 | iratik | thats where locate can be useful |
| 14:37.21 | Dr-Linux|home | iratik: any suggestion for my questoin? |
| 14:37.25 | d-k-t | iratik, d'oh |
| 14:37.31 | MaliutaLap | iratik: or find, if you know how to use it properly |
| 14:37.54 | iratik | Dr-Linux: I am not the pro with dialplans... I make myne in ruby using adhearsion now adays |
| 14:38.19 | Dr-Linux|home | ok |
| 14:38.23 | iratik | d-k-t: thanks ... MaliuataLap: thanks |
| 14:38.29 | *** join/#asterisk sakajawebe (n=chazz@216.207.245.1) |
| 14:38.33 | MaliutaLap | iratik: and if you feel the need to build from source you should roll your own package, otherwise what's the point of using a package managed system? |
| 14:38.55 | iratik | MaliuataLap: I've never considered making my own package |
| 14:39.15 | Netsnipe | package maintenance is a black-art half the time |
| 14:39.20 | iratik | considers asterisk dialplan language a brutal form of S&M |
| 14:39.25 | Netsnipe | when upstream uses brain-dead hacked compile systems |
| 14:39.45 | Netsnipe | I should know...I used to be a Debian maintainer |
| 14:40.17 | MaliutaLap | Netsnipe: maintaining packages for a distro, yes. Maintaining for a single organisation, not so much |
| 14:40.51 | Netsnipe | if you're going to package something, you have to do it right the first time |
| 14:41.04 | Netsnipe | otherwise you're going to just shoot yourself in the foot on upgrades |
| 14:41.11 | MaliutaLap | iratik: so how do you replicate installs of your hand compiled software over multiple machines |
| 14:41.26 | iratik | MaliuataLap: image and copy |
| 14:41.36 | tzafrir_laptop | MaliutaLap, dpkg -L and dpkg -S can be of use to you, BTW |
| 14:41.45 | MaliutaLap | Netsnipe: or get the package right before distributing it to production sytstems |
| 14:41.53 | MaliutaLap | tzafrir_laptop: I know that |
| 14:42.20 | *** join/#asterisk defswork (n=andy@mx1.3gcomms.co.uk) |
| 14:42.21 | MaliutaLap | tzafrir_laptop: I'm not stupid |
| 14:42.28 | Netsnipe | "touchtone1: If you're calling from a rotary dial phone, hangup, go to a phone store, and purchase one of those new fangled inventions called a touch tone phone. |
| 14:42.29 | Netsnipe | touchtone2: And if you're calling from a rotary dial phone, hello, there's this thing call touch tone sweety you might want to look into. |
| 14:42.29 | Netsnipe | touchtone3: If you're calling from a rotary dial phone, hangup, go to a phone store, and purchase one of those 21'st century new fangled inventions called a touch tone phone. |
| 14:42.29 | Netsnipe | " |
| 14:42.32 | iratik | MaliuataLap: But I hear you.. just haven't considered it or really considered that it would be possible to create my own local repository with my own local packages .... wondering how difficult that is |
| 14:42.38 | Netsnipe | best ever asterisk sound files ever! |
| 14:43.14 | MaliutaLap | iratik: not very |
| 14:43.27 | tzafrir_laptop | iratik, I use reprepro |
| 14:43.43 | MaliutaLap | iratik: I have created both debian and RHEL repos |
| 14:43.57 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:44.05 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 14:44.13 | Dr-Linux|home | tzafrir: any clue |
| 14:44.40 | Dr-Linux|home | http://phpfi.com/330536 << my question is on top of pastebin |
| 14:44.47 | tzafrir_laptop | iratik, the packages in http://updates.xorcom.com/rapid are mostly auto-built from pkg-voip SVN using a hacked up build daemon. I can send you my script |
| 14:44.49 | MaliutaLap | Dr-Linux|home: stick it in a perl agi |
| 14:46.09 | Dr-Linux|home | MaliutaLap: my problem is that how can send caller back to uper priority i.e. label2 |
| 14:46.22 | Netsnipe | iratik: if you're going to build your own packages, install and read the docs in the developers-reference and debian-policy packages |
| 14:47.07 | Netsnipe | debian's cdbs is also a nice build system, but your sources must be using autoconf/automake for it to be of much yes |
| 14:47.30 | Netsnipe | s/yes/use/ |
| 14:47.54 | anonymouz666 | anybody have a Polycom around that can test if exten => XXX,n,SIPAddHeader(Alert-Info: ringType=5) changes the ringtype? |
| 14:48.22 | MaliutaLap | Netsnipe: he did say it wasn't a debian system |
| 14:48.45 | [TK]D-Fender | Dr-Linux|home: OMG, that is horrible... |
| 14:49.01 | anonymouz666 | [TK]D-Fender: you? as a polycom lover... :D |
| 14:49.16 | TheH | 666: exten => i,1,SIPAddHeader(Alert-Info:http://127.0.0.1\;info=External) |
| 14:49.16 | TheH | exten => i,2,Goto(default,4722,1) |
| 14:49.38 | Dr-Linux|home | [TK]D-Fender: any solution for that? |
| 14:49.46 | [TK]D-Fender | anonymouz666: you call the class, not the # |
| 14:50.00 | [TK]D-Fender | anonymouz666: (Alert-Info: Ring-Answer) |
| 14:50.18 | [TK]D-Fender | anonymouz666: You sepcify the class, and your sip.cfg matches and uses whats set for that class |
| 14:51.49 | [TK]D-Fender | Dr-Linux|home: NEVER run IVR's off anything other than "s", and NEVER with a PATTERN. it will call itself RECURSIVELY. |
| 14:51.58 | anonymouz666 | nevermind... I don't even know where is the sip.cfg file... I just have a remote WEB Polycom interface on my screen.... guess I should read more about the phone.... |
| 14:52.27 | iratik | Where can I get help with festival? |
| 14:52.28 | [TK]D-Fender | anonymouz666: Anyone configuring those via web interface should be dragged out and #%$ing shot. |
| 14:53.28 | [TK]D-Fender | Dr-Linux|home: And you never initialize TRIES either. Do you havy any programming experience at all? |
| 14:54.28 | Dr-Linux|home | [TK]D-Fender: ofcos i'm running IVR with "s" but i'm here i'm using patterns to get input |
| 14:55.13 | Dr-Linux|home | [TK]D-Fender: right now all is working for me, but I can't send the caller back to pattern priority |
| 14:55.18 | [TK]D-Fender | Dr-Linux|home: Yease except you are having it call ITSELF, never initializing your TRIES counter, and you can't match "3" or "4" |
| 14:55.53 | [TK]D-Fender | Dr-Linux|home: and NO, youa ren't running that off "s", you're runningit off a nasty super-wildcard exten |
| 14:56.23 | [TK]D-Fender | Dr-Linux|home: And, have no means of dealing with invalid input. |
| 14:56.30 | *** join/#asterisk Peri (n=redanti@66.185.87.56) |
| 14:56.38 | [TK]D-Fender | Dr-Linux|home: Time to go right back to the drawing board on this one. |
| 14:56.38 | Segnale007 | hello .. I have a question |
| 14:56.44 | [TK]D-Fender | ~ask |
| 14:56.44 | jbot | ask is, like, Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:57.04 | Segnale007 | I am an newbie and I am going to buy an tdm413b |
| 14:57.15 | [TK]D-Fender | Segnale007: Whats the FXS for? |
| 14:57.15 | Segnale007 | does it work on sparc64 arch ? |
| 14:57.32 | Segnale007 | for switch my pstn |
| 14:57.39 | [TK]D-Fender | Segnale007: I seriously doubt it. Call Digium direct for that info. |
| 14:57.46 | Segnale007 | ok |
| 14:57.47 | [TK]D-Fender | Segnale007: "huh? |
| 14:57.49 | Segnale007 | ? |
| 14:58.03 | Segnale007 | I am still studying |
| 14:58.09 | [TK]D-Fender | Segnale007: "for switch my pstn" <- I have no idea what you're saying |
| 14:58.09 | Segnale007 | I need it for try . |
| 14:58.23 | Segnale007 | ok I expain to you what I want to do |
| 14:58.31 | Dr-Linux|home | [TK]D-Fender: there could be many other mistakes, but my question is can i send back the caller to uper pattern priority? |
| 14:58.47 | Segnale007 | analog line > server pbx > pstn phone |
| 14:59.11 | Segnale007 | analog line to fxs > fxo server pbx > fxo pstn phone |
| 14:59.13 | [TK]D-Fender | Dr-Linux|home: Sure. Your Goto is bad. |
| 14:59.14 | Segnale007 | maybe I am wrong |
| 14:59.17 | Segnale007 | probabilu |
| 14:59.28 | Segnale007 | I am still trying to understand |
| 14:59.36 | [TK]D-Fender | Segnale007: So you want to put * between an existing PBX and the telco? |
| 14:59.41 | *** join/#asterisk oilinki (n=oil@ppp-124-120-6-247.revip2.asianet.co.th) |
| 15:00.00 | Segnale007 | no I dont' have any pbx right now |
| 15:00.06 | Segnale007 | I am going to build one |
| 15:00.09 | Dr-Linux|home | [TK]D-Fender: any hint just to send the caller back to pattern priority? |
| 15:00.27 | Segnale007 | thats why I need an analog card |
| 15:00.34 | Segnale007 | with fxs and fxo modules |
| 15:00.42 | Segnale007 | and echo cancellation as well |
| 15:01.34 | Segnale007 | maybe I am wrong about how to build it ? |
| 15:01.37 | Segnale007 | probabily .. |
| 15:01.44 | Segnale007 | I am still reading a lots of stuff .. |
| 15:01.55 | [TK]D-Fender | Dr-Linux|home: You are using the "else" clause to send them back to the label, but that label isn't for the "t" exten <- |
| 15:01.58 | Segnale007 | if somebody can help me to understand would be great |
| 15:02.22 | *** join/#asterisk oilinki (n=oil@ppp-124-120-6-247.revip2.asianet.co.th) |
| 15:02.30 | [TK]D-Fender | Segnale007: If you just want to us an analog phone with *, I would advise using an ATA instead of a zaptel FXS module. |
| 15:02.54 | Segnale007 | why ? |
| 15:03.08 | Segnale007 | is less expensive ? |
| 15:03.11 | Segnale007 | I can do it .. |
| 15:03.23 | [TK]D-Fender | Segnale007: Less expensive, less trouble. |
| 15:03.35 | Segnale007 | but I want to learn as much thing as I can from asterik |
| 15:03.40 | Segnale007 | I seee .. |
| 15:03.57 | [TK]D-Fender | Segnale007: ATAs are typically a much more functional and flexible way to do FXS |
| 15:04.07 | Segnale007 | oh .. |
| 15:04.11 | [TK]D-Fender | Segnale007: and it won't take away from your learning experience with * |
| 15:04.11 | Segnale007 | I see .. |
| 15:04.29 | Segnale007 | well .. thanks for you idea .. I apreciate .. |
| 15:04.32 | [TK]D-Fender | Segnale007: Zaptel FXO (for lines) is a good thing though |
| 15:04.34 | Dr-Linux|home | [TK]D-Fender: and the way to do that? specifically ..... ?t,3:label2) |
| 15:05.03 | Segnale007 | sounds good .. |
| 15:05.14 | [TK]D-Fender | Dr-Linux|home: You just don't seem to get it. you are IN "t" and want to jump to a label for ANOTHER exten. Go read the instructions for Goto again. |
| 15:05.29 | Segnale007 | what kind of analog card you suggest to me ? |
| 15:05.38 | [TK]D-Fender | Segnale007: We advise the Linksys SPA ATA series. |
| 15:05.47 | Segnale007 | good |
| 15:05.55 | [TK]D-Fender | Segnale007: The TDM410P is still a decent bet for your FXO needs. |
| 15:05.56 | Segnale007 | is cheap |
| 15:06.16 | Segnale007 | but its without echo cancellation hw |
| 15:06.19 | [TK]D-Fender | Segnale007: for FXO we generally advise PCI cards, for FXS, ATA's instead |
| 15:06.20 | Segnale007 | right ? |
| 15:06.26 | *** join/#asterisk nicoAMG (i=asgalt@216.25.160.214) |
| 15:06.28 | Segnale007 | oh I see .. |
| 15:06.31 | Segnale007 | nice to know that |
| 15:06.34 | Segnale007 | I appreciated |
| 15:06.38 | Segnale007 | ty |
| 15:06.40 | [TK]D-Fender | Segnale007: You can get HWEC for the TDM410 series (not the 400 series) |
| 15:06.47 | MaliutaLap | if you are going to put in a TMD400P you might aswell get an FXS aswell, |
| 15:07.02 | [TK]D-Fender | Segnale007: How many lines are you using it for? And what kind of use? Home? Small business? |
| 15:07.15 | MaliutaLap | I have a 400P with 1fxo 1fxs |
| 15:07.24 | [TK]D-Fender | Maliuta : module isn't cost effective, less flexible, and more of a PITA for nothing. |
| 15:07.44 | Segnale007 | it would be for my home |
| 15:07.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:07.52 | Segnale007 | and I am planing 3 lines |
| 15:07.57 | Segnale007 | thats all |
| 15:08.03 | Segnale007 | just for start |
| 15:08.08 | MaliutaLap | [TK]D-Fender: how do you figure less flexible? |
| 15:08.20 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:08.29 | [TK]D-Fender | Segnale007: Ok, then look for the TDM410P with HWEC, or a Sangoma A200d |
| 15:08.45 | Segnale007 | nice |
| 15:08.56 | Segnale007 | do u know a good store to buy ? |
| 15:08.56 | MaliutaLap | [TK]D-Fender: I can still drop it into the context of my choosing, the handsets are cordless ... |
| 15:09.04 | Segnale007 | I only know 888voipstore.com |
| 15:09.09 | Segnale007 | they have good price |
| 15:09.16 | [TK]D-Fender | MaliutaLap: You need to wire it direct to your * server which can suck for wiring. A problem on that wiring can fry your card/server. ATA's can have SIP redundency, and are easily relocatable. |
| 15:09.17 | Segnale007 | I think so .. |
| 15:09.30 | MaliutaLap | Segnale007: you will need to buy one that meets the requirements of your national standards |
| 15:09.32 | [TK]D-Fender | MaliutaLap: AND it leaves his card open for expansion. He was about to FILL it. |
| 15:09.53 | Segnale007 | I don't understandnow |
| 15:09.59 | [TK]D-Fender | Segnale007: Shop locally first... import & shipping WILL suck... |
| 15:10.06 | Segnale007 | ah I see |
| 15:10.11 | Segnale007 | thanks |
| 15:10.11 | [TK]D-Fender | Segnale007: and for US pricing, try www.telephonydepot.com |
| 15:10.31 | Segnale007 | ty ;) |
| 15:10.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 15:10.50 | [TK]D-Fender | Segnale007: Feel free to call around to confirm the "landed" price that it will cost you in the end. Take into consideration that if you have a problem and need to RMA something that those factors will come up again. |
| 15:11.13 | Segnale007 | ok .. |
| 15:11.16 | MaliutaLap | Segnale007: if the card you buy from OS isn't passed by you national telecommunications body you could find yourself in all sorts of expensive legal problems |
| 15:11.41 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:11.44 | Segnale007 | I see .. |
| 15:11.53 | Segnale007 | then Ill shope in europe frist |
| 15:11.58 | Segnale007 | *shop |
| 15:12.14 | [TK]D-Fender | Segnale007: And call up Digium & Sangoma to confirm their card's certification in your area. |
| 15:12.34 | MaliutaLap | Segnale007: for example if I don't buy something that has been properly imported and A-Tick tested/approved it could cost me the, probably, million or so dollars to rebuild the exchange I am connected to |
| 15:13.32 | Segnale007 | ohh I see.. |
| 15:13.42 | Segnale007 | Ill call them frist .. |
| 15:14.40 | MaliutaLap | you should be able to find someone in country to buy from, google is your friend |
| 15:14.53 | MaliutaLap | and the wiki should help you along the way |
| 15:15.15 | spokra | I have a digium T1 card i got at a class. any one need one. it's worth i think around $400. static bag has never been opened. |
| 15:15.22 | *** join/#asterisk afink (n=chatzill@72-164-59-242.dia.static.qwest.net) |
| 15:16.29 | MaliutaLap | <MaliutaLap> afink: and you're not attempting to include the trunkdial context anywhere |
| 15:16.30 | MaliutaLap | <MaliutaLap> afink: after that we need to start looking at what context you're dropping things into |
| 15:16.44 | MaliutaLap | afink: and I am about to go to bed |
| 15:16.55 | afink | ok, Thank you for your help |
| 15:17.00 | *** join/#asterisk notjohn (n=notjohn@216.68.73.132) |
| 15:18.20 | notjohn | can someone point me in the right direction for asterisk hardware...whatever the piece is that hooks up to the pc? |
| 15:23.03 | *** join/#asterisk StooJ (n=stooj@johnston37.plus.com) |
| 15:23.40 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:24.00 | [TK]D-Fender | notjohn: there are hundreds of pieces of hardware that can work with *. Perhaps you should be a bit more specific. |
| 15:24.20 | [TK]D-Fender | notjohn: what exactly is it that you need to do? |
| 15:24.35 | notjohn | [TK]D-Fender: i figured so... |
| 15:24.47 | *** join/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-9c60d01c2e51d3f3) |
| 15:25.52 | notjohn | [TK]D-Fender: I'd like to be able to setup asterisk to handle some simple stuff... connect a land line and a voip line or something similar so I can call into my home number from my mobile and then make an outgoing call |
| 15:26.15 | notjohn | so i don't give out my mobile # on caller id |
| 15:26.20 | kensuke_ | hi, i have a question, i need install the g729 codec on x86_64 xeon |
| 15:26.21 | notjohn | that's one thing |
| 15:26.46 | [TK]D-Fender | notjohn: basic home use? |
| 15:26.54 | kensuke_ | what binaries use? core2.so? |
| 15:27.05 | kensuke_ | http://asterisk.hosting.lv/#bin |
| 15:27.13 | [TK]D-Fender | kensuke_: If its a core2, sure. Go try |
| 15:27.37 | notjohn | [TK]D-Fender: yes...very basic, I remember looking into asterisk a year ago but never set it up... I just can't remember what the hardware piece I looked at was callled |
| 15:27.39 | kensuke_ | [TK]D-Fender: thanks |
| 15:27.50 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:28.05 | [TK]D-Fender | notjohn: I'd suggest a Linksys SPA-3102 for you then. |
| 15:28.32 | tzanger | how do you pronounce Jeffrey Bezos? bee-zose? bezose? |
| 15:30.46 | lmadsen | bee-zoes is how I would say it |
| 15:31.29 | *** join/#asterisk ACiDV (n=acidv@24-230-219-82.dr.cgocable.ca) |
| 15:32.28 | [TK]D-Fender | tzanger: I'm bet : bee'z-oh's |
| 15:35.13 | *** part/#asterisk ipso (n=ipso@S0106005004c32d38.ok.shawcable.net) |
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| 15:37.01 | notjohn | [TK]D-Fender: thanks... with that Linksys work with say Gizmo and Grand Central? |
| 15:37.23 | [TK]D-Fender | notjohn: So far I was mentioning it just for FXO (access to your LINE). |
| 15:37.26 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 15:37.44 | [TK]D-Fender | notjohn: Connecting to ITSP's should *'s job |
| 15:38.17 | notjohn | [TK]D-Fender: gotcha |
| 15:38.27 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
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| 15:52.20 | hsv-al | oh well, having issues just trying to get simple voicemail working, with book examples, mail isnt even being delivered |
| 15:52.22 | hsv-al | will try later |
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| 15:53.31 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:54.26 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:56.31 | hsv-al | just trying to setup a * box at my apartment, so when people call in to landline pbx, goes to voicemail prompt after lets say 30 seconds......leaves a voicemail, and boom its stored...also sending the clip to 2 different email addresses |
| 15:56.40 | hsv-al | following book steps, i get all sorts of insane errors in the console oh well! |
| 15:57.16 | CanWood | I know I'll get in trouble for mentioing it here, but have you considered freepbx? |
| 15:57.39 | hsv-al | wont use it, ive been using this manually for a few months now, but now im trying to get voicemail working |
| 15:57.44 | hsv-al | its just a matter of reading more, and experimenting |
| 15:57.49 | hsv-al | but just a tad frustrating now |
| 15:58.03 | hsv-al | i have 8 phones linked up through my * box , via 8 diff states |
| 15:58.11 | CanWood | I'm going the other way. I started with freepbx and am now working on writing manual dial plans |
| 15:58.27 | hsv-al | im charging like 7 people 200/month so they can have laptop to laptop communication |
| 15:58.35 | hsv-al | in their business, its running off my * box at home heh |
| 15:58.56 | hsv-al | they dont want voicemail, just soft clients for insurance agents to speak to each other while their on the road :) |
| 15:59.42 | hsv-al | but im trying to experiment getting voicemail working for em |
| 15:59.44 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 16:00.18 | hsv-al | when voicemail is running correctly, and the email address is specified in mb => pass,name[,email[,pager_email[,options]]] |
| 16:00.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:00.31 | hsv-al | all of the background smtp communication stuff is taken care of out of the box i assume? |
| 16:00.48 | [TK]D-Fender | hsv-al: You need sendmail installed and functional. |
| 16:01.06 | outtolunc | never got a box.. damn |
| 16:01.08 | hsv-al | well, it was written as if it auto sends |
| 16:01.13 | [TK]D-Fender | hsv-al: Or an equivalent with sendmail compatibility layer installed |
| 16:01.15 | hsv-al | ill have to get it configured then later |
| 16:01.25 | hsv-al | book doesnt mention that being necesary |
| 16:01.28 | hsv-al | not an issue, but whatever |
| 16:01.35 | [TK]D-Fender | hsv-al: Most common distro's will work "out of the box" |
| 16:01.45 | hsv-al | im using 1.4 on 8.04 |
| 16:02.06 | [TK]D-Fender | hsv-al: and you're also showing us NOTHING. |
| 16:02.17 | hsv-al | :) my conf files are basic no bloat |
| 16:02.20 | hsv-al | ill pastebin links later |
| 16:03.21 | *** part/#asterisk chigambamukoko (n=junk@71.55.10.211) |
| 16:12.07 | CVirus | I think I found a mistake in the ATFOT book ... where do I report this ? |
| 16:12.41 | CVirus | page 125 ... the note at the bottom .. it says zaptel.conf instead of zapata.conf ... I think that's a mistake |
| 16:13.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:14.57 | CVirus | hopes that [TK]D-Fender doesn't kick his ass as he usually does |
| 16:17.17 | [TK]D-Fender | CVirus: I don't see it.. pastebin the excerpt (a big chunk please) |
| 16:18.40 | *** join/#asterisk gcarrillog (n=gcarrill@201.151.84.209) |
| 16:18.42 | gcarrillog | hi |
| 16:18.47 | CVirus | [TK]D-Fender: http://rafb.net/p/ebNSmo18.html |
| 16:18.59 | gcarrillog | someone use spa400 with asterisk? |
| 16:19.53 | [TK]D-Fender | CVirus: `indeed, that should have been "zapata.conf" |
| 16:20.04 | [TK]D-Fender | gcarrillog: Yes, I'm sure there are. |
| 16:20.13 | CVirus | I'm glad |
| 16:20.37 | [TK]D-Fender | CVirus: ask lmadsen where to submit comments like this when he's about next. |
| 16:20.59 | gcarrillog | i have the next message Forbidden - wrong password on authentication for REGISTER for 'spa400' |
| 16:21.10 | gcarrillog | but spa400 dont use passwd |
| 16:21.54 | CVirus | [TK]D-Fender: cool .. thanks |
| 16:22.10 | gcarrillog | i havent had that problem before |
| 16:22.16 | [TK]D-Fender | gcarrillog: www.voxilla.com , www.voip-info.org <- go read the forums & wiki on how to set this device up |
| 16:23.30 | gcarrillog | thanks but, last week spa400 had working perfectly |
| 16:23.55 | [TK]D-Fender | gcarrillog: Then somethign changed. |
| 16:24.42 | notjohn | you can install the asterisk gui on a normal asterisk installation, right? |
| 16:26.10 | [TK]D-Fender | notjohn: Yes |
| 16:28.23 | drako | needs a spa400 |
| 16:28.41 | notjohn | is there any preferred distro? what asterisk now is packaged with?? |
| 16:28.48 | [TK]D-Fender | drako: it has "quirks".... not on my list.. |
| 16:29.02 | [TK]D-Fender | notjohn: Pick a common one you are comfortable administering. |
| 16:29.40 | mmlj4 | I've been asked to put * in front of an existing PBX, and "bridge" (stealing an IP metaphor) the incoming T1 channels to the old PBX, all except for a couple of individual channels... this can be done, right? |
| 16:30.14 | [TK]D-Fender | mmlj4: Sure |
| 16:30.18 | mmlj4 | notjohn: Digium recommends debian and redhate/fedora |
| 16:30.44 | mmlj4 | notjohn: but any server-class distro will work |
| 16:30.53 | [TK]D-Fender | notjohn: CentOS is probably one of the best choices given its user base. |
| 16:31.17 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 16:31.21 | mmlj4 | and centos basically equals RHEL |
| 16:31.45 | [TK]D-Fender | yup |
| 16:31.55 | notjohn | i use cent most of the time but wasn't sure if need something so beefy for a home project |
| 16:32.12 | [TK]D-Fender | notjohn: I use it myself... |
| 16:32.48 | *** join/#asterisk pikachu2000 (n=pikachu2@196.209.182.220) |
| 16:33.14 | notjohn | sometimes i like to play it a bit more bleeding edge when i can get away with it :) |
| 16:33.47 | notjohn | but then it's hard to keep up with fedora |
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| 16:35.56 | *** part/#asterisk gcarrillog (n=gcarrill@201.151.84.209) |
| 16:36.16 | drako | [TK]D-Fender, a good alternative around the same price? |
| 16:39.12 | *** join/#asterisk StooJ (n=stooj@johnston37.plus.com) |
| 16:42.40 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 16:43.21 | hsv-al | <PROTECTED> |
| 16:43.34 | Qwell | mmm, Taco Bell |
| 16:43.35 | Qwell | good idea! |
| 16:43.38 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 16:43.38 | Qwell | thanks |
| 16:43.54 | hsv-al | i actually had TB last night, its like crack |
| 16:44.00 | hsv-al | fast food every 3 months is like doing meth for the first time |
| 16:44.19 | Qwell | I'll agree with you on Hardee's though. Disgusting. |
| 16:44.35 | Qwell | Carl's Jr. <3 |
| 16:45.00 | hsv-al | This Pom brand juice is crazy expensive |
| 16:45.06 | hsv-al | its $3.99 for 16oz |
| 16:45.21 | Qwell | good though.. |
| 16:45.55 | Kobaz | where can i get a fairly cheap rj22 headset |
| 16:46.07 | hsv-al | qwell you all got it made there |
| 16:46.14 | hsv-al | just take the elevator downstairs, and boom a coffee shop |
| 16:46.23 | Qwell | sips his mocha and nods |
| 16:47.15 | russellb | mocha != free ... coffee + packet of hot chocolate = free |
| 16:47.26 | hsv-al | put in a request to HR |
| 16:47.34 | hsv-al | that all Dig employees eat/drink at aromas for free ftw |
| 16:47.36 | russellb | i take the free route most days |
| 16:49.14 | *** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167059025.pppoe-dynamic.nb.aliant.net) |
| 16:49.16 | hsv-al | dont you guys get free weekly meetings at laredo at BS once a month? |
| 16:49.22 | hsv-al | lunches rather? |
| 16:49.25 | Qwell | eh? |
| 16:49.34 | hsv-al | someone said something about that, maybe they were bs'ing |
| 16:49.35 | hsv-al | cantina |
| 16:49.44 | hsv-al | mexican , bridge street |
| 16:49.50 | russellb | no .. |
| 16:50.13 | hsv-al | nothing like green paste there mmmm |
| 16:50.19 | hsv-al | ;() |
| 16:50.38 | russellb | let's try to keep the channel on topic, please :) |
| 16:50.51 | *** join/#asterisk Yourname` (i=chatzill@unaffiliated/yourname/x-837320) |
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| 16:59.01 | [TK]D-Fender | drako: What do you need? |
| 16:59.23 | drako | [TK]D-Fender, FXO |
| 16:59.36 | drako | around 2 or 4 ports |
| 17:00.45 | lmadsen | CVirus: report to errata@oreilly.com |
| 17:00.48 | [TK]D-Fender | Audiocodes is acceptably prices, but I'd rather pay for a PCI FXO solution most of the time. |
| 17:01.58 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 17:02.10 | CVirus | lmadsen: thanks |
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| 17:14.57 | RoyK | lmadsen: hei. trodde du hadde gitt opp asterisk :P |
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| 17:21.49 | *** part/#asterisk Xamusk (n=Xamusk@189.1.136.223) |
| 17:27.52 | jaytee | I'm using Polycom IP330 phones and I haven't programmed anything for Transfer or Forward in features.conf file. I can transfer and forward calls fine using the softkeys on the Polycom. How do I transferring or forwarding a call to an outside number for a specific extension? |
| 17:28.26 | jaytee | oops, I meant how do I BLOCK transferring or forwarding for a specific extension |
| 17:29.40 | thomas | is it posible to show the traffic from iax peer X ? |
| 17:30.35 | [TK]D-Fender | jaytee: Any transfer is jsut a transfer |
| 17:30.47 | [TK]D-Fender | jaytee: There is no such thing as "internal" or "external" |
| 17:31.38 | [TK]D-Fender | jaytee: if you want to do selective auto-divert for certain people, add them to your Directory, and enable the auto-divert on them. |
| 17:31.55 | [TK]D-Fender | thomas: "sip debug peer [peername]" |
| 17:32.07 | thomas | [TK]D-Fender: also for iax, yes? |
| 17:32.17 | [TK]D-Fender | jaytee: And you should never need or use features.conf if you're using phones like that. |
| 17:32.20 | [TK]D-Fender | thomas: yes |
| 17:32.40 | thomas | [TK]D-Fender: ah, ok. thank you very much. |
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| 17:36.46 | *** join/#asterisk nightkhaos (i=4e566f7e@gateway/web/ajax/mibbit.com/x-2992fc09c1376c4c) |
| 17:37.07 | nightkhaos | Alright... I've got zaptel working and asterisk installed... now what? :) |
| 17:37.38 | lanning | Profit! |
| 17:37.52 | nightkhaos | In theory. |
| 17:38.04 | jaytee | [TK]D-Fender, I haven't used features.conf since the Polycoms have transfer and forward functions built-in. |
| 17:38.18 | [TK]D-Fender | nightkhaos: ... |
| 17:38.19 | [TK]D-Fender | ~nowwhat |
| 17:38.20 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 17:38.34 | [TK]D-Fender | jaytee: Good. |
| 17:38.44 | nightkhaos | My question is more... where the hell do I go to get a guide on configuring? I would rather NOT go through 101 man pages and try and get my head around configuration files. |
| 17:38.44 | [TK]D-Fender | jaytee: only point for it now is for recording. |
| 17:39.10 | [TK]D-Fender | nightkhaos: Guess what.. thats what * uses. You need to focus on your dialplan. Thats the most important thing |
| 17:39.15 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:39.25 | Peri | Anyone here ever worked with boardworks AS and SBCs? |
| 17:39.33 | [TK]D-Fender | nightkhaos: Device setup is stiny. Dialplan is everything. |
| 17:40.13 | jaytee | [TK]D-Fender, yeah that's what I was looking at for earlier but I haven't needed it. I'm just trying to figure out how to setup classes of users where some can call forward or transfer to a 7 digit external number and others can't but I want everyone or most everyone to be able to call 7 digit or 10 digit numbers. |
| 17:40.44 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 17:40.46 | nightkhaos | [TK]D-Fender: Okay... assume my zaptel service is working fine... and I'm a newbie... and I have absolutely no idea where to start... where would you send me? Any tutorials? |
| 17:41.04 | [TK]D-Fender | nightkhaos:... |
| 17:41.06 | [TK]D-Fender | ~book |
| 17:41.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:41.06 | Peri | www.asteriskguru.com |
| 17:41.11 | [TK]D-Fender | nightkhaos: Chapter 5 |
| 17:41.18 | Peri | oh yes |
| 17:41.20 | Peri | the book |
| 17:41.24 | [TK]D-Fender | nightkhaos: For some "inspiration" : |
| 17:41.26 | [TK]D-Fender | ~jerjerguide |
| 17:41.27 | 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/ |
| 17:42.14 | [TK]D-Fender | nightkhaos: JerJer's guide is pretty minimalistic but will show you usage of some contexts, "includes", etc. |
| 17:43.33 | nightkhaos | [TK]D-Fender: alright... mind resending me those URLs in a minute, I'm gonna switch to my Lappy. |
| 17:43.41 | Kobaz | [TK]D-Fender: i'm looking at the queue docs because i'm finnaly fixing my stuff to not use AgentRingbackLogin... is the new AEL dialplan language usable in 1.4 ? |
| 17:43.47 | nightkhaos | brb |
| 17:44.35 | [TK]D-Fender | Kobaz: Yes, though I've never seena need for it. |
| 17:44.51 | Kobaz | [TK]D-Fender: it's much more straightforward than the usual syntax |
| 17:45.13 | Kobaz | [TK]D-Fender: it's a real language almost, rather than coding in BASIC |
| 17:45.24 | [TK]D-Fender | Kobaz: Poorer documentation, much fewer quality people to assist. Your call. |
| 17:45.50 | Kobaz | but it's probably going to be the preferred method in 1.6... wouldn't it? |
| 17:46.14 | [TK]D-Fender | Kobaz: LOL. No. |
| 17:46.19 | Kobaz | heh |
| 17:46.26 | [TK]D-Fender | Kobaz: You don't quite seem to understand what AEL really is. |
| 17:46.39 | Kobaz | a more structured approach to the dialplan |
| 17:46.53 | [TK]D-Fender | Kobaz: thats a partial answer missing crucial bits. |
| 17:47.09 | Kobaz | it's also more easily maintainable |
| 17:47.21 | Kobaz | with labels, and switches's and clear conditionals |
| 17:47.39 | ManxPower | Kobaz: AEL is a "language" for dialplan stuff. It is converted to regular extensions.conf format when the AEL file is loaded. |
| 17:47.41 | [TK]D-Fender | Kobaz: Its a parser that generaltes dialplan to the best of its ability. It is therefor less capable and is subject to "issues" that straight extensions.conf is not. |
| 17:47.51 | ManxPower | There is *NOTHING* you can do in AEL that you cannot do in the dialplan |
| 17:48.16 | [TK]D-Fender | Kobaz: So in the end you introduce points of failure (bugs and limitations). |
| 17:48.34 | ManxPower | I've heard good things about AEL in 1.4 |
| 17:48.38 | ManxPower | in 1.2, it sucked |
| 17:48.48 | [TK]D-Fender | ManxPower: And conversely quite possibly things in the dialplan you CAN'T do in AEL. |
| 17:49.00 | Kobaz | [TK]D-Fender: assuming the bugs are fixed up though, it looks to be a better approach to setting a dialplan |
| 17:49.03 | ManxPower | [TK]D-Fender: *nod*. I like the idea of AEL. |
| 17:49.12 | ManxPower | Kobaz: It is NOT better, just different |
| 17:49.33 | CanWood | Hey folks. In a dial plan, I have a Playback() followed by a Read() and am looking for the Playback to be able to be interrupted by the entering of digits. BackgroundDetect() doesn't seem to quite do the job, as it takes what's entered and tries to transfer me to that extension. I just want to read a variable. Any other suggestions? |
| 17:49.36 | [TK]D-Fender | ManxPower: "means well". The actual problem is *extensions.conf* is the base. THAT should be killed off and completely reconceived |
| 17:50.03 | [TK]D-Fender | CanWood: Read can playbacka file all by itself. |
| 17:50.06 | unpaidbill | use read to play back the file |
| 17:50.08 | unpaidbill | arrr |
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| 17:50.57 | CanWood | would ya look at that! tks [TK]D-Fender. I missed that |
| 17:51.28 | ManxPower | CanWood: : does "core show application read" say you can play a file. |
| 17:53.19 | *** join/#asterisk NightKhaos (n=nightkha@78-86-111-126.zone2.bethere.co.uk) |
| 17:53.23 | NightKhaos | right |
| 17:53.24 | NightKhaos | back |
| 17:54.06 | NightKhaos | [TK]D-Fender; what were those URLs again? :) |
| 17:54.09 | CanWood | ManxPower: yep, as [TK]D-Fender pointed out that's exacty what I need. I'm trying to read all docs before asking here but I missed that bit. My bad |
| 17:54.29 | ManxPower | CanWood: there is an entire directory of docs included in Asterisk |
| 17:54.31 | [TK]D-Fender | ~book |
| 17:54.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:54.33 | ManxPower | there is also the CLI docs |
| 17:54.34 | [TK]D-Fender | ~jerjerguide |
| 17:54.35 | 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/ |
| 17:55.08 | CanWood | I have the book open in fornt of me and missed the <i>filename</i>. part. Again. My bad, my apologies |
| 17:55.56 | ManxPower | CanWood: a single mistake in setting up a phone system could allow outside callers to use your system to make long distance calls billed to your line. You should be sure to be careful. |
| 17:56.04 | NightKhaos | Now is it worth using a GUI front end? |
| 17:56.27 | ManxPower | NightKhaos: not for anyone here or they would be on the channels for Asterisk GUIs |
| 17:56.30 | CanWood | I agree, I should be, and am trying to be, and appreciate the warning |
| 17:56.51 | outtolunc | hang him! |
| 17:57.13 | NightKhaos | ManxPower: a simple "no" would surfice. :) |
| 17:57.16 | unpaidbill | manxpower it would have to be quite a mistake, like transferring someone into your local contexts though... would it not? |
| 17:57.34 | ManxPower | unpaidbill: no. Just allowing T and t on the Dial line could do it. |
| 17:57.37 | unpaidbill | err -though. |
| 17:57.40 | unpaidbill | oh good point |
| 17:57.48 | ManxPower | or being in and IVR that would let you dial out. |
| 17:57.56 | ManxPower | because of a mistaken include => |
| 17:58.00 | jaytee | [TK]D-Fender, I'm looking through the Polycom SIP 3.0 Administrator's Guide but there's only one page where it references auto-divert. I can see where I can set a divert contact number but is there another manual I should reference to get a better understanding of how that works? |
| 17:58.44 | [TK]D-Fender | jaytee: its a 502 redirect like any other |
| 17:58.52 | ManxPower | T/t could allow anyone to transfer themselves to an external number just by dialing *number |
| 17:58.54 | [TK]D-Fender | jaytee: nothing more to say about it |
| 17:59.06 | ManxPower | wile ON a call |
| 17:59.12 | unpaidbill | well, T could |
| 17:59.22 | ManxPower | unpaidbill: depending on the direction of the dial |
| 17:59.29 | unpaidbill | yeah |
| 17:59.51 | [TK]D-Fender | NightKhaos: No. Just go learn the dialplan. Its worth it. |
| 17:59.54 | ManxPower | imagine if you called someone outside and had the wrong T/t on the dial line? |
| 18:00.44 | unpaidbill | that would be fun |
| 18:00.44 | NightKhaos | [TK]D-Fender: okay, final questions... 1.4 vs 1.2.27... advantages? |
| 18:00.44 | ManxPower | unpaid I suspect there are HUNDREDS of systems setup in a way that would allow that |
| 18:00.44 | unpaidbill | hrm |
| 18:00.44 | unpaidbill | time to get out the old wardialer |
| 18:00.44 | unpaidbill | :P |
| 18:00.49 | [TK]D-Fender | NightKhaos: 1.4, latest. 1.4 has more features, and is SUPPORTED |
| 18:00.53 | ManxPower | once you get your first $10,000 phone bill people tend to fix that |
| 18:00.57 | unpaidbill | yeah that's a good point though, i guess it is pretty easy to make the tT mistake |
| 18:01.13 | ManxPower | unpaidbill: easy to make the include => mistake too |
| 18:01.27 | NightKhaos | Tt? |
| 18:01.31 | [TK]D-Fender | unpaidbill: All mistakes are easy. Its just a question of the severity of the consequences. |
| 18:02.13 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
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| 18:02.33 | [TK]D-Fender | NightKhaos: When using Dial, you can permit one end or BOTh to transfer the call. Depending on your goof and what context they have access to they can dial who-knows-what, at your expense. |
| 18:03.53 | NightKhaos | [TK]D-Fender: I guess safe guarding against this is covered in the book? :) |
| 18:04.25 | [TK]D-Fender | NightKhaos: read your applications INSTRUCTIONS when you build your dialplan. |
| 18:05.27 | NightKhaos | [TK]D-Fender: is that a yes? lol |
| 18:05.40 | [TK]D-Fender | NightKhaos: And no... there shouldn't be anything on this really. The function does what it says. If you can't read the instructions and realize (Oh you mean THEY can transfer themselves anywhere in this context?!) this, then you get what you deserve. |
| 18:06.28 | outtolunc | actually it is anywhere in that context (or the TRANSFER_CONTEXT if set)( |
| 18:06.36 | [TK]D-Fender | NightKhaos: Guns shouldn't come with a safety label saying "don't shoot yourself", or "not suitable for cleaning cars". If you can't realize this, you shouldn't own one. |
| 18:07.06 | NightKhaos | [TK]D-Fender: and yet... people do. Your faith in humanity is strangly misplaced, but I do conceed to your point. :) |
| 18:08.28 | unpaidbill | let evolution sort it out. |
| 18:09.41 | NightKhaos | unpaidbill: saddly evolution has created mechinisms to counter evolution. In our society these developments come as doctors and medical professionals. |
| 18:13.35 | lanning | and we cherish the underdog |
| 18:16.32 | *** join/#asterisk mmattice (i=mmattice@unaffiliated/mmattice) |
| 18:17.39 | mmattice | what normal ubuntu progs can play * wav files? |
| 18:19.35 | *** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 18:19.49 | *** join/#asterisk mikkel (n=mikkel@eye.exoro.dk) |
| 18:21.32 | Nugget | there's no such thing as a "normal" unix program. :) |
| 18:21.50 | Qwell | sh |
| 18:24.17 | mmattice | apparently sox's play can't handle them |
| 18:24.35 | jblack | mmatice: play, xmms, probably xine |
| 18:24.42 | Qwell | play = sox |
| 18:24.46 | Qwell | and yes, sox can handle wav |
| 18:25.14 | mmattice | it can't seem to handle the gsm wav files |
| 18:25.28 | jblack | sox can handle can do gsm with the gsm plugin. |
| 18:25.40 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 18:25.42 | jblack | That's what I use to convert wav to gsm, since so few things can work with it. |
| 18:25.46 | jblack | (I wish audacity did!) |
| 18:25.54 | Qwell | audacity can |
| 18:26.10 | bobbym | someone could please recommend an softphone (g729 support) to macosx ? |
| 18:26.22 | bobbym | i'm using the xlite demo version but i want to have something better... |
| 18:26.35 | jblack | qwell: That's great news! |
| 18:27.48 | HonestWorker | I have got to go |
| 18:27.50 | HonestWorker | Bye bye |
| 18:27.54 | *** part/#asterisk HonestWorker (n=Wothanaz@201.87.225.101) |
| 18:35.24 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:35.51 | mikkel | I have a Sangoma 500 BRI card, I have ingoing calls working, but my local connected phone is not. Anyone has some experience with that ? |
| 18:36.16 | thomas | i can recieve Fax only with 9600 - why not 14400 ? |
| 18:36.28 | mikkel | I have set the card to NT and connected power to the card. |
| 18:36.38 | thomas | fax > isdncard > asterisk > iaxmodem > hylafax |
| 18:36.50 | ManxPower | bobbym: there will be NO free softphone with G729, only a pay one |
| 18:38.49 | bobbym | ManxPower: eyebeam is a good one? |
| 18:39.17 | *** join/#asterisk legend1222_ (n=legend@66.178.252.218) |
| 18:39.52 | ManxPower | bobbym: I would never ever use a softphone. |
| 18:40.55 | legend1222_ | Is there really that big of a difference between asterisk and asterisk-now that it needs its own support channel? |
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| 18:41.17 | ManxPower | legend1222_: yes |
| 18:41.44 | ManxPower | the Asterisk config files have to be pretty complex for a GUI to manage them |
| 18:41.44 | legend1222_ | Can you elaborate at all? (other then the GUI) |
| 18:42.11 | Qwell | AsteriskNOW is also a full Linux distribution. |
| 18:42.11 | legend1222_ | So basically all the ... em... garbage that I see when going thru the configs manually is there because of the GUI. |
| 18:42.20 | ManxPower | legend1222_: This is for GUIs in general. We ask for a pastebin of a failed call. In normal Asterisk it should be a couple of lines, in a GUI configured Asterisk the output could easily run over 100 lines. |
| 18:43.09 | legend1222_ | I get the impression that, overall, AsteriskNow is kinda a bad idea. True? |
| 18:43.29 | ManxPower | and, at least in some GUIs all the useful stuff is hidden inside an AGI. |
| 18:43.41 | ManxPower | legend1222_: All GUIs are a bad idea. |
| 18:43.45 | ManxPower | ~zeeek |
| 18:43.46 | jbot | somebody said zeeek was someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 18:43.48 | Qwell | legend1222_: No. If you need a full Linux distribution, and a GUI...go with AsteriskNOW |
| 18:43.57 | Qwell | if you want to play around with Asterisk, no, don't. |
| 18:44.53 | legend1222_ | Well, thats the thing. I've never had a problem with asterisknow, and I have taken to doing some manual configuration because I did find the GUI limited. But now I've got an issue, and there doesnt appear to be anyone listening the asterisknow irc room. |
| 18:45.23 | ManxPower | legend1222_: If you can't get support is it really a something you want to use? |
| 18:45.42 | legend1222_ | lol. My point. I kinda assumed asterisk was asterisk, just each with a pretty gui. |
| 18:46.08 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 18:46.41 | ManxPower | like with all the GUIs, all the users are newbies |
| 18:47.21 | legend1222_ | Thats exactly it. I started as a newbie, and basically still am. But I do have an install I need to debug. Can I give you guys an idea of the problem, and maybe at least point me in the right direction? |
| 18:47.47 | ManxPower | legend1222_: if you delete all your config files and start from scratch I'm sure we could help you. |
| 18:48.00 | legend1222_ | Grand. Thats helpful. |
| 18:48.09 | ManxPower | you might have better luck on the mailing list, but I doubt it. |
| 18:48.11 | CanWood | legend1222_: I don't know if you've ever done web ddesign, but think of the GUIs vs straight asterisk to be kinda like FrontPage generated sites vs straight HTML (or so I have found) |
| 18:49.04 | ManxPower | CanWood: an excellent comparison. Do you really think we want to wade thru all the front page crap to find your problem? |
| 18:49.57 | legend1222_ | Near as I can tell the issue is with asterisk itself, or the hardware. Nuthin to do with the GUI. I don't much care if I have to use a GUI or not. However, business world total noobs can't do anything else, and are terrified of the command line. And thats where this phone system is installed. Sounds like it was a bad idea. |
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| 18:50.58 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:52.16 | *** part/#asterisk legend1222_ (n=legend@66.178.252.218) |
| 18:52.42 | ManxPower | bkruse: legend1222_ is someone with an AsteriskGUI problem |
| 18:52.49 | ManxPower | of course, he exited just as you arrived. |
| 18:53.24 | bkruse | lol |
| 18:53.33 | bkruse | that's how it always is, isn't it ManxPower :P |
| 18:53.34 | NightKhaos | 1.4 is installing... had to fiddle with some overlays. |
| 18:53.45 | Qwell | NightKhaos: gentoo? don't use packages |
| 18:53.50 | ManxPower | that's twice now you have not been around when AsteriskNow users needed help. Slacker 8-) |
| 18:53.53 | Qwell | install directly from downloaded tarballs |
| 18:54.07 | bkruse | ManxPower: I got em now |
| 18:54.09 | NightKhaos | NightKhaos: yes, gentoo. I had to use the voip overlay. |
| 18:55.14 | Strom_M | Qwell: BUT BUT BUT BUT YOU DON'T KNOW THE AMAZING POWER AND VERSATILITY OF THE GENTOO PACKAGE OVERLAY MANAGEMENT SYSTEM AND HURRRRRRK JESUS I'M THROWING UP |
| 18:55.38 | NightKhaos | Strom_M: I feel your pain. |
| 18:55.42 | mikkel | Does anyone here have a Sangoma A500 card with local ISDN phones attached, that works ? |
| 18:56.16 | NightKhaos | Qwell: I am compling from source. |
| 18:56.27 | Qwell | Gentoo isn't "compiling from source". |
| 18:56.41 | Qwell | it's "compiling some package with arbitrary untested patches" |
| 18:57.15 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 18:57.39 | NightKhaos | Qwell: I'll let your ignorance slide. I don't like agruements over things as petty as package management systems. |
| 18:58.32 | Qwell | first off |
| 18:58.53 | Qwell | I heavily run Gentoo - I love Gentoo. I know how Gentoo works. |
| 18:59.09 | NightKhaos | And yet... you don't know what layman is? |
| 18:59.11 | Qwell | second, I'm a core developer of Asterisk. I've seen how their packages work, and they are terrible. |
| 18:59.39 | Qwell | calling somebody ignorant is not a way to make friends |
| 19:00.08 | NightKhaos | Well do you know what layman is? |
| 19:00.14 | Qwell | Yes, I do. |
| 19:00.31 | Qwell | just because you're using an overlay, doesn't mean the package is any better |
| 19:01.17 | NightKhaos | arbitary untested patches as you call them are used all the time. |
| 19:01.34 | NightKhaos | Look at BSD ports. |
| 19:02.17 | Qwell | You clearly missed my point, so, have fun |
| 19:03.02 | thomas | what is the transfer limit with iaxmodem ? the rate? |
| 19:03.09 | thomas | 9600 ? |
| 19:04.22 | hardwire | higher |
| 19:04.42 | hardwire | but you can use AT commands to query that info :) |
| 19:04.58 | thomas | hardwire: hm. how i can set higher? |
| 19:05.21 | hardwire | that all depends on your AT compatible application using iaxmodem's tty's |
| 19:05.23 | hardwire | what are you using? |
| 19:05.47 | thomas | hardwire: asterisk + iaxmodem +hylafax |
| 19:06.00 | hardwire | are you using iaxmodem's sample configs for hylafax? |
| 19:06.13 | hardwire | cause those init it correctly |
| 19:06.23 | thomas | hardwire: jep. sample. can i send you |
| 19:06.29 | hardwire | you also have to make sure the fax machine you are using to send to hylafax via iaxmodem supports more than 9600bps |
| 19:06.38 | hardwire | otherwise there will be no reason for a higher negotiation |
| 19:06.47 | NightKhaos | No. I did not. You are concerned that these patches are determental to the function of the service as supplied by the developers, i.e. you, and that by applying these patches I may, or may not, get the expected behaviour from my system. My counter agruement is that the developer, i.e. you, does not always account for every single circumstane of sysem configuration, BSD deals with this by ensuring the packages are installed in the |
| 19:06.49 | hardwire | fax modems top out at around 19200bps |
| 19:07.33 | hardwire | thomas: fax modulation is different than data modem modulation.. that's why you see lower rates. |
| 19:07.45 | hardwire | afaik |
| 19:08.15 | hardwire | if you were to use a data modem to dial into your iaxmodem, you would probably get a pretty neat connect string. |
| 19:08.55 | NightKhaos | hardwire: I seen 22.8 fax modems. |
| 19:09.33 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 19:11.37 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:13.27 | hardwire | NightKhaos: I stand corrected. |
| 19:14.19 | NightKhaos | hardwire: yeah it was an HP combination printer, copier, scanner, fax... |
| 19:15.09 | thomas | hmm. |
| 19:15.17 | thomas | hardwire: and you have iaxmodems? |
| 19:15.25 | hardwire | 4 |
| 19:15.31 | hardwire | hooked up to hylafax |
| 19:15.34 | hardwire | over gigabit :) |
| 19:15.36 | thomas | hardwire: and what is the limit for recieve ? |
| 19:15.38 | hardwire | it's creepy. |
| 19:15.44 | hardwire | thomas: haven't tested it yet. |
| 19:15.47 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 19:15.49 | hardwire | you could totally test it yourself |
| 19:15.58 | thomas | i have |
| 19:16.03 | thomas | i have send to the own iaxmodem |
| 19:16.08 | thomas | but 9600 is the limit :-( |
| 19:16.21 | *** part/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 19:16.36 | hardwire | thomas: get two iaxmodems working |
| 19:16.39 | thomas | alaw is correct or ulaw? |
| 19:16.40 | hardwire | make sure one can dial the other |
| 19:16.42 | hardwire | slin |
| 19:17.16 | hardwire | use slin |
| 19:17.17 | hardwire | :) |
| 19:17.22 | hardwire | make sure they can dial eachother in the dialplan |
| 19:17.42 | hardwire | then from one iaxmodem (turn off hylafax on both) dial the other using ATDT0001 (0001 being the example extension) |
| 19:17.54 | hardwire | and on the other side type in ATA |
| 19:17.55 | thomas | hardwire: http://paste.keks.be/69 |
| 19:18.05 | hardwire | you can do this via minicom or screen /dev/ttyIAX00 115200 |
| 19:18.06 | thomas | its local send and local recieve |
| 19:18.15 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 19:18.22 | hardwire | anyways, thems the foundation for testing what you want to know. |
| 19:18.32 | hardwire | you should totally be using slinear as the codec for these, btw. |
| 19:18.41 | thomas | slinear? |
| 19:18.47 | thomas | *translate* |
| 19:18.51 | hardwire | no need |
| 19:18.53 | hardwire | google it. |
| 19:18.57 | *** part/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-9c60d01c2e51d3f3) |
| 19:19.16 | thomas | hardwire: codec=slinear ? |
| 19:19.27 | thomas | aaeh allow=slinear ? |
| 19:19.35 | thomas | can you paste your config? paste.keks.be |
| 19:19.36 | thomas | <PROTECTED> |
| 19:19.38 | thomas | your iax |
| 19:19.39 | hardwire | it's in the example docs. |
| 19:19.52 | thomas | ah ok |
| 19:20.03 | hardwire | allow=slin |
| 19:20.15 | hardwire | the same goes into your iaxmodem config in /etc/iaxmodem/* |
| 19:20.39 | thomas | the same? no |
| 19:20.42 | thomas | codec slinear |
| 19:20.44 | thomas | ór ? |
| 19:21.17 | hardwire | yes.. that's fine |
| 19:22.27 | hardwire | uhoh |
| 19:22.33 | hardwire | bye thomas |
| 19:22.41 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
| 19:22.46 | ThoMe | re |
| 19:22.50 | hardwire | welcome back |
| 19:22.54 | ThoMe | what you have written? |
| 19:22.58 | NightKhaos | Finally! Asterisk is installed |
| 19:22.59 | thomas | screen is away :/ |
| 19:23.37 | ThoMe | hardwire: is it ok http://paste.keks.be/70 ? |
| 19:24.40 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 19:25.00 | hardwire | does it work? |
| 19:25.03 | hardwire | you don't need me to answer you.. :) |
| 19:26.00 | NightKhaos | right that's enough for now... I'll be back latter |
| 19:27.11 | bkruse | ManxPower: Ended up being a zaptel issue,. it looks like |
| 19:27.19 | bkruse | he was not a normal "gui" user, so it wasn't that bad :P |
| 19:29.51 | mikkel | Please, could soneone help my connect a ISDN phone to my Sangoma A500 card. Why does this have to be so difficult.... |
| 19:31.05 | mikkel | I have done all the sangoma says on there website. Incomming calls is working fine, but the ISDN phone is not ringing and the phone just says line busy |
| 19:32.50 | ThoMe | hardwire: :P |
| 19:33.53 | ThoMe | no.. only Jul 9 21:32:49 backup FaxGetty[5840]: RECV FAX (000000204): from freeLINE GmbH, page 1 in 0:12, INF, 3.85 line/mm, 2-D MMR, 9600 bit/s |
| 19:36.17 | hardwire | ThoMe: so.. up the quality from your fax machine |
| 19:36.35 | hardwire | see if it tries to negotiate a faster speed. |
| 19:36.41 | hardwire | your fax machine may be the limit here. |
| 19:36.52 | ThoMe | how i can test it? ;) |
| 19:37.08 | hardwire | look up the specs online |
| 19:37.16 | hardwire | doy ou have multiple iaxmodems? |
| 19:37.34 | *** join/#asterisk diegoferreira (n=tecnodie@mail.grupoabv.com.br) |
| 19:37.44 | ThoMe | jep. |
| 19:38.37 | hardwire | can you use sendfax at all? |
| 19:38.45 | ThoMe | hardwire: jep? |
| 19:39.00 | hardwire | you could attempt to send a fax to one of your other iaxmodems, from an iaxmodem. |
| 19:39.19 | hardwire | that would show off the capabilities a bit more with howeevr you have the phone network set up |
| 19:40.01 | hardwire | I need food |
| 19:40.05 | hardwire | ThoMe: Order me a pizza. |
| 19:40.21 | hardwire | you can probably fax a place.. |
| 19:41.28 | ThoMe | have send with sendfax -b 14400 -s a4 -n -f "bla" -D -d 08989223822 < /etc/resolv.conf |
| 19:41.54 | ThoMe | but 9600 bit/s :/ |
| 19:42.23 | hardwire | 08989223822 |
| 19:42.26 | hardwire | that's another IAXmodem? |
| 19:42.40 | ThoMe | jep |
| 19:43.50 | hardwire | and the hylafax process says 9600 baud for both iaxmodems? |
| 19:44.12 | ThoMe | hm, send i dont know. recieve 9600 |
| 19:44.25 | ThoMe | how i can get the value for send? |
| 19:44.37 | hardwire | check the logs |
| 19:47.03 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 19:47.09 | hardwire | I guess it just says it when recieved |
| 19:47.12 | hardwire | Jul 9 11:47:01 anc-hylafax-01 FaxGetty[13425]: RECV FAX (000000006): from ttyIAX001, page 1 in 0:13, INF, 3.85 line/mm, 2-D MMR, 14400 bit/s |
| 19:47.14 | hardwire | moo ha ha |
| 19:47.18 | hardwire | check out my bits baby! |
| 19:47.32 | ThoMe | hö?! |
| 19:47.40 | ThoMe | hardwire: local? |
| 19:47.40 | hardwire | anyways.. that was dialing from an iaxmodem, out the pri, in the pri, then back into another iaxmodem |
| 19:47.43 | hardwire | no big deal |
| 19:49.33 | *** join/#asterisk chigambamukoko (n=junk@71.55.10.211) |
| 19:50.46 | chigambamukoko | Greetings to all in the name of the creator |
| 19:50.49 | Peri | Here's a little background of my current learning project. I'm trying to configure an asterisk box as a voicemail server only, I can provision the users etc no problem, where i run into issues is when a call is redirected to the voicemail server from the platform, asterisk keeps responding with a 407 Authentication required response. The question is, is that because the domain on the VM server is different than that of the platform? |
| 19:51.50 | x86 | I've got one single analog user (out of say, 48) that is reporintg a rather strange issue... it seems sometimes when they dial the phone, it gives them a loud "robotic belching" sound and hangs up the channel |
| 19:51.59 | x86 | ManxPower: ever seen something like that? |
| 19:52.17 | x86 | just started happening "after lunch" today |
| 19:52.26 | x86 | could be bad phone? |
| 19:52.27 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 19:53.05 | Qwell | problems that start "after lunch" usually means "I spilled my coke into it" |
| 19:53.28 | x86 | but it's random, she can get some calls to go out without a robotic belching |
| 19:53.38 | x86 | and she said it happened a few times last week |
| 19:53.55 | x86 | but this is the only time it's done it this week |
| 19:54.07 | x86 | analog + asterisk == frustrating as hell :( |
| 19:54.23 | x86 | wish it worked as well as SIP does |
| 19:54.24 | chigambamukoko | check your harddrive in case has bad sectors etc |
| 19:54.41 | Qwell | what? |
| 19:54.44 | x86 | chigambamukoko: well I already checked the flux capacitor once... |
| 19:55.06 | x86 | chigambamukoko: although it could be the continium transfunctioner, I suppose |
| 19:56.18 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:56.27 | chigambamukoko | anyway, I'm looking for guys based outside of the US who may want to do a Voip partnership |
| 19:56.42 | macros73 | Anyone here have experience with distributed Asterisk implementations? IE, an Asterisk appliance in several branch offices using IAX trunks to connect to one of two Asterisk server/gateways? Each office would have 1-2 POTS connected to their appliance as a backup. |
| 19:57.19 | Qwell | chigambamukoko: "partnership"? |
| 19:57.35 | bkruse | Qwell: he wants to be the next EU vonage |
| 19:57.52 | Qwell | makes sense |
| 19:58.51 | chigambamukoko | anyone up to the challenge I can give you more info, system is up and running all i need is the pple |
| 19:58.54 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 19:59.00 | Qwell | to do what? |
| 19:59.08 | macros73 | to take his money, sign me up! |
| 19:59.18 | chigambamukoko | marketing, tech support, |
| 19:59.25 | Qwell | partnership implies investment |
| 19:59.26 | chigambamukoko | installation |
| 19:59.41 | chigambamukoko | no money involved |
| 19:59.45 | chigambamukoko | well.. |
| 19:59.49 | chigambamukoko | i take that back |
| 19:59.57 | macros73 | You better. :D |
| 20:00.00 | bkruse | Qwell: Western Union |
| 20:00.20 | chigambamukoko | I have the system up and running but you don't need to invest to become part of this |
| 20:00.35 | bkruse | chigambamukoko: When I purchase a pbx do I pay via western union, moneygram, or paypal? |
| 20:00.45 | Qwell | chigambamukoko: and why do people need to be outside the US? |
| 20:01.01 | *** join/#asterisk NightKhaos (n=nightkha@78-86-111-126.zone2.bethere.co.uk) |
| 20:01.02 | *** join/#asterisk lalmia (n=cmrahman@64.129.97.77) |
| 20:01.16 | chigambamukoko | because i think so many pple can benefit with voip compared to those in the us |
| 20:01.29 | chigambamukoko | actually if you are in the US, u r welcome to join |
| 20:01.37 | bkruse | chigambamukoko: Have you thought about such things as....latency? |
| 20:01.47 | macros73 | What is the pay range, assuming properly qualified? |
| 20:02.38 | chigambamukoko | we can do a conference call with those interested and go over my game plan to see if there is any adjustments we can make to match your country |
| 20:02.53 | chigambamukoko | latency is no issue |
| 20:03.10 | chigambamukoko | because we will create a vpn stratight to our boxes |
| 20:03.56 | jpeeler | because VPNs reduce latency |
| 20:04.05 | macros73 | Yes, VPNs are well known to reduce latency. |
| 20:04.27 | chigambamukoko | as far as pay range, those are issues that needs to be discussed because each individual will be different |
| 20:05.08 | chigambamukoko | depending on their level of participation |
| 20:06.44 | Qwell | uh huh |
| 20:07.19 | chigambamukoko | if this is something that catches your attention just pm me and we go from there, only those serious please |
| 20:08.47 | chigambamukoko | I am very much interested in those pple that have enough Voip and speak English and their native toungue |
| 20:08.51 | macros73 | Can I make $120,000 a year if I dedicate myself to your cause full-time? |
| 20:09.04 | macros73 | (Two weeks pay in advance, please, as a retainer.) |
| 20:10.28 | chigambamukoko | dude, i don't know if you know the statistics of Voip but as far as i know, about 30% is using Voip, which leaves about 70%, now calculate how many customers you at say 100/month |
| 20:10.56 | Qwell | 70% of the world population? |
| 20:11.01 | chigambamukoko | no |
| 20:11.05 | Qwell | well, the last time I looked, it was like 6 billion people |
| 20:11.08 | chigambamukoko | potential users |
| 20:11.37 | chigambamukoko | we are not even talking the world population here |
| 20:11.45 | Qwell | $5040000000000 per month. |
| 20:11.53 | chigambamukoko | there u go |
| 20:11.53 | Qwell | $5,040,000,000,000 |
| 20:12.02 | Strom_M | Qwell: is that in zimbabwe dollars? |
| 20:12.18 | chigambamukoko | na |
| 20:12.21 | Qwell | I'm pretty sure my math is off by several factors. |
| 20:12.22 | chigambamukoko | US my friend |
| 20:12.34 | Strom_M | because if we bring VoIP to the good people of zimbabwe, then they can talk over the phone about how their lives are shit |
| 20:12.35 | chigambamukoko | anyway |
| 20:15.11 | macros73 | Won't the good people of Zimbabwe try to kill us and claim our VoIP servers for their own, like they did with all the farm land? |
| 20:15.48 | jackson__ | Shoot, they won't be able to run the servers just like they can't farm the land... |
| 20:16.04 | chigambamukoko | they might, but we have so many servers its like wac-a-mole |
| 20:18.08 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-185.hsi.ish.de) |
| 20:19.40 | macros73 | chigambamukoko: Doesn't your service already exist under the name of "Skype?" |
| 20:19.52 | wonderworld | hey guys |
| 20:20.27 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 20:21.58 | chigambamukoko | not really, you c skype has a different model |
| 20:22.02 | *** part/#asterisk fogo (n=fogo@72.8.104.15) |
| 20:22.13 | chigambamukoko | our model is completely different |
| 20:23.47 | chigambamukoko | since the pool of potential customers is so vast, it really does not matter what skyp, vonage or anyone else is doing |
| 20:24.04 | Qwell | and what is your model? |
| 20:24.13 | chigambamukoko | :) |
| 20:24.22 | chigambamukoko | u want me tell you everything don't u? |
| 20:24.29 | Qwell | or anything |
| 20:24.42 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 20:24.56 | chigambamukoko | this is no scam or gimmick boys and girls |
| 20:25.08 | chigambamukoko | relax |
| 20:25.55 | wonderworld | skype succeded because they completely ignored standards and good behaviour to solve the NAT-problem. well... it works fine though. |
| 20:25.57 | NightKhaos | relaxes |
| 20:26.22 | chigambamukoko | funny NightKhaos |
| 20:26.28 | NightKhaos | I am |
| 20:26.41 | NightKhaos | gets back to configuring his Asterisk PBX |
| 20:26.52 | Strom_M | IRC is just like third grade |
| 20:27.00 | Strom_M | especially with the poking each other in the eye |
| 20:27.14 | NightKhaos | pokes Strom_M in the eye. |
| 20:27.25 | chigambamukoko | i feel you my friend |
| 20:27.31 | MatBoy | I have a choppy sound on a sip client that runs in a vmware windows host. Have more people seen this ? |
| 20:28.28 | chigambamukoko | MatBoy: choppy sound, what processor and ram do u have on that box |
| 20:30.29 | *** join/#asterisk PepOSX (n=angeldav@190.199.206.138) |
| 20:30.50 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 20:31.11 | MatBoy | chigambamukoko: Q6600 with 4 GB, 2 CPU's for the Vm and 2GB also |
| 20:33.15 | chigambamukoko | thats good, now how about hd drive, do you have specs like rpm, size brand |
| 20:33.16 | macros73 | MatBoy: I believe there is a fix for that mentioned on the pbxinaflash forum |
| 20:34.02 | macros73 | MatBoy: This might help, http://pbxinaflash.com/forum/showthread.php?t=66 |
| 20:34.21 | MatBoy | macros73: ok, let me look, I know there were PBX issues before with SMP |
| 20:34.33 | MatBoy | chigambamukoko: 7200rpm sataII disks |
| 20:34.57 | chigambamukoko | perfect, that eliminates what i was thinking |
| 20:35.04 | MatBoy | ok :) |
| 20:35.22 | MatBoy | yes I'm quite sticky with my specs, I need actually 8GB and 15K raptors once :P |
| 20:37.20 | MatBoy | chigambamukoko: I know about those 100HGz kernels, I always run them on vmware guests, but my windows XP workstation is also a Vm on that Q6600 |
| 20:37.27 | MatBoy | and i hav ethe idea that it's happening there |
| 20:38.31 | chigambamukoko | did u check that link macros73 provided? |
| 20:42.27 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 20:42.31 | macros73 | Anyone here try out OnSip? |
| 20:44.58 | *** join/#asterisk n9urk (n=IceChat7@rrcs-70-63-204-32.midsouth.biz.rr.com) |
| 20:45.16 | *** join/#asterisk dr_gogeta86 (n=gogeta@ppp-244-248.32-151.iol.it) |
| 20:45.18 | n9urk | hi all, May i get some help with IAX? |
| 20:45.28 | mvanbaak | ~ask |
| 20:45.29 | jbot | i heard ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:45.40 | n9urk | duh, lol |
| 20:45.43 | n9urk | I am setting up a 2nd * box and need the old to connect with the new |
| 20:46.00 | n9urk | and i keep getting an error |
| 20:46.34 | n9urk | "No registration for peer" |
| 20:46.37 | n9urk | is the error |
| 20:46.55 | x86 | register => foo:bar@baz in iax.conf should solve that |
| 20:46.56 | mvanbaak | did you read: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 20:47.02 | MatBoy | chigambamukoko: I'm checking Descheduled timeservice also |
| 20:47.32 | n9urk | this is the first time I have used 1.4.x. Has there been some change in IAX config with 1.4.x vs 1.4.2? |
| 20:47.37 | n9urk | I will look to see if I read that one |
| 20:48.49 | n9urk | mvanbaak thanks. looking at that one now. voip-info can be hard to navigate through |
| 20:51.24 | chigambamukoko | MatBoy: did you try to implent the mentioned changes, they seem like they could solve your problem, |
| 20:51.38 | n9urk | x86 that did not work |
| 20:51.54 | n9urk | x86 i had already tried taht and that was what gave me the error |
| 20:54.54 | n9urk | mvanbaak what should I look at next? |
| 20:55.09 | n9urk | I followed the instructions on the page |
| 20:55.22 | ManxPower | n9urk: all the changes are documented in upgrade.txt |
| 20:55.45 | ManxPower | you should look at the 1.2 upgrade.txt too (I believe it's included in recent versions of 1.4.x) |
| 20:56.03 | jaytee | could someone familiar with Polycom configuration clarify for me about the auto-divert function. I'm trying to figure out how to block specific phones from being able transfer to an external number or call forward to an external number. |
| 20:57.24 | jaytee | [TK]D-Fender suggested that config option earlier but everything I can find in the Polycom docs is both vague and seems to be more for handling calls to a phone not from a phone. |
| 20:57.33 | MatBoy | chigambamukoko: which one do you think about ? I don' t have issues on my asterisk host, but my sip-client guest :) |
| 20:57.51 | MatBoy | so a windows XP that also runs in vmware on another machine |
| 20:57.55 | ManxPower | jaytee: he has been wrong in the past |
| 20:58.35 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 20:58.37 | jaytee | ManxPower, not very often though and neither are you. At least neither of you are anything like the noob I am |
| 20:58.53 | jaytee | It's simple on a Nortel system per phone set. |
| 20:59.39 | jaytee | he said it's a 502 redirect but I'm not sure whether there's a way to trap that in * in the context for the extension or in the Polycom config. |
| 21:00.18 | ManxPower | jaytee: I doubt you can |
| 21:00.39 | *** join/#asterisk zerocan (n=ZeRoCoDe@88.238.21.165) |
| 21:01.04 | jaytee | so my system is wide open to toll fraud from internal employees then |
| 21:01.28 | chigambamukoko | MatBoy: so XP is the host, and thats where the client is running from correct? |
| 21:01.47 | jaytee | or we'll have to have someone reviewing outbound call reports from CDR on a frequent basis. |
| 21:02.10 | zerocan | himm I wanna install asterisk on my linux system but which linux distribution I have to use to make it up and running and stable as well could you share your experience? |
| 21:02.21 | jaytee | CentOS |
| 21:02.33 | jaytee | or Debian |
| 21:02.44 | jaytee | I prefer CentOS myself |
| 21:02.59 | MatBoy | chigambamukoko: no, linux is the host, XP is the guest |
| 21:03.24 | MatBoy | and I have another vmware machine which is a server, elastix runs there also in vmware server with a 100Hz kernel |
| 21:03.40 | zerocan | jaytee thx |
| 21:03.48 | chigambamukoko | i missed that part, but the client is in xp, correct |
| 21:03.53 | MatBoy | yes |
| 21:04.12 | MatBoy | I'm reinstalling vmware tools in some minutes |
| 21:04.43 | n9urk | is there any reason why I can't register one * box on the other when the iax.conf files seem correct? |
| 21:05.39 | chigambamukoko | if u look under performance, how much of each is being used, such as ram, cpu and drive? |
| 21:05.58 | chigambamukoko | i think u have to log into Administrative tools, and find performance |
| 21:06.10 | chigambamukoko | monitor or some such wording |
| 21:08.19 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 21:12.08 | MatBoy | chigambamukoko: all OK |
| 21:12.21 | MatBoy | I know my systems, I'm very sticky on that |
| 21:12.44 | chigambamukoko | what r u getting on the hardrive tho? |
| 21:12.48 | MatBoy | I also need to find out why my incoming sip call is " hanged up" every time |
| 21:12.59 | MatBoy | chigambamukoko: not that much usage |
| 21:13.19 | chigambamukoko | k, was this ever working or never worked to begin with? |
| 21:14.14 | MatBoy | I actually ever had this problem with sound I thought, so I'm going to reinstall the vmware tools and set teh Descheduled time service on the xp host and install vmtools again on the elastix server |
| 21:18.58 | *** join/#asterisk iNetForce (n=f@adsl-074-246-021-235.sip.mia.bellsouth.net) |
| 21:19.00 | jaytee | well, it's quittin time |
| 21:19.07 | jaytee | be back later from the homefront |
| 21:19.48 | iNetForce | Can I configure distintive ringing for call waiting on the appliance? I want the transfer calls to have a different ringtone than outside incoming calls |
| 21:20.06 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-4c8204927560b00c) |
| 21:24.03 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-185.hsi.ish.de) |
| 21:24.09 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 21:25.44 | *** join/#asterisk dr_gogeta86 (n=gogeta@ppp-244-248.32-151.iol.it) |
| 21:26.08 | iNetForce | Can I configure distintive ringing for call waiting on the appliance? I want the transfer calls to have a different ringtone than outside incoming calls |
| 21:26.21 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 21:26.49 | ManxPower | iNetForce: all Asterisk Appliance support should be handled by Digium. Call them. |
| 21:34.57 | iNetForce | i just want to know if it is posible |
| 21:35.05 | *** join/#asterisk rvhi (n=chatzill@udp197017uds.hawaiiantel.net) |
| 21:36.39 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:38.49 | ManxPower | iNetForce: since none of us have ever used the Asterisk Appliance and know nothing about it, since all support is handled by Digium, you would have about as much luck asking your question on #nokia |
| 21:41.02 | ManxPower | In Asterisk you would configure your phone (Asterisk does not setup phones) for this feature, then you would edit your dialplan to be sure to set the SIP Header before sending a call to the SIP device so that the SIP header triggers the different ring/call waiting. |
| 21:41.07 | hardwire | anybody seen a good hack or maybe real product for having a wireless headset on an spa94x series SANS lifter? |
| 21:41.12 | hardwire | with remote pickup? |
| 21:41.15 | hardwire | other than throwing a shoe at the phone. |
| 21:41.48 | ManxPower | For polycoms this is documented on voip-info.org |
| 21:43.04 | *** join/#asterisk PepOSX (n=angeldav@190.72.145.54) |
| 21:43.21 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-185.hsi.ish.de) |
| 21:44.38 | *** part/#asterisk mmattice (i=mmattice@unaffiliated/mmattice) |
| 21:44.53 | hardwire | looks around |
| 21:44.55 | hardwire | me? |
| 21:48.02 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 21:49.47 | *** join/#asterisk dmz (n=dmz@dsl-209-90-141-81.tor.primus.ca) |
| 21:50.33 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:59.15 | *** join/#asterisk Bananaskin (n=mike@user-514f01c8.l1.c3.dsl.pol.co.uk) |
| 22:04.55 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 22:05.14 | unpaidbill | holy hell, voip-info.org changed their site theme |
| 22:05.28 | seanbright | now if only their content didn't suck |
| 22:05.37 | unpaidbill | haha |
| 22:05.47 | unpaidbill | name a place with better asterisk related content |
| 22:05.51 | unpaidbill | i'd like to use it |
| 22:06.57 | jameswf-home | technically its a wiki so you could improve the content that you dislike |
| 22:07.34 | unpaidbill | whoa whoa whoa, you mean do something other than shit talking... hey man you got the wrong guys |
| 22:07.46 | unpaidbill | :P |
| 22:07.56 | seanbright | technically some jackhole can come behind me and say "WARNING: in Asterisk 1.0 this doesn't work, but in 1.0.1.4.6.8.9.12 it got fixed and broken again" |
| 22:08.02 | *** join/#asterisk serialthrilla (n=noemail@adsl-71-131-145-38.dsl.sntc01.pacbell.net) |
| 22:08.27 | seanbright | or my favorite: "This works like this" "NOTE: The previous statement is wrong!" |
| 22:08.35 | seanbright | morons. |
| 22:08.54 | Strom_M | the problem with being the lone person fixing the wiki is that you have to contend with the 3475130945716350834 other morons who keep breaking it again |
| 22:09.05 | seanbright | ding ding ding |
| 22:14.57 | lmadsen | sounds like the Asterisk Documentation Project needs to be revived :) |
| 22:16.28 | seanbright | might be a good complement to the docs xml stuff that mvanbaak et al are working on |
| 22:16.44 | lmadsen | yes, exactly |
| 22:16.58 | lmadsen | I would like to host the XML docs on that site, and then write articles around it, how-tos, etc... |
| 22:17.06 | unpaidbill | where are these xml docs |
| 22:17.13 | lmadsen | unpaidbill: they don't exist yet |
| 22:17.19 | lmadsen | the infrastructure if being developed |
| 22:17.25 | unpaidbill | ah, nice |
| 22:17.27 | seanbright | unpaidbill: there is a thread on the -dev mailing list discussing it |
| 22:17.32 | lmadsen | seanbright: TFoT HTML and PDF versions would also be hosted there |
| 22:17.54 | seanbright | lmadsen: who controls that site? you and jared? |
| 22:17.59 | lmadsen | yes |
| 22:18.39 | seanbright | wiki might suffice... a non-public one :) |
| 22:18.42 | seanbright | err |
| 22:18.45 | seanbright | non-public-editable |
| 22:18.53 | lmadsen | ya... we've looked at using something like Plone |
| 22:19.09 | lmadsen | where you can submit documentation, but it goes through an editing process before being published |
| 22:19.15 | seanbright | yeah |
| 22:19.18 | seanbright | that'd be hot |
| 22:20.35 | lmadsen | I've wanted to do it for a while, but haven't had the time... but it looks like things may be changing that could allow me to follow through on that plan |
| 22:20.46 | lmadsen | here's hoping! |
| 22:21.19 | Strom_M | lmadsen: NOTE!!!! This application is TOTALLY BROKEN in Asterisk 0.7.3!!!!!!! |
| 22:21.21 | seanbright | yup! :) |
| 22:21.24 | seanbright | haha |
| 22:21.43 | seanbright | that is what KILLS me about the wiki |
| 22:22.12 | seanbright | someone says something, and then someone else comes along and, instead of deleting the bogus content, they add a note saying the content is bogus |
| 22:22.14 | serialthrilla | dang, we should start attaching "beta" to those versions |
| 22:23.25 | Strom_M | oh, and in case anyone ever wants to get their polycom phone talking to a Commodore 64, let me paste all that stuff into the "Polycom IP430" page |
| 22:24.24 | ManxPower | http://www.fnords.org/~eric/polycom-config-examples/ |
| 22:29.24 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 22:30.14 | pcrane | morning all |
| 22:33.51 | *** part/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 22:39.23 | *** join/#asterisk vader-- (n=me@c-71-226-192-99.hsd1.nj.comcast.net) |
| 22:39.24 | vader-- | hello |
| 22:40.35 | *** join/#asterisk nephfl (n=none@wsip-70-168-186-225.ga.at.cox.net) |
| 22:40.58 | vader-- | was wondering how do you guys handle adding and removing users from asterisk? |
| 22:41.14 | vader-- | im trying to work something out a process, script or something to add and remove users from asterisk |
| 22:41.29 | vader-- | when we get a new hire i have to Create an account in our student management system, get the ID from that, create a domain account, add exchange, add live communications, then add them to their groups, then i have to add them into our phone system, give them a voicemail box, then add them into the copiers for access codes, then create another account in the student management system that allows them to do grades and attendance for studen |
| 22:41.37 | vader-- | then i have to assign them a laptop |
| 22:41.44 | vader-- | oh and a long distance code for dialing out |
| 22:41.52 | vader-- | i use excel spreadsheets for the Laptop, Copier Code and Phone System |
| 22:42.06 | vader-- | some people have phones, some people just have voicemail |
| 22:42.09 | vader-- | some have both |
| 22:42.30 | nephfl | sounds like a fun project |
| 22:43.11 | vader-- | ya |
| 22:43.16 | pcrane | vader--: we're using mysql realtime iax/sip peers |
| 22:43.25 | pcrane | had a look at that? |
| 22:43.29 | vader-- | na |
| 22:43.40 | pcrane | an entry in a database, and voila, up and running |
| 22:43.40 | vader-- | im just tired of messing with all these config files |
| 22:43.48 | pcrane | he |
| 22:43.51 | pcrane | heh* |
| 22:43.54 | pcrane | know what you mean |
| 22:44.04 | vader-- | sounds good but i don't knwo if i could adapt my current setup |
| 22:44.28 | nephfl | yeah, most of the projects rebuild the extensions from mysql ...so you just write a script to edit the extensions in mysql... |
| 22:44.51 | vader-- | right now i have been pushing to use one record ID for each faculty member |
| 22:44.59 | vader-- | through the student management system |
| 22:45.05 | pcrane | uses the same data fields as to create the sip/iax buddies, it just sends them to the DB instead of a config files |
| 22:45.18 | vader-- | how about voicemail? |
| 22:45.23 | vader-- | and dialplans? |
| 22:45.25 | pcrane | yep |
| 22:45.38 | pcrane | those can be taken care of with realtime mysql stuff too |
| 22:45.42 | vader-- | how about zap chans? |
| 22:45.50 | pcrane | that I don't know about... |
| 22:46.00 | vader-- | i have 24 analog channels |
| 22:46.07 | pcrane | but then, they're hardware based... why'd you need to change them? |
| 22:46.08 | vader-- | that handle things like fax machines |
| 22:46.17 | vader-- | and some users |
| 22:46.31 | vader-- | i have phones in just rooms with no users attached to them |
| 22:46.40 | serialthrilla | dont you just specify that it dials out that interface in the dialplan? |
| 22:47.31 | vader-- | ? |
| 22:54.57 | vader-- | can i use a voicemail.conf and mysql database at the same time? |
| 22:55.05 | vader-- | so i can test the sql realtime first? |
| 22:55.17 | ManxPower | vader--: using a database for less then a couple of thousand users is silly |
| 22:55.34 | ManxPower | it much harder to diagnose issues, the system is much more complex |
| 22:55.36 | vader-- | making changes to a database and doing lookups is easier |
| 22:57.34 | MatBoy | chigambamukoko: it's a vmware guest issue |
| 22:58.34 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:04.40 | *** join/#asterisk macros73_ (n=cs@c-67-165-65-27.hsd1.pa.comcast.net) |
| 23:05.11 | ManxPower | vader--: so do it for stuff you need in a database like customer account information |
| 23:05.55 | *** join/#asterisk TrentCreek (n=kvirc@red1.cs.panam.edu) |
| 23:05.58 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:06.00 | tompaw | Hi guys. |
| 23:06.21 | tompaw | Just to be sure: to use a single PSTN analogue line in asterisk to receive and make calls I need an FXO, right? |
| 23:06.34 | TrentCreek | Howdy grampaw.....what's for supper? |
| 23:07.05 | ManxPower | ~fxofxs |
| 23:07.06 | jbot | extra, extra, read all about it, fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 23:07.09 | ManxPower | tompaw: correct |
| 23:07.49 | TrentCreek | Well how about this one? |
| 23:08.17 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:08.24 | TrentCreek | I did asterisk -vvvvvvvvvvvvvvvvvvvvvr and I am not seeing call progress coming in from IAX channel |
| 23:08.59 | TrentCreek | ahh..i think I know why |
| 23:09.08 | TrentCreek | no i dont |
| 23:09.10 | tompaw | ManxPower: thanks, and FXS I'd use if I got an 'operator' route at my place, right? |
| 23:09.19 | TrentCreek | yes i so |
| 23:11.17 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 23:12.59 | tompaw | erm.. so those popular Linksys' SIP Gateways... they are in fact FXSes, arent' they? |
| 23:13.18 | serialthrilla | iax2 debug? |
| 23:15.30 | jaytee | tompaw, some are FXOs |
| 23:15.43 | jaytee | to connect to a phone line, not a phone |
| 23:15.46 | jaytee | some have both |
| 23:17.52 | tompaw | clear then, thank you |
| 23:18.09 | tompaw | and which FXO would you recomment for asterisk? x100p or digium's x100m modules? |
| 23:19.02 | andrewy | tompaw: both have a lot of problems. if you want a cheap FXO, look at something like a sipura 3000 (now a linksys 3something) |
| 23:19.02 | jaytee | I'd avoid the X100P since it's not supported or made anymore and most of what you see out there are cheap clones that lock up all the time. |
| 23:21.27 | tompaw | ok |
| 23:21.46 | tompaw | does this: http://store.digium.com/productview.php?product_code=1TDM422EF also cause problems? it's an official digium hardware, isn't it? |
| 23:22.10 | tompaw | andrewy: I'd like anything with the "linksys" name on it ;) |
| 23:23.09 | andrewy | tompaw: any of digium's current products will work well; the concern with the x100p is that it may be a clone |
| 23:23.20 | andrewy | it was discontinued a long time ago |
| 23:24.02 | jaytee | tompaw, that's a good card. it supports up to 4 FXO modules or 4 FXS module or any combination of either equalling a total of 4. |
| 23:24.05 | Nugget | pap2 > tdm* > gouging your eyes out with a rusty grapefruit spoon > x100p |
| 23:24.23 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 23:24.52 | jaytee | pap2t would be alot cheaper than a TDM410 card or any variation of it. |
| 23:25.04 | Nugget | and less of a headache |
| 23:25.13 | tompaw | :-) |
| 23:25.13 | C4colo | what about two x100p cards? |
| 23:25.26 | Nugget | I'd rather perform a self-vasectomy. |
| 23:25.29 | jaytee | what about raping yourself in the ass with a claw hammer? |
| 23:25.30 | tompaw | I already got pap2 and it's working flawlessly. |
| 23:25.37 | C4colo | yea |
| 23:25.44 | C4colo | x100ps are good for one thing |
| 23:25.46 | macros73_ | anyone here have hands on with the aa50? |
| 23:25.47 | tompaw | now I need to terminate the PSTN (receive and make calls) with Asterisk |
| 23:25.59 | C4colo | cheap zap timing source |
| 23:26.35 | jm|laptop | is PAP2 like the 3102 but without the router? |
| 23:26.36 | tompaw | I understand that with this Digium card I can hook up PSTN's rj11 directly into Asterisk, while SPA-3000 would be a IAX/SIP peer/ |
| 23:26.41 | tompaw | is that correct? |
| 23:26.45 | jaytee | yes |
| 23:27.08 | TrentCreek | WARNING[1545]: chan_sip.c:6797 determine_firstline_parts: Bad request protocol |
| 23:27.08 | TrentCreek | <PROTECTED> |
| 23:27.28 | jaytee | the SPA-3000 connects to * via SIP over ethernet. The Digium card goes in the server and requires a zaptel driver |
| 23:27.42 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 23:27.43 | Nugget | zaptel, erm, dahdi. :) |
| 23:27.56 | tompaw | right... as long as they're on LAN (spa and *) I guess it doesn't make much difference, does it? |
| 23:28.02 | TrentCreek | Ooooops! WARNING[1545]: chan_sip.c:6797 determine_firstline_parts: Bad request protocol |
| 23:28.25 | jaytee | tompaw, I don't think you'd notice any performance difference |
| 23:28.25 | C4colo | so is there something that would prevent the musiconhold=somethingelse from working in sip.conf for a user? |
| 23:28.28 | tompaw | and also I think spa gives me much more possibilities, cause ultimately I may want to pass this pstn to some remote asterisk over the internet. |
| 23:28.30 | wwalker | I'm setting up a new asterisk and trying not to pull in every conf file and every module. Things seem OK, but the extensions.conf file isn't being read. What module reads extensions.conf? |
| 23:29.36 | andrewy | tompaw: you could still pass it over the internet from your local asterisk box, but the spa is going to be significantly cheaper |
| 23:29.46 | wwalker | nm, found it |
| 23:30.36 | tompaw | andrewy: and doesn't require my asterisk to be switched on. it's just that this digium card LOOKS so nice ;-) |
| 23:31.01 | tompaw | one last question: SPA-3102 |
| 23:31.04 | jm|laptop | eek |
| 23:31.06 | tompaw | it says it also is both a FXS and FXO. |
| 23:31.09 | jm|laptop | has a SPA-3102 |
| 23:31.23 | tompaw | and what can you say about it? |
| 23:31.44 | jm|laptop | I found it difficult to configure |
| 23:31.59 | jm|laptop | you can arse around with settings for ages to get balance right for volume / echo |
| 23:32.17 | tompaw | it seems to be twice cheaper than SPA-3000. I wonder what it is so. |
| 23:32.50 | jm|laptop | it does some PRETTY STRANGE internal conversions between the FX channels and SIP - which adds latency and arguably echo |
| 23:33.11 | jm|laptop | it's good value; but I've had mine nearly two years now and still haven't got it /just right/ |
| 23:33.53 | jm|laptop | it also seems picky about the phone you plug into it |
| 23:34.07 | tompaw | so in general the best solution seems to be SPA-3000 |
| 23:34.10 | jm|laptop | I got shocking results with one DECT phone, but much better with a standard wired unit and another twin DECT set |
| 23:34.33 | jm|laptop | the SPA-3102 /is/ a 3000 isn't it? |
| 23:34.42 | tompaw | that digium card is tempting, but I got my asterisk running @ 64-bit build, I'm worried those zaptel drivers may not like it. |
| 23:34.53 | jm|laptop | but with a crapy PPPoE extension that I don't/can't use being PPPoA here |
| 23:35.04 | MatBoy | he, no sleep tonight, I need to fix my incoming sip trunk |
| 23:35.11 | tompaw | also, I won't have to get my PSTN line straight to the asterisk computer |
| 23:35.34 | [hC] | Yeah the SPA3102 is what alot of people want, but requires a lot of config dicking around. I wish there was a simpler version. |
| 23:35.36 | jm|laptop | tompaw: PSTN --> ATA ---> * though? |
| 23:35.39 | jm|laptop | what's the last leg? |
| 23:35.59 | jm|laptop | well PSTN --> FXS interface on ATA --> * |
| 23:36.01 | jaytee | tompaw, I ran 2 of those Digium cards on 64 bit RHEL 5 with no problems. I recently switched to the TE212P for PRI circuits so I'm not using them anymore but they worked great. |
| 23:37.06 | C4colo | the default text in sip.conf doesn't say anything about the musiconhold= option |
| 23:37.11 | jm|laptop | tompaw: I have a truck carrier and free incoming numbers [SipGate and similar] ... they all beat the PSTN option hands down |
| 23:37.14 | C4colo | is this actually an option? |
| 23:37.17 | jm|laptop | s/truck/trunk/ |
| 23:37.50 | tompaw | jm|laptop: I simply need an FXO to experiment with my card calling software. A pstn number that I already got in the wall for answering calls :-) |
| 23:38.02 | jm|laptop | Cisco 7912 --> Asterisk --> Sip provider [ ---> { h0h0magic } --> PSTN ] |
| 23:38.12 | nephfl | ive got a simple issue and i cant find the stupid issue... im using trixbox and my ivr is executing hangup after a certain amount of time...i cant find a timeout or anything |
| 23:38.29 | [hC] | AbsoluteTimeout? |
| 23:38.39 | tompaw | doesn't understand why SPA-3000 is more than 2x more expensive than SPA-3102 |
| 23:38.40 | jaytee | C4colo, musicclass NOT musiconhold |
| 23:38.48 | tompaw | jm|laptop: no need to, already gog PAP2T :-) |
| 23:38.57 | jm|laptop | hm |
| 23:39.02 | C4colo | ok, I'm going off of bad information then |
| 23:39.05 | nephfl | where is absolutetimeout set, i didnt see anything in globals or anything |
| 23:39.13 | C4colo | what is the option to override the music for a specific user? |
| 23:39.15 | C4colo | musicclass? |
| 23:39.16 | jm|laptop | I've been reading about that; it looks like the same chipset as the 3xxx |
| 23:39.31 | ManxPower | C4colo: it's IN sip.conf.sample Go Read It |
| 23:39.42 | C4colo | that's what I have open |
| 23:39.51 | jaytee | C4colo, pg 363 of the book |
| 23:39.54 | ManxPower | then you should see the music on hold options. |
| 23:39.56 | C4colo | "Users and peers have different settings available ... " |
| 23:39.58 | tompaw | now I'm confused. should I get 3000 or 3102 then? |
| 23:40.08 | C4colo | nothing listed for musicXXXX under there |
| 23:41.09 | tompaw | jm|laptop: but now as you mentioned it, I wouldn't mind one decent SIP phone. that cisco 7912 is pretty nice you say? |
| 23:41.25 | jm|laptop | er |
| 23:41.52 | jm|laptop | it comes with CSSP[?] (skinny) firmware so needs flashing |
| 23:42.12 | tompaw | anything else you'd recommend? how about linksys-branded phones? |
| 23:42.16 | ManxPower | C4colo: the file even TELLS YOU how to do that |
| 23:42.20 | jm|laptop | that's fun[!] - tftpd/dhcpd ahoy - plus to get the firmware you might need a Cisco support account (cough) |
| 23:42.42 | jm|laptop | tompaw: I've only used Ciscos but I have heard good of some Snoms ... not sure which models though |
| 23:42.50 | ManxPower | C4colo: but for more information look at channelvariables.txt in the doc dir for the CHANNEL variable |
| 23:42.57 | jm|laptop | the firmware for the 79xx is a bit rubbish, too. Only one line allowed; no distinctive rings etc. |
| 23:43.06 | jm|laptop | I was tempted to try skinny on them just to see |
| 23:43.29 | C4colo | musicclass doesn't work for the user either |
| 23:43.31 | ManxPower | C4colo: you must have missed upgrade.txt as well |
| 23:43.41 | tompaw | jm|laptop: forums confirm that on 3102 this routing feature cause some confusion from time to time. spa-300 it is, then! |
| 23:43.56 | jm|laptop | gl |
| 23:44.09 | Nugget | my impression of #asterisk: |
| 23:44.10 | Nugget | "Hi everyone, I need help. I'm trying to run trixbox with realtime patches on an eMachines server I found in the storage closet. I've got 40 users with a mixture of grandstream phones and x-lite (unregistered). I'm using four clone x100p cards I bought off ebay and I compiled a pirated version of the g729 codec. Can you help me set up fax over sip?" |
| 23:44.42 | ManxPower | Nugget: that's amazingly accurate! How DO you do it??? |
| 23:44.47 | Nugget | "I need help right away, because the phones are all down and my boss is pissed" |
| 23:45.04 | tompaw | jm|laptop: thanks. I just have to confirm if it's compatible with european PSTNs |
| 23:45.17 | jm|laptop | afaik it's ok for UK BT |
| 23:45.39 | C4colo | ManxPower: I have it printed on my desk with orange highlighter on the important bits |
| 23:45.44 | jm|laptop | not sure I'd buy anything called "PAP" |
| 23:45.54 | jaytee | C4colo, you need to set the musicclass in your musiconhold.conf file first then set which class each user gets in sip.conf with musicclass=classical or musicclass=rock etc, etc. |
| 23:45.56 | jm|laptop | its reputation sort of precedes it ... |
| 23:46.08 | ManxPower | C4colo: looks like you found a bug in the sip.conf.sample for 1.4 |
| 23:46.11 | wwalker | what first digits should I avoid in extensions? digits that are mapped to soemthing in asterisk by deafult (parking, etc.) |
| 23:46.45 | TrentCreek | How can I troubleshoot why IAX is not registering with a DID? |
| 23:46.46 | ManxPower | wwalker: You configure every thing, so you set up your dialplan and codes as you want. |
| 23:46.49 | wwalker | any reason to choose 3 digits over 4 digits or vice versa in an office that will never have 50 phones? |
| 23:46.52 | *** part/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:46.52 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:46.57 | ManxPower | TrentCreek: "iax2 show peers" |
| 23:46.58 | jm|laptop | er oops |
| 23:47.12 | TrentCreek | ManxPower: Thanks..trying now |
| 23:47.24 | ManxPower | wwalker: because those offices always end up having more than 50 extensions eventually |
| 23:48.02 | jaytee | wwalker, only if you have many 7 digit local DID numbers and you want to point the incoming call to the proper 4 digit extension that matches the last 4 of a 7 digit DID, otherwise 3 digits works fine in a 50 user office. |
| 23:48.21 | TrentCreek | ManxPower: It shows Offline, but does not tell me who |
| 23:48.24 | TrentCreek | *why |
| 23:48.46 | ManxPower | TrentCreek: then turn on iax2 debug and see what's happening |
| 23:48.49 | tompaw | jm|laptop: I need it for Polish TP, but since it's available in stores here, I think it will be fine. |
| 23:49.28 | C4colo | by the way, has anyone else had issues with reloading asterisk 1.4 causing it to hang? i.e. service asterisk stop / CLI> stop now / CLI> restart now ... etc are all unable to stop asterisk? |
| 23:49.39 | jm|laptop | crazy Poles |
| 23:49.48 | C4colo | I have had this on multiple systems, all 1.4 |
| 23:49.48 | jm|laptop | importing all your sexy women to the UK |
| 23:50.08 | jaytee | C4colo, no I've never had that problem nor have I ever heard of anyone having that problem. you must be the first :-) |
| 23:50.18 | C4colo | hmm |
| 23:50.37 | jm|laptop | [no seriously - it was in the news. Native women amongst the immigrant community are saying they're stealing all the men] |
| 23:50.54 | ManxPower | C4colo: I've only seen that when your DNS is seriously broken or Asterisk is getting ready to crash |
| 23:50.59 | jm|laptop | "They're so slim because they've been poor, like" was QOTD for me |
| 23:51.09 | C4colo | one of my customers has a system I didn't set up, running a very basic dialplan (it is a call router) that if it is reloaded asterisk just dies, or hangs, during the reload (about 50/50 die:hang) |
| 23:51.26 | C4colo | then on a system I am running with freepbx + asterisk (both from source) does it occassionally |
| 23:51.31 | C4colo | and an elastix system does it too |
| 23:51.42 | ManxPower | C4colo: they are of course running the latest 1.4.x so they are not encourntering one of the hundreds of bugs that were fixed since 1.4.0 |
| 23:51.49 | ManxPower | right, C4colo? |
| 23:52.02 | C4colo | 1.4.19 on the first |
| 23:52.03 | ManxPower | freepbx? Nevermind. |
| 23:52.14 | ManxPower | C4colo: perhaps you should try the latest 1.4 |
| 23:52.17 | C4colo | 1.4.19 on the freepbx + asterisk |
| 23:52.33 | C4colo | 1.4.5 on the elastix |
| 23:52.39 | jaytee | shit, if he can't even set a music on hold class right and he's setting up systems for other people I'm damn glad I'm not one of his customers. |
| 23:52.53 | C4colo | the point is, on all three systems, with different configurations, the same issue arrises from time to time |
| 23:52.54 | jm|laptop | meow |
| 23:53.20 | ManxPower | jaytee: dog help us all. It does NOT help that the option he needs to use was removed from the docs |
| 23:53.22 | jaytee | and the only thing they all have in common is?....................the person that set it up? |
| 23:53.31 | C4colo | no |
| 23:53.45 | C4colo | the first one was set up by someone else, I took over the contract when he receive a better offer from another company |
| 23:53.52 | jaytee | ManxPower, they removed it from the docs? seriously? |
| 23:54.03 | C4colo | the other two i set up, I thought it was something I did until I ran into it on the other system |
| 23:54.13 | *** join/#asterisk trevisa (n=chatzill@201-27-130-182.dsl.telesp.net.br) |
| 23:54.43 | C4colo | wait the other system is 1.4.4 |
| 23:54.47 | ManxPower | jaytee: I'm sure it was a mistake |
| 23:55.08 | ManxPower | jaytee: try to find musicclass in sip.conf.sample in a 1.4.x (I looked at 1.4.13) |
| 23:55.34 | C4colo | although, I can't vouch for the other guy who set the system up since he put "musiconhold=default" on all of the sip users |
| 23:55.38 | trevisa | Guys, I´m trying to install asterisk to Centos3 server, but it does not recognizes the kernel-sources packages I´ve isntalled. What should I do? |
| 23:55.46 | jaytee | C4colo, so they're all different versions of * then? one with freepbx. Is there anything you can think of that they all have in common if they all exhibit the same problem? |
| 23:55.54 | C4colo | which is where I got confused, since musicclass isn't listed under the options for users/peers |
| 23:56.04 | C4colo | heavy usage? |
| 23:56.16 | C4colo | well one doesn't have more than 4 or 5 calls at peak |
| 23:56.22 | jaytee | ManxPower, maybe it got deprecated from 1.2 but never got cut from the book? |
| 23:56.24 | C4colo | one has 60-70 calls at peak |
| 23:56.31 | ManxPower | are you using analog cards, C4colo? |
| 23:56.38 | C4colo | no |
| 23:56.48 | C4colo | most modules are disabled as we do not use them |
| 23:56.53 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 23:56.54 | ManxPower | jaytee: I don't think so, as there seems to be reference to it, it's just not listed |
| 23:56.57 | C4colo | including only ULAW and SLIN as codecs |
| 23:57.14 | C4colo | iax2 is disabled |
| 23:57.18 | ManxPower | C4colo: how are you connecting to the PSTN? How are the phones connected? |
| 23:57.38 | C4colo | on one system it is going through a Squire ss7 router |
| 23:57.42 | C4colo | via a SIP trunk |
| 23:58.02 | C4colo | and/or through a SIP trunk for long distance |
| 23:58.21 | TrentCreek | ManxPower is very busy |
| 23:58.22 | C4colo | on systems 2 and 3 they both use multiple SIP providers |
| 23:58.49 | C4colo | all connections in and out are SIP |
| 23:59.45 | C4colo | mostly I just wanted to find out if there were others having this problem ... i just thought it was isolated when it was just the two systems, but now I'm thinking there may be a bug |
| 23:59.55 | C4colo | I can research it more, disable more modules and such |