| 00:01.15 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:02.07 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-203.shs.nyc.wayport.net) |
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| 00:13.00 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 00:17.21 | heelios | Could anyone please recommend me some kind of FXO/FXS interface that would give my a decent bang for my buck? |
| 00:17.56 | heelios | s/my/me <_< |
| 00:20.36 | *** join/#asterisk salvatore2 (n=canberk@85.107.44.197) |
| 00:20.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 00:20.49 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:20.59 | salvatore2 | hi hi hi |
| 00:21.58 | exvito | heelios: how many fxo/fxs do you need ? for what kind of application ? |
| 00:22.58 | heelios | exvito: home, possibly (very) small office in the future. 1fxs/1fxo. |
| 00:23.35 | exvito | ...I would consider the utilization of an ATA such as the linksys/sipura SPA-3102 |
| 00:24.01 | exvito | 1 FXS + FXO gateway to SIP |
| 00:24.21 | exvito | ...with several builtin features... and quite competitive on $$$ |
| 00:25.24 | lirakis | ahh food |
| 00:25.27 | lirakis | yum |
| 00:25.59 | lirakis | [TK]D-Fender: yeah .. saw that about sco .. finally |
| 00:25.59 | heelios | exvito: alright. thanks a lot. |
| 00:29.09 | heelios | on another note, are grandstream phones any good? |
| 00:30.45 | exvito | ...my experience is limited but I'd advise them on a test / learn environment. And *against* them on production environments... |
| 00:30.58 | exvito | (limited to GXP-2000 / BT-102) |
| 00:31.44 | heelios | exvito: oh? what's bad about the GXP-2000? |
| 00:32.30 | [TK]D-Fender | ~gs |
| 00:32.37 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 00:32.37 | exvito | heelios: sound quality, build quality, firmware release quality, support quality... their QA sure needs improvement... |
| 00:32.38 | [TK]D-Fender | ~grandstream |
| 00:32.38 | jbot | rumour has it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 00:33.18 | exvito | ...small extra $$$ for Linksys SPA-9xx or SNOM phones is really worth it, IMHO. |
| 00:33.47 | heelios | aye. it's a shame. i thought they looked rather classy on pictures. <_< |
| 00:33.58 | salvatore2 | do you guys have an idea how to remote maintain several pap2 devices |
| 00:34.04 | salvatore2 | just like vonage doing |
| 00:34.08 | exvito | ...not so good on your hands... :-( |
| 00:34.22 | salvatore2 | i want my unlocked pap2 to download a firmware with all the configs in it |
| 00:35.00 | salvatore2 | ~book |
| 00:35.01 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:35.04 | [TK]D-Fender | In North America there is very little reason to consider anything but Polycom... |
| 00:35.29 | [TK]D-Fender | ~phones |
| 00:35.30 | jbot | rumour has it, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 00:35.37 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 00:35.42 | [TK]D-Fender | Snom is pretty far down the list. |
| 00:36.06 | engrxyz | <PROTECTED> |
| 00:36.26 | [TK]D-Fender | engrxyz, Looks like a phone checking for VM and the box doesn't exist |
| 00:38.45 | engrxyz | how to fix it |
| 00:39.26 | engrxyz | i have a sip phone connecting to asterisk in sip |
| 00:39.32 | [TK]D-Fender | engrxyz, stop telling your phone to ask for things it can't have, or check that your VM box is set up right |
| 00:39.50 | [TK]D-Fender | ok, time to head out, back later... |
| 00:39.57 | lirakis | later |
| 00:44.33 | salvatore2 | [TK]D-Fender, pls tell me, why don't u ever ask questions to this channel but answering questions all the time? |
| 00:53.13 | lirakis | salvatore2: is not here |
| 00:53.37 | salvatore2 | anyone with linksys remote provisioning knowledge? |
| 00:54.08 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
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| 01:32.55 | lirakis | hrmmm im having some issues with sound file playback and AGI |
| 01:33.15 | lirakis | i get audio when i use "EXEC PLAYBACK soundfile" |
| 01:33.26 | lirakis | but not when i use "GET DATA soundfile" |
| 01:33.34 | lirakis | or "STREAM soundfile" |
| 01:41.06 | citats | lirakis: whats the return value that you get from GET DATA or STREAM FILE? also are you sure your call has been properly answered? |
| 01:42.13 | lirakis | citats: ooo i dont know if i called Answer |
| 01:42.20 | lirakis | citats: let me check |
| 01:42.42 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 01:43.09 | lirakis | citats: yes .. i execute answer ... then i call AGI(myscript.agi) |
| 01:44.03 | lirakis | citats: im checking on the returns |
| 01:44.54 | tzanger | citats: word! |
| 01:45.11 | citats | heya tz, hows life? |
| 01:45.56 | tzanger | not bad and you? |
| 01:45.59 | tzanger | it's been a long time |
| 01:46.27 | citats | aside from a cold i've got not, everythings pretty good. trying to keep busy |
| 01:46.38 | citats | yeah sometimes life gets in the way of having fun :) |
| 01:46.53 | lirakis | citats: woah... wierd... i added a debug statement before the output and it played my GET DATA statement.. |
| 01:46.56 | lirakis | uhhh oookay |
| 01:47.27 | tzanger | How's your bro? What've you been up to? |
| 01:48.20 | citats | tzanger: my bro is good. on his 3rd marriage. this one came with 2 kids and just a few months back they had another girl. i think he'll stick with this one for a while |
| 01:48.31 | tzanger | heh |
| 01:48.36 | tzanger | I'm about to get married the 2nd time |
| 01:48.39 | tzanger | I've got 2, she's got 2 |
| 01:48.49 | tzanger | and you? You're still single? |
| 01:49.42 | citats | i finally got married, hmmm must be 2 years back. to my long time girlfriend. funny story, we flipped a coin in vegas andit came up heads so we got married |
| 01:50.47 | lirakis | wtf!? this is being hyper finicky |
| 01:51.15 | citats | lirakis: have you turned on agi debugging in asterisk to see what is actually being sent and received? |
| 01:51.20 | tzanger | citats: hahaha nice |
| 01:52.02 | lirakis | set agi debug? |
| 01:52.26 | lirakis | one sec i got it |
| 01:52.41 | citats | we got married shortly before my wife was moving out here to WA for a job. she finished up her PhD and was offered a position at a national lab. so if we were married they would pay to move me out too |
| 01:52.52 | citats | lirakis: "agi debug" at the console |
| 01:54.02 | tzanger | shit I do remember that now |
| 01:54.09 | lirakis | citats: yeah.. i did that.. im looking at it now trouble shooting |
| 01:54.11 | tzanger | we spoke last just when you were moving |
| 01:54.15 | lirakis | citats: thanks |
| 01:54.24 | citats | ahh ok, its hard to keep track of things |
| 01:57.42 | x86 | re |
| 01:58.28 | lirakis | citats: here is a pastebin of the agi debug from cli ... its pretty short .. dont worry ;) |
| 01:58.28 | lirakis | http://pastebin.com/d58c613c |
| 01:59.17 | lirakis | citats: it does not actually playback astcc-digit-account-number |
| 01:59.34 | lirakis | it just says "please enter your ten ... you entered zero" |
| 02:01.24 | citats | lirakis: GET DATA should have some default values but you might want to try passing a timeout and max digits. like GET DATA astcc-digit-accont-number 15000 4 |
| 02:01.46 | citats | tzanger: how long before you get hitched again? |
| 02:02.08 | tzanger | citats: 4 weeks |
| 02:02.50 | *** join/#asterisk CoolGuy21 (n=Tilt@cpe-76-175-234-137.socal.res.rr.com) |
| 02:02.51 | CoolGuy21 | hi |
| 02:02.51 | citats | tzanger: well congratulations. i hope everything works out well. having a big ceremony and everything? |
| 02:02.55 | CoolGuy21 | can someone help me |
| 02:02.58 | lirakis | citats: same thing :( |
| 02:03.07 | CoolGuy21 | im in asterisk cli and i dont understand what this means : "-- Got SIP response 405 "Method Not Allowed" back from 216.55.136.9" |
| 02:04.18 | lirakis | coolgu21: Client Error — the request contains bad syntax or cannot be fulfilled at this server |
| 02:04.34 | *** join/#asterisk guillote_GNU (n=guillote@host3.200-117-189.telecom.net.ar) |
| 02:05.24 | lirakis | citats: I dont know whats going on .. this is a simple like 5 line agi script.. the GET DATA was working ... now it stopped... i literally havent changed anything but the white space around it |
| 02:05.55 | citats | lirakis: very strange. in your output i dont see asterisk sending a response to your get data at all |
| 02:06.02 | lirakis | CoolGuy21: specifically "method not allowed" |
| 02:06.33 | CoolGuy21 | hummm |
| 02:06.41 | CoolGuy21 | how do i know which method are we talking about? |
| 02:06.48 | lirakis | citats: yeah.. it does actually show "Playing 'astcc-digit-account-number' (language 'en')" |
| 02:07.02 | lirakis | CoolGuy21: you are being far too vague |
| 02:07.13 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 02:07.58 | lirakis | citats: do you know another way to get input from a user with AGI? |
| 02:08.04 | *** join/#asterisk nighty^ (n=nighty@p3132-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 02:08.40 | tzanger | citats: no, not real big at all. and thanks |
| 02:08.51 | tzanger | I don't think I'm as brave as your brother to try a 3rd time :-) |
| 02:09.19 | citats | lirakis: you can loop using "wait for digit" |
| 02:10.18 | x86 | tzanger: dude... DONT DO IT... IT'S A TRAP |
| 02:10.34 | lirakis | citats: i got some thing... i added an extra "read" on stdin .. before i called GET DATA .. and now it works |
| 02:10.36 | citats | tzanger: i'm not sure brave is the right word for it. i tried to tell him not to go for the first one, or at least to wait a while before actually getting married. the 2nd one was just a rebound thing and happened quickly. i think this 3rd one will last |
| 02:11.01 | citats | lirakis: whats your agi written in? |
| 02:11.03 | lirakis | citats: i dont know wtf is going on with that.. i guess i just need to be doubly sure the stdin buffer is clear |
| 02:11.07 | lirakis | citats: php |
| 02:11.21 | citats | lirakis: theres a php module to make agi easier, should handle all the reading and writing |
| 02:11.24 | tzanger | x86: there's a meme I haven't heard in a while |
| 02:11.34 | tzanger | citats: yeah it's no fun that is for sure |
| 02:15.04 | lirakis | citats: ... thats reallly .. not hard... i mean.. i just wrote two little functions .. its basically fgets($in, 4096) or fputs($out,$data) |
| 02:16.55 | lirakis | citats: im going to stay away from STREAM FILE though... apparently it can be a bit buggy.. EXEC PLAYBACK seems to be solid |
| 02:17.01 | citats | lirakis: i just find it easier to use a module to handle repetative tasks, thats why i wrote the perl agi module |
| 02:24.16 | lirakis | citats: .. i almost bought the oreilly "programming perl" book last weekend... but i have too many projects now.. learning another language is not going to speed shortening my list ;) ... but i here cpan is a great thing |
| 02:27.01 | citats | lirakis: i always say stick with what you know. the language is just a tool |
| 02:28.58 | lirakis | citats: yeah.. im looking at phpagi.. it does look pretty good and straitforward.... i was worried that using some abstraction layer would muck it all up.. but .. its 1 simple class .. |
| 02:29.08 | *** join/#asterisk miroesq13 (i=miroesq1@81.10.25.12) |
| 02:31.27 | citats | guess i should go make some dinner. i promise i'll be back again soon :) |
| 02:31.43 | miroesq13 | i just bought two g729a licenses from digium. i have a 2.00 ghz celeron processor and was wondering which codec package i should be downloading as they have so many versions. any help is appreciated. |
| 02:36.44 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 02:38.34 | miroesq13 | anyone? |
| 02:42.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:43.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:48.33 | miroesq13 | hello |
| 02:49.22 | *** join/#asterisk ming_zym (n=ming_zym@124.254.55.239) |
| 02:52.24 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 02:52.25 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:55.57 | x86 | miroesq13: what is available? |
| 02:56.22 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 02:58.50 | *** join/#asterisk mog (n=mog@adsl-220-184-114.mob.bellsouth.net) |
| 02:58.50 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:58.57 | miroesq13 | thank you. they have the following |
| 02:59.10 | miroesq13 | c3-2 |
| 02:59.13 | miroesq13 | c3 |
| 02:59.21 | miroesq13 | i386 |
| 02:59.23 | miroesq13 | i586 |
| 02:59.26 | miroesq13 | i686 |
| 02:59.35 | miroesq13 | pentium-m |
| 02:59.41 | miroesq13 | pentium3m |
| 02:59.49 | miroesq13 | pentium4m |
| 02:59.58 | miroesq13 | and a couple of other that i think are just for amd |
| 03:00.30 | x86 | can't go wrong with generic i686 |
| 03:02.00 | miroesq13 | ok, but the i686 zaptel drivers were not doing so well in my system using the latest trixbox 2.2.4. i586 is what loaded when i did yum -y update |
| 03:02.58 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 03:04.29 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:08.29 | miroesq13 | x86: thanks, will go try it now :) |
| 03:10.16 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:23.36 | *** join/#asterisk mtaht4 (n=m@dns1.eruditium.org) |
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| 03:44.52 | lirakis | hrmm i should be going to bed soon |
| 03:47.23 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 03:48.52 | Mavvie | Asterisk SVN-branch-1.2-r57753M built by root @ tardis.barnet.com.au on a i686 running Linux on 2007-03-05 06:31:48 UTC |
| 03:48.54 | Mavvie | 3... |
| 03:48.55 | Mavvie | 2... |
| 03:48.55 | Mavvie | 1... |
| 03:48.58 | Mavvie | Asterisk SVN-branch-1.4-r78907M built by root @ tardis.barnet.com.au on a i686 running Linux on 2007-08-11 03:40:32 UTC |
| 03:50.16 | Mavvie | CRAAAAAAAAAAAAAAAAAAACK! |
| 03:50.18 | Mavvie | asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: span_set_message_handler |
| 03:50.29 | *** join/#asterisk WindBack (n=jorge@host141.190-31-69.telecom.net.ar) |
| 03:50.42 | WilliamK | mavvie, are you trying to emulate my knee? |
| 03:50.53 | WilliamK | it goes crack and pop constantly |
| 03:50.55 | WilliamK | :( |
| 03:51.01 | WindBack | smtp |
| 03:51.01 | Mavvie | I hope not for you... |
| 03:51.09 | WindBack | pop3 |
| 03:51.14 | WindBack | or imap |
| 03:51.30 | WindBack | yes |
| 03:51.45 | WindBack | but i don't know |
| 03:51.58 | WindBack | and?? |
| 03:52.20 | WindBack | ok |
| 03:52.55 | WindBack | If I connect the sip protocol with the MTA |
| 03:53.34 | *** part/#asterisk WindBack (n=jorge@host141.190-31-69.telecom.net.ar) |
| 03:53.45 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:56.18 | Mavvie | it is defined in libspandsp.so.0.0.2... |
| 03:56.42 | Mavvie | maybe asterisk isn't linked with libspandsp |
| 03:57.10 | Mavvie | app_rxfax surely isn't. |
| 03:57.38 | SwK | *yawn* |
| 03:59.59 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:03.22 | Mavvie | *CLI> unload app_rxfax |
| 04:03.22 | Mavvie | Segmentation fault |
| 04:03.24 | Mavvie | nice |
| 04:05.53 | Mavvie | but app_rxfax now works again. |
| 04:11.17 | *** join/#asterisk IguanaNed (n=you@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 04:11.36 | IguanaNed | good morning good evening and good night |
| 04:11.42 | IguanaNed | anyone around? |
| 04:11.53 | Mavvie | yes |
| 04:12.23 | IguanaNed | prbbly been ask a million time here.. but I am having major trouble getting meetme to work |
| 04:12.34 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 04:12.36 | Mavvie | do you have a hardware card? |
| 04:12.39 | IguanaNed | ztdummy |
| 04:12.41 | IguanaNed | nope |
| 04:12.51 | PioneerVM2 | anyone know how to clear the registry table in asterisk |
| 04:12.56 | PioneerVM2 | for sip |
| 04:13.00 | Mavvie | exten => 93353697,1,MeetMe(,cDiMp) |
| 04:13.04 | Mavvie | that's what I have, works fine. |
| 04:13.33 | PioneerVM2 | it seems asterisk keeps trying to reach my home server and poking a hole thru firewall, its been throwing me for a loop for an hour, its kept the IP of my home machine if i do "sip show peers" and keeps pinging it or something |
| 04:13.45 | PioneerVM2 | even if the box is off at home |
| 04:14.15 | IguanaNed | Mavie I had it working before |
| 04:14.54 | IguanaNed | my hosted server was rebooted.. and the hosting company had to boot with a prior version of the kernel |
| 04:14.57 | bbryant | PioneerVM2, if you delete the SIP entry from the database, and reload sip, the problem should be fixed |
| 04:15.05 | PioneerVM2 | any easier way like a command? |
| 04:15.10 | bbryant | you can remove it by doing "db del SIP/whatever" iirc |
| 04:15.12 | Mavvie | IguanaNed: did you rebuild ztdummy and asterisk? |
| 04:15.18 | bbryant | then "sip reload" |
| 04:15.21 | Mavvie | (after the kernel changes) |
| 04:15.37 | IguanaNed | i rebuilt ztdummy |
| 04:15.40 | PioneerVM2 | hmm db del doesnt work |
| 04:15.43 | PioneerVM2 | says no such command |
| 04:16.11 | IguanaNed | modprobe ztdummy does not work... should it? |
| 04:16.13 | bbryant | it's: database del |
| 04:16.14 | bbryant | sorry |
| 04:16.19 | Mavvie | Igbothom_3rd: yes |
| 04:17.36 | IguanaNed | I dont understnd why ztdummy always gives me a hard time... this is my third time having to reinstall my OS cuz my hosting company swcrewed up |
| 04:17.53 | IguanaNed | everytime I have rebuiltI always run into problems with ztdummy |
| 04:17.53 | PioneerVM2 | hmm cant find it in the database ill just remove it as u suggested and see if that works |
| 04:17.55 | Mavvie | it's a kernel module. |
| 04:19.33 | IguanaNed | at this pont I am ready to scrap meetme |
| 04:19.45 | IguanaNed | anyone have ay experience with app_conference? |
| 04:20.16 | PioneerVM2 | bb: that worked thanks! |
| 04:20.25 | PioneerVM2 | freaking been driving me nuts -- i thought my firewall was screwed up. |
| 04:20.41 | PioneerVM2 | i kept blocking all IP to my asterisk server and my home Linksys box kept reconnecting somehow |
| 04:21.02 | PioneerVM2 | turns out since i was registering, the asterisk box was poking back at me thru the firewall which opened up a hole to me |
| 04:21.18 | PioneerVM2 | even if i reset the linksys box |
| 04:21.23 | IguanaNed | modprobe zaptel |
| 04:21.23 | IguanaNed | FATAL: Could not load /lib/modules/2.6.20-1.2948.fc6/modules.dep: No such file or directory |
| 04:22.48 | IguanaNed | uanme -r reports 2.6.20-1.2948.fc6 |
| 04:22.59 | PioneerVM2 | is there a way to supply a "host" IP in the registry manually -- like force register a device on the asterisk side without receiving a remote register? |
| 04:23.07 | Mavvie | hello linux land... |
| 04:23.15 | PioneerVM2 | like supply an IP in the sip.conf file of the remote box? |
| 04:23.45 | IguanaNed | but I only have /lib/modules/2.6.18-1.2798.fc6 |
| 04:23.59 | IguanaNed | I am gettin very confused |
| 04:28.41 | SwK | in sql anyone know the correct syntax for doing something like 'foo?bar:foobar' in C |
| 04:29.09 | Qwell | SwK: IIF |
| 04:29.25 | Qwell | something like IIF(conf,true,false) |
| 04:30.22 | Qwell | erm, cond |
| 04:31.25 | SwK | ok |
| 04:31.32 | SwK | know that you tell me I remember |
| 04:31.34 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 04:31.41 | SwK | but it appears thats not good in mysql haha |
| 04:32.50 | SwK | hah mysql its just if |
| 04:34.18 | PioneerVM2 | anyone here know how stun works? specifically -- if you are using it does it only check the stun server the first time it makes a connection, or on every call ? |
| 04:34.29 | PioneerVM2 | (or every time it reregisters) |
| 04:34.32 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
| 04:39.07 | SwK | Qwell, thanks man |
| 04:39.15 | Qwell | np |
| 04:40.41 | PioneerVM2 | anyone know if there are any web hosting services that specialize in offering asterisk servers? |
| 04:40.56 | PioneerVM2 | or would i just get a regular web hosting box and drop asterisk on it? |
| 04:41.36 | SwK | PioneerVM2, just get a decent one |
| 04:41.51 | JT | don't get shared web hosting obviously |
| 04:41.56 | PioneerVM2 | yea |
| 04:41.57 | SwK | those with severly oversold bandwidth and such are bad for voip |
| 04:42.08 | PioneerVM2 | my issue is i have my own colo and like 15 machines behind a cisco pix |
| 04:42.16 | PioneerVM2 | first, cisco pix apparently sucks for voip |
| 04:42.24 | JT | it sucks for everything |
| 04:42.42 | PioneerVM2 | second, i want to open up the asterisk server to allow for dynamic connections but keep the rest of my stuff secure |
| 04:42.43 | SwK | PioneerVM2, pix do suck for voip but SIP will work thru it |
| 04:42.55 | heelios | pix is as bad as it gets. |
| 04:43.00 | PioneerVM2 | yea i had to use stun for my home machine :( |
| 04:43.08 | PioneerVM2 | pain in the a$$ |
| 04:43.19 | SwK | is the PIX doing NAT/ |
| 04:43.36 | PioneerVM2 | i believe so |
| 04:43.41 | SwK | thats the suck |
| 04:43.51 | PioneerVM2 | i did get it to work |
| 04:44.03 | PioneerVM2 | the probelm i have occasionally is i get a call in and cant hear anyone, but if they call right back it works |
| 04:44.07 | PioneerVM2 | its only 1/10 calls |
| 04:44.25 | PioneerVM2 | otherwise it works fine, thats to my home box though thru the asterisk server -- calls into the asterisk box to voice mail seem to work fine |
| 04:44.44 | PioneerVM2 | so its some issue between asterisk server behind pix and my home boxes (linksys fw) |
| 04:44.57 | PioneerVM2 | been trying to figure it out, wondering if connection getting dropped due to timeouts or something |
| 04:45.13 | JT | the issue is probably that you have a pix ;) |
| 04:45.23 | PioneerVM2 | yea im pretty sure thats the problem |
| 04:45.39 | PioneerVM2 | what do u suggest to replace pix |
| 04:46.00 | PioneerVM2 | im looking for some off the shelf solution with easy config |
| 04:46.35 | SwK | PioneerVM2, you manage the pix yourself? |
| 04:46.40 | PioneerVM2 | yes |
| 04:49.02 | PioneerVM2 | ? |
| 04:49.30 | SwK | if you have sip fixup or the sip helper enabled disable it |
| 04:49.36 | PioneerVM2 | yes i did that already |
| 04:49.40 | SwK | ok |
| 04:49.42 | PioneerVM2 | sip to my sip peers works fine |
| 04:49.52 | PioneerVM2 | to my home i had to use stun AND disable those fixups |
| 04:49.58 | SwK | thats one of the biggest problems w/ pix |
| 04:50.01 | PioneerVM2 | yea |
| 04:50.23 | PioneerVM2 | any suggestions of another firewall to use? |
| 04:50.32 | PioneerVM2 | preferably solid state |
| 04:52.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:54.05 | SwK | PioneerVM2, off the shelf PC running OpenBSD/PF using DOC or CF for the HDD |
| 04:54.26 | PioneerVM2 | id prefer something already integrated if possible |
| 04:54.34 | PioneerVM2 | ive been told openbsd many times but hate to deal with yet another PC |
| 04:55.25 | PioneerVM2 | i deal with too much other crap than to deal with something new, thats kinda why i like pix as much as it sucks |
| 04:55.28 | SwK | PioneerVM2, actaully its quite robust and you dont have to load anything but the base OS... PF is default in the kerbel |
| 04:55.55 | PioneerVM2 | whats DOC, CF=compact flash i presume |
| 04:56.11 | PioneerVM2 | does anyone sell an openbsd box ready to go like that? |
| 04:56.12 | SwK | DOC == Disk on Chip |
| 04:56.27 | SwK | PioneerVM2, hawk.com in NYC |
| 04:56.36 | SwK | http://www.nmedia.net/~chris/soekris/ check that out |
| 04:57.12 | SwK | I dont think its upto date that... need to email chris and ask him whats up |
| 04:57.28 | SwK | there is also m0n0wall and that has a gui |
| 04:58.26 | PioneerVM2 | does hawk.com sell a ready to go openbsd firewall, or just a computer provider |
| 04:58.51 | PioneerVM2 | i'd be open to trying if i can find someone thats already done the dirty work -- with a ready to go box with a solid state drive and openbsd installed |
| 04:59.25 | PioneerVM2 | http://www.jtan.com/jtanoss/cdboot/ |
| 04:59.54 | SwK | PioneerVM2, i used to work there we sold obsd boxes ready to go for firewalls |
| 05:00.19 | PioneerVM2 | cool ill give them a call |
| 05:00.47 | PioneerVM2 | this cd thing may work, i guess i just need a machine with a CD and flash drive |
| 05:01.32 | SwK | PioneerVM2, check your PMs |
| 05:01.39 | PioneerVM2 | PM? |
| 05:03.53 | SwK | private messages |
| 05:04.24 | PioneerVM2 | how do i check those here |
| 05:05.19 | SwK | in icechat i am not sure... another window possibly |
| 05:05.32 | PioneerVM2 | hmm i dont see anything |
| 05:05.52 | SwK | did you see what I said about who to tell him sent you? |
| 05:06.02 | PioneerVM2 | no |
| 05:06.16 | PioneerVM2 | just see the regular conversation, no pms |
| 05:06.17 | PioneerVM2 | hm |
| 05:06.22 | SwK | hmmm |
| 05:09.59 | PioneerVM2 | yes |
| 05:19.56 | JT | it helps to know how to use the irc client you have ;) |
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| 05:56.56 | obnauticus | Whats a good outbound provider? |
| 05:57.00 | obnauticus | cheap/good quality |
| 05:57.52 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:01.04 | obnauticus | :( |
| 06:12.30 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
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| 06:49.20 | littleball | hello, who can give me hint about how to modify app_xxx.c file and to add some special channel variable? |
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| 06:59.53 | x86 | anyone still up? |
| 07:00.14 | Pagautas | allready up ) |
| 07:00.15 | Pagautas | :) |
| 07:01.28 | x86 | hehe |
| 07:01.38 | Pagautas | 10:01:27 < x86> hehe |
| 07:01.58 | x86 | Sat Aug 11 02:01:58 CDT 2007 |
| 07:02.19 | Pagautas | :) |
| 07:02.49 | x86 | where are you located? |
| 07:02.55 | Pagautas | lithuania |
| 07:03.16 | x86 | ah cool |
| 07:03.22 | x86 | what do you use Asterisk for? |
| 07:03.42 | Pagautas | yes |
| 07:03.48 | x86 | littleball: add the channel variable when you create the channel :p |
| 07:04.26 | x86 | littleball: exten => s,1,Set(foo=bar) exten => s,n,Dial(SIP/foo@bar) |
| 07:05.27 | Pagautas | so what do you want to ask? :) |
| 07:06.59 | x86 | Pagautas: what do you use Asterisk for? |
| 07:07.09 | x86 | personal, or for work? |
| 07:07.29 | Pagautas | at work |
| 07:07.49 | x86 | ah cool |
| 07:07.52 | x86 | how big? |
| 07:08.01 | Pagautas | 1200 users |
| 07:08.17 | Pagautas | its university network :) |
| 07:08.22 | x86 | wow :) |
| 07:08.28 | x86 | what kind of channels are you using? |
| 07:08.35 | x86 | Zap, SIP, IAX2? |
| 07:08.47 | Pagautas | sip, iax and h323 |
| 07:08.55 | x86 | ewww h323 _sucks_ |
| 07:08.57 | Pagautas | h323 is from old system |
| 07:09.02 | Pagautas | i know |
| 07:09.08 | Pagautas | we are trying to change it |
| 07:09.14 | x86 | so you have no physical trunks? |
| 07:09.32 | Pagautas | 4xE1 |
| 07:09.53 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 07:09.58 | x86 | cool :) |
| 07:10.03 | x86 | 1 asterisk server? |
| 07:10.05 | Pagautas | im using 2x cisco routers with 2xE1 |
| 07:10.10 | Pagautas | yes one |
| 07:10.19 | x86 | a lot to handle for a single server |
| 07:10.40 | Pagautas | :) |
| 07:11.05 | Pagautas | it seems to be enough for now |
| 07:11.21 | Pagautas | ahh for h323 i have another server |
| 07:12.18 | Pagautas | x86: what harware using for fxs? |
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| 07:12.39 | x86 | Rhino CB24-FXS |
| 07:12.47 | x86 | use a _bunch_ of those |
| 07:14.04 | x86 | I run 6 Asterisk servers at work right now... |
| 07:14.31 | x86 | in the middle of mass migration from our legacy PBX systems (some made in the early 1980's) to brand new Asterisk servers |
| 07:16.36 | x86 | using Polycom phones (mostly SoundPoint IP301, some IP501), Sangoma T1/E1 cards (mostly A102DX, some A104DX), Rhino FXS channel banks (our sales people all just have basic analog phones), and HP DL380 servers (dual redundant power, RAID-1 hard drives, mirrored hot-standby memory, two dual-core procs) |
| 07:16.38 | Pagautas | :) i'm still migrating from pbx make in 95' |
| 07:17.10 | x86 | when I'm done, we'll have around 12 brand new Asterisk servers |
| 07:17.21 | Pagautas | how much users? |
| 07:17.27 | x86 | ~200 |
| 07:17.27 | jbot | somebody said 200 was the standard price in Swedish cities. |
| 07:17.40 | Pagautas | and you need 12 servers? |
| 07:17.42 | x86 | but they are all in different offices |
| 07:18.10 | x86 | yeah, we put one server in each branch office, and we are going to use a few in the main headquarters |
| 07:18.36 | x86 | well actually, they just hired a bunch of new people for a brand new office, so something like 230 users now |
| 07:18.43 | Pagautas | how much max active calls |
| 07:18.44 | Pagautas | ? |
| 07:18.53 | x86 | depends on the branch |
| 07:19.00 | x86 | some branches only have 4 or 5 people ;) |
| 07:19.16 | Pagautas | i'm talking about all network |
| 07:19.23 | x86 | I have no idea yet |
| 07:19.39 | x86 | I can't get that information from our legacy PBX's that are still in some locations |
| 07:19.48 | Pagautas | we have about 70-100 max |
| 07:20.02 | x86 | that's not supported by Digium ;) |
| 07:20.07 | Pagautas | :) |
| 07:20.19 | x86 | Asterisk Business Edition only supports 40 maximum calls at a time |
| 07:20.59 | x86 | and we're an outbound call center... so most of my users are on the phone all the time |
| 07:22.26 | x86 | each sales person is required to have at least 5 hours of logged phone time per 8 hour work day |
| 07:22.55 | x86 | I would say ~180 of our ~230 users are sales people |
| 07:23.04 | x86 | if that gives you an idea ;) |
| 07:23.20 | x86 | so I'll have more active calls to support than you will ;) |
| 07:32.21 | Pagautas | :) |
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| 08:55.45 | Sweeper | any way do do an autodial when a polycom's handset is picked up, and disable the numpad? |
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| 09:35.02 | Sweeper | x86: please tell me you're not paying for asterisk BE..... |
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| 10:13.45 | lirakis | morning |
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| 12:23.55 | KaiHanari | if anyone is alive, im looking at getting a dual fxs. i can get a sipura 2000 or a linksys PAP2, both same priced. which would (anyone here) suggest? |
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| 12:33.36 | lirakis | KaiHanari: sipurra is supposed to be better... the pap2 is obsoleted by the pap2t i think too |
| 12:33.56 | KaiHanari | ah. the sipura also seems nicer for docs |
| 12:43.25 | tsurko | hello |
| 12:44.25 | tsurko | I'm trying to compile res_config_ldap.c. I've added it in the res dir in asterisk trunc source, it compiles, but when I run * I get this error: |
| 12:44.38 | tsurko | asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_ldap.so: undefined symbol: ldap_initialize |
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| 14:14.30 | SuPrSluG | hello all |
| 14:15.14 | SuPrSluG | i'm not getting ringing tones for people calling in. what's the config for that |
| 14:16.10 | lirakis | SuPrSluG: thats not a very clear description of the problem... your phones dont ring.. the callers dont get ring.. ? |
| 14:16.45 | SuPrSluG | phones ring caller hears no ringing |
| 14:17.05 | lirakis | SuPrSluG: are they using your pbx? ... are they calling from a cell phone.. the pstn what? |
| 14:18.07 | *** join/#asterisk _bobweever_ (n=_bobweev@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:18.33 | SuPrSluG | niether cell or regular land lines hear ringing. they get to my prompt and press the menu entry for me and it rings here fine. but they heat dead air |
| 14:18.55 | SuPrSluG | hear dead air |
| 14:19.44 | lirakis | SuPrSluG: its possibly a nat issue for signaling .. but i dont know what you have setup for you provider. |
| 14:20.13 | lirakis | SuPrSluG: are you using a sip provider .. or a tdm card |
| 14:20.22 | lirakis | SuPrSluG: (see how much you left out ) |
| 14:20.37 | SuPrSluG | sip provider 2 different so ? |
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| 14:21.00 | lirakis | SuPrSluG: .. what you just said .. makes no sense.. clarify please |
| 14:22.04 | SuPrSluG | 2 different providers so seem like an asterisk issue. i'm running it on an asus router |
| 14:23.05 | lirakis | SuPrSluG: so if its on the router .. is the ip nat'd or public? |
| 14:23.24 | lirakis | SuPrSluG: it sounds like some kind of nat issue... but i am not certain. |
| 14:23.28 | lirakis | ~sipnat |
| 14:23.29 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:24.42 | SuPrSluG | nat . the calls go through fine with audio both ways. just no ringing sound to the caller |
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| 14:26.01 | lirakis | SuPrSluG: there can be nat issues for signaling too.. check your ports |
| 14:27.27 | SuPrSluG | firewall rules here http://pastebin.ca/653512 |
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| 14:34.55 | SuPrSluG | lirakis:think i found the answer. I had asterisk just enten => answer() when i added Wait in front of it it was able to generate the ring tones |
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| 14:47.36 | lirakis | SuPrSluG: .. so it just answered the call immediately.. before they even heard a ring?.. okay.. thats a pretty.. uh incredibly low pdd that whoever is dialing in is getting |
| 14:52.24 | easimon | hi, i'm using a bristuff'd asterisk with zaphfc and a connected cordless isdn phone, and since an upgrade to 1.4 i cannot dial any numbers off-hook anymore. pri debug shows the digits coming in, but the corresponding WaitExten doesn't react. it used to work with asterisk 1.0. any clues on that? |
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| 14:58.29 | easimon | hm. 258 users, no one listening. :-/ |
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| 15:03.21 | pourriture | Am I correct that the Background() app tells Asterisk to expect the caller to dial some digit or series of digits and that the Waitexten() app tells Asterisk to keep listening for input for the specified number of seconds? |
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| 15:07.59 | JT_ | easimon: quantity is irrelevant ;) |
| 15:09.06 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 15:09.46 | easimon | JT_: seems like, yea. |
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| 15:11.42 | De_Mon | pourriture more or less |
| 15:12.16 | De_Mon | pourriture from the *CLI> show application background |
| 15:12.52 | De_Mon | why are applications case insensitive whereas functions are case senseitive? |
| 15:14.31 | pourriture | The *CLI> does should that the app background is running, but if I dial a digit I get no response , either on the phone or the CLI |
| 15:15.02 | pourriture | s/should/show |
| 15:16.33 | pourriture | It eventually times out and goes to t |
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| 15:20.48 | pourriture | it is as if it is not hearing me dial a digit at all |
| 15:22.16 | pourriture | I am using an X100P as proof of concept, so maybe it is so riddled with static that it can't figure out what I am pressing and it chooses to ignore me completely :) |
| 15:23.00 | creativx | easimon: lest not forget, tis saturday |
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| 15:29.21 | JT | heh |
| 15:29.28 | JT | it's sunday here now |
| 15:29.38 | JT | my colo was inaccessible for 20mins, how annoying |
| 15:31.46 | creativx | hehe |
| 15:31.46 | creativx | i see |
| 15:31.50 | creativx | my saturday is just starting |
| 15:32.04 | creativx | damn home office is keeping me from getting unsober |
| 15:32.47 | *** join/#asterisk WindBack (n=Administ@host141.190-31-69.telecom.net.ar) |
| 15:33.54 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 15:34.07 | JT_ | there it goes again |
| 15:34.17 | JT_ | 25% packet loss then complete loss of communication |
| 15:34.19 | JT_ | grr |
| 15:35.09 | WindBack | If I include in a context "A" another context "B" who include a context "C". How can include only in A the context B without the C ?? |
| 15:36.20 | *** join/#asterisk decobbb (n=bbradley@66.255.125.146) |
| 15:36.35 | decobbb | hello everyone |
| 15:38.29 | decobbb | anyone in here? |
| 15:38.44 | Nugget | WindBack: make a context "D". |
| 15:38.46 | [TK]D-Fender | WindBack, B should not include C then. You should make a D that would contain B & C, and an E that contains A & B |
| 15:40.11 | decobbb | i am able to comlete calls but am unable to hear any of the voice prompts |
| 15:40.17 | decobbb | complete |
| 15:40.22 | decobbb | any ideas |
| 15:40.43 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 15:41.09 | Nugget | decobbb: nat? |
| 15:42.32 | Nugget | with so little detail about your setup, for all we know the problem might be excessive ear wax. :) |
| 15:43.06 | WindBack | Nugget, [TK]D-Fender ok.. you are rigth |
| 15:43.11 | Nugget | is it sip? iax? pots? what kind of phone. what voice prompts? |
| 15:43.36 | [TK]D-Fender | is oit only the prompts tht you can't here, or ALL audio? |
| 15:44.26 | [TK]D-Fender | What is the airspeed velocity of an unladen swallow? |
| 15:45.03 | decobbb | sip |
| 15:45.12 | decobbb | the prompts for voicemail |
| 15:45.17 | decobbb | and the confrence center |
| 15:45.34 | decobbb | and parking calls |
| 15:45.54 | decobbb | im not using any nat |
| 15:46.23 | decobbb | i am using both soft phones and hard sip phones |
| 15:46.32 | decobbb | niether can get prompts |
| 15:46.41 | decobbb | voice prompts |
| 15:46.54 | [TK]D-Fender | decobbb, do you see CLI output of the call attempt? |
| 15:46.59 | decobbb | yep |
| 15:47.11 | decobbb | [Aug 11 11:08:53] WARNING[3920]: app_voicemail.c:6194 vm_authenticate: Couldn't read username |
| 15:47.13 | [TK]D-Fender | decobbb, check your firewall |
| 15:47.35 | [TK]D-Fender | decobbb, that error can mean that your entry is wrong or that DTMF isn't reaching * at all. |
| 15:47.59 | [TK]D-Fender | decobbb, and that isn't an audio problem. |
| 15:48.01 | lirakis | [TK]D-Fender: african ? or european? |
| 15:48.08 | [TK]D-Fender | lirakis, I don't know! |
| 15:48.13 | [TK]D-Fender | aaaaaaaarrrrrrrrrrggggggggggghhhhhhhhhhhhh! |
| 15:48.19 | lirakis | lol |
| 15:49.04 | decobbb | is there a specific file i should check? |
| 15:49.42 | [TK]D-Fender | decobbb, You should start being more specific about your problem(s), they seem to be multiplying |
| 15:49.52 | decobbb | wow |
| 15:49.55 | decobbb | ok |
| 15:50.03 | [TK]D-Fender | decobbb, You start with one thing and switch to something completly different |
| 15:50.19 | [TK]D-Fender | decobbb, can you HEAR it ask for your password? |
| 15:50.23 | decobbb | no |
| 15:50.36 | [TK]D-Fender | decobbb, pastebin yo iptables. |
| 15:50.37 | [TK]D-Fender | ~pb |
| 15:50.38 | jbot | A Pastebin is a place to paste your stuff 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 |
| 15:50.40 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 15:50.47 | [TK]D-Fender | your* |
| 15:51.06 | decobbb | i have iptables turned off |
| 15:52.47 | [TK]D-Fender | decobbb, pastebin your sip.conf, CLI output of the failed attempt at verbose 10 |
| 15:54.41 | lirakis | in Dial(SIP/telasip/2121234567) .. is telasip/2121234567 the "identifier" .. or is SIP/telasip the type and 2121234567 the identifier? |
| 15:55.25 | [TK]D-Fender | lirakis, telesip is the peer entry to use which will ID the ip/host, auth, etc. the /number will be the "to:" |
| 15:55.27 | De_Mon | SIP is the "resource", telasip is the user and 212 is data.. |
| 15:55.38 | *** join/#asterisk saftsack (n=oliver@p54A7BE6C.dip.t-dialin.net) |
| 15:55.54 | decobbb | ok i pasted in the output |
| 15:56.43 | lirakis | [TK]D-Fender: i am using an exec_dial() function of the phpagi class http://phpagi.sourceforge.net/phpagi2/docs/ ... i am trying to figure out how to pass the variables to the function |
| 15:56.56 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 15:57.05 | [TK]D-Fender | decobbb, LINK PLEASE |
| 15:57.13 | lirakis | [TK]D-Fender: it asks for type, identifier, timeout, and options |
| 15:57.16 | De_Mon | pastebin newbie |
| 15:57.27 | [TK]D-Fender | lirakis, not sure what you mean "pass variables".. this just calls app_dial |
| 15:57.30 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 15:57.55 | decobbb | the sip.conf is under decobbb |
| 15:58.13 | [TK]D-Fender | decobbb, wE DON'T KNOW WHICH SITE OF THE ten THAT WERE LISTED THAT YOU USED! |
| 15:58.22 | [TK]D-Fender | decobbb, paste the link! |
| 15:58.23 | decobbb | www.pastebin.com |
| 15:58.29 | lirakis | [TK]D-Fender: .. no.. exec_dial() is a member function of the phpagi class .. its not exec("Dial....") ... so it actually has a function definition that takes those variables... and i am trying to figure out what is what in my dial statement |
| 15:58.45 | decobbb | http://pastebin.com/m1cc3f805 |
| 15:58.48 | [TK]D-Fender | lirakis, not sure... |
| 15:59.03 | decobbb | http://pastebin.com/m104cbb5b |
| 15:59.05 | lirakis | [TK]D-Fender: i am sure.. underlying.. it does call dial.. but .. like i said.. it has its own function def. |
| 15:59.25 | [TK]D-Fender | decobbb, Sorry, you're running a GUI built setup. AsteriskNAOW is not supported here. |
| 15:59.50 | [TK]D-Fender | decobbb, And you only provided 1 part of what I asked for. |
| 16:00.11 | decobbb | well i programmed it by the .conf files |
| 16:00.19 | decobbb | not by the gui |
| 16:00.42 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 16:01.13 | *** join/#asterisk mutilator (i=WebChat@the.drinkproject.com) |
| 16:02.00 | [TK]D-Fender | decobbb, really... so where are your phones set up? |
| 16:02.14 | decobbb | users.conf and extensions.conf |
| 16:02.33 | decobbb | i have the gui there so that the helpdesk guy and add extensions |
| 16:02.42 | decobbb | can |
| 16:04.58 | decobbb | i posted up both things you asked for |
| 16:05.03 | decobbb | im not sure what you man e |
| 16:05.07 | decobbb | mean |
| 16:05.31 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 16:06.51 | *** join/#asterisk ManxPower (n=manxpowe@015-828-991.area5.spcsdns.net) |
| 16:08.08 | [TK]D-Fender | decobbb, how about the CLI output I asked for? Go replicate the problem and show us. |
| 16:08.25 | decobbb | it only gives 3 lnes |
| 16:08.28 | decobbb | lines |
| 16:08.47 | [TK]D-Fender | decobbb, When SIP debug enabled <----------- |
| 16:08.55 | decobbb | ok |
| 16:08.56 | [TK]D-Fender | with* |
| 16:09.35 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 16:10.01 | rhombus | Has or is anyone here using Mitel SIP phones with Asterisk? |
| 16:10.41 | lirakis | hrm... im trying to use the L option with Dial to limit the duration of the call.. no luck so far :\ |
| 16:11.06 | [TK]D-Fender | lirakis, pastebin... :) |
| 16:13.17 | *** join/#asterisk fholmes (n=fholmes@c-98-198-197-44.hsd1.tx.comcast.net) |
| 16:13.24 | decobbb | http://pastebin.com/m614a0717 |
| 16:13.30 | decobbb | thats the sip debug |
| 16:13.39 | *** join/#asterisk saftsack (n=oliver@84.167.235.70) |
| 16:13.55 | lirakis | [TK]D-Fender: .. i just set up a quick test extension .. to try it http://pastebin.com/d42317e8d |
| 16:14.19 | lirakis | <PROTECTED> |
| 16:15.35 | lirakis | <PROTECTED> |
| 16:16.03 | *** join/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net) |
| 16:16.09 | *** join/#asterisk RageMax (n=RageMax@128.2.178.9) |
| 16:16.36 | *** part/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net) |
| 16:16.39 | RageMax | I'm getting an error with the MP3Player app: app_mp3.c:121 timed_read: Poll timed out/errored out with 0 |
| 16:17.02 | RageMax | mpg123 is installed |
| 16:17.09 | [TK]D-Fender | lirakis, limit to 10s w/ 2s warning?! |
| 16:17.18 | RageMax | timing issues perhaps? |
| 16:17.49 | lirakis | [TK]D-Fender: its just for testing |
| 16:18.03 | [TK]D-Fender | lirakis, make your test a little more sane :) |
| 16:18.30 | [TK]D-Fender | decobbb, I'd suggest restricting your codecs to ULAW on all sides |
| 16:18.46 | decobbb | ok |
| 16:19.17 | decobbb | i will change it to allow only ulaw in the sip.conf |
| 16:20.33 | decobbb | same error |
| 16:23.35 | [TK]D-Fender | decobbb, astebin your iptables dump to be sure. Any other routers between * and your phones? |
| 16:23.57 | decobbb | just layer 3 switches |
| 16:24.03 | lirakis | [TK]D-Fender: trying 60s and 20s warning |
| 16:24.19 | decobbb | how do i do an iptables dump again |
| 16:24.23 | decobbb | been a while |
| 16:24.44 | [TK]D-Fender | lirakis, add the repeat parameter on |
| 16:24.55 | [TK]D-Fender | decobbb, "iptables --list" |
| 16:25.00 | decobbb | k |
| 16:25.06 | lirakis | [TK]D-Fender: ok |
| 16:26.05 | *** join/#asterisk RageMax (n=RageMax@128.2.178.9) |
| 16:26.48 | decobbb | http://pastebin.com/m27a5d3d1 |
| 16:26.58 | decobbb | iptables dump |
| 16:29.06 | [TK]D-Fender | decobbb, You have a secondary class-c net as well? |
| 16:29.22 | decobbb | i have 3 nets |
| 16:29.26 | decobbb | 300 users |
| 16:29.35 | decobbb | 7 sites around the world |
| 16:29.38 | [TK]D-Fender | decobbb, hrm. |
| 16:29.56 | decobbb | luckily im not live yet |
| 16:30.00 | decobbb | with * |
| 16:30.21 | decobbb | but i am suppose to test my pri links tonite |
| 16:31.19 | [TK]D-Fender | decobbb, ok, I'm a little dry on ideas right now... keep some output handy and maybes someone else will have some inspiration in a bit. |
| 16:31.25 | decobbb | k |
| 16:31.27 | decobbb | thnx |
| 16:31.36 | decobbb | its wierd |
| 16:31.41 | decobbb | one |
| 16:31.53 | decobbb | i had it all working before i put it in the server room |
| 16:32.42 | [TK]D-Fender | decobbb, just for a test, flush your iptables. |
| 16:33.03 | [TK]D-Fender | decobbb, not sure if I interpreted that all properly. |
| 16:35.33 | lirakis | [TK]D-Fender: yeah ... still no warning.. just hangs up |
| 16:39.39 | *** part/#asterisk ming_zym (n=ming_zym@124.254.55.239) |
| 16:47.33 | decobbb | what is the command to flush the iptables |
| 16:48.43 | [TK]D-Fender | .... |
| 16:48.46 | [TK]D-Fender | iptables --flush |
| 16:49.09 | decobbb | thnx |
| 16:49.47 | decobbb | still nothing |
| 16:49.58 | decobbb | thnx though |
| 16:51.03 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 16:51.30 | *** join/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net) |
| 16:51.53 | lirakis | [TK]D-Fender: is this the right way to call an actual EXEC Dial from agi (not phpagi) |
| 16:52.04 | *** part/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net) |
| 16:52.14 | lirakis | "DIAL SIP/telasip/2123810963||L(60000,30000,5000)" |
| 16:52.37 | lirakis | i wasnt sure if the || was required to pass null for the timeout |
| 16:52.52 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 16:54.30 | elixer | i think you want something like this... |
| 16:54.40 | elixer | EXEC Dial "SIP/telasip/2123810963||L(60000,30000,5000)" |
| 16:54.52 | elixer | but i may be full of it |
| 16:55.18 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 16:56.30 | lirakis | elixer: .. i know the EXEC is supposed to be infront of it.. i was just curious about the || .. i checked voip-info.. it is the seperator for arguments when calling EXEC |
| 16:56.42 | lirakis | .. okay.. i need to get some food.. |
| 16:56.48 | elixer | well |
| 16:56.48 | lirakis | bbl |
| 16:56.52 | *** part/#asterisk lirakis (n=eric@cpe-68-175-38-65.nyc.res.rr.com) |
| 16:56.53 | elixer | i meant that you wanted to quote the args |
| 16:56.57 | elixer | nice |
| 16:58.09 | elixer | i don't get why he doesn't just TRY it instead of asking about every little detail every 30 seconds |
| 16:58.12 | elixer | whatever. |
| 16:59.28 | *** join/#asterisk Garwezys (n=gwz@ctv-86-100-202-48.ip.rygveda.lt) |
| 17:04.15 | lmadsen | elixer: amen |
| 17:12.09 | *** join/#asterisk saftsack (n=oliver@p54A7C987.dip.t-dialin.net) |
| 17:13.29 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
| 17:14.00 | AllanLima | how to install asterisk on cpanel? is possible? |
| 17:14.22 | elixer | what is cpanel? |
| 17:14.51 | AllanLima | hosting manager |
| 17:14.56 | elixer | ah. |
| 17:15.33 | elixer | doesn't look like it -> http://forums.digium.com/viewtopic.php?t=5407 |
| 17:16.42 | AllanLima | hum |
| 17:16.46 | AllanLima | let me see |
| 17:19.31 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 17:20.16 | bbryant | AllanLima, you would have to have shell access (there's a shell client in cpanel) |
| 17:22.51 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 17:23.42 | AllanLima | bbryant i have a dedicated server, root access into server |
| 17:24.33 | *** join/#asterisk ccesario_ (n=ccesario@201-0-53-62.dsl.telesp.net.br) |
| 17:26.56 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 17:28.21 | AllanLima | i want to install this on cpanel, but how? :( |
| 17:30.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:32.51 | JT_ | AllanLima: you can't. why would you want to? |
| 17:33.46 | AllanLima | JT_ to use, pc to pc |
| 17:34.02 | AllanLima | :( |
| 17:34.33 | AllanLima | why i can't? |
| 17:35.31 | [TK]D-Fender | "to use, pc to pc" <- huh?! |
| 17:35.39 | nick125 | You can install it on your server, but, not with cpanel. IT would be completely separate from cPanel. |
| 17:37.53 | AllanLima | nick125 how to make this? |
| 17:37.57 | *** join/#asterisk Op3r (n=Op3r@121.97.147.190) |
| 17:39.33 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 17:42.33 | nick125 | AllanLima: Well, first you'll need to install asterisk, and then you'll need to configure it, both which are very many guides online for. |
| 17:43.10 | nick125 | cPanel doesn't really matter in this case. |
| 17:43.41 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 17:44.25 | AllanLima | :\ |
| 17:44.29 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 17:44.29 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 17:45.54 | AllanLima | but i need cpanel, used for hosting business |
| 17:46.12 | elixer | no one is suggesting you get rid of it |
| 17:46.53 | elixer | just saying that installing asterisk on that box has nothing to do with cpanel |
| 17:47.33 | AllanLima | is possible install on vps? |
| 17:47.58 | elixer | yes, of course. (of course i have no idea what vps is) |
| 17:49.05 | AllanLima | vps = virtual private server or vds = virtual dedicated server |
| 17:49.42 | elixer | ah |
| 17:49.45 | elixer | yeah, sure. |
| 17:49.47 | elixer | heh |
| 17:49.49 | elixer | i have no clue |
| 17:49.57 | AllanLima | hehe |
| 17:50.05 | elixer | i just buy servers, install asterisk on them, and move on with life |
| 17:50.26 | elixer | "managed" to me is just a synonym for "piss you off" |
| 17:50.41 | AllanLima | hum |
| 17:51.34 | AllanLima | buy where? |
| 17:51.36 | elixer | dell |
| 17:51.40 | elixer | usually |
| 17:51.47 | elixer | i'm not helping you, i know |
| 17:51.59 | elixer | i just have no idea how to get you up and running |
| 17:54.11 | AllanLima | :) |
| 17:54.23 | elixer | its better than leading you astray |
| 17:54.24 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:56.07 | *** join/#asterisk CunningPike (n=arodgers@154.5.233.251) |
| 17:56.38 | AllanLima | is possible make this on remote server, with root ssh access, sure? |
| 17:56.43 | [TK]D-Fender | AllanLima, Unlike the rest of what cpanel manages, Asterisk possibilities are MASSIVE and the implementation style too huge to suitably limit to some schmuck interface like that. Telcos vary by region, thent here are hardware concerns, etc, GUI integration, etc. Someone would ahve to go through a lot of trouble to try and make something usable for that, and then you probably would hate how it works because of those limitations. |
| 17:56.57 | [TK]D-Fender | AllanLima, Asterisk should be best managed all by itself. |
| 17:57.51 | [TK]D-Fender | AllanLima, Of course you should know this already if you ever actually WORKED with Asterisk.... |
| 17:58.58 | AllanLima | hmmm |
| 17:59.27 | AllanLima | i want use for "pc to pc" |
| 17:59.41 | [TK]D-Fender | AllanLima, And what does THAT mean? |
| 18:01.18 | AllanLima | what "that" mean? |
| 18:02.10 | [TK]D-Fender | AllanLima, clarify what you mean by "pc to pc" |
| 18:03.38 | AllanLima | simple voip, no external |
| 18:04.28 | AllanLima | only internet calls, from pc to pc |
| 18:04.29 | [TK]D-Fender | AllanLima, Go install and learn *. |
| 18:04.30 | [TK]D-Fender | ~book |
| 18:04.31 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:04.35 | [TK]D-Fender | ~wiki |
| 18:04.36 | AllanLima | pc = personal computer |
| 18:04.43 | [TK]D-Fender | ~wikis |
| 18:04.44 | jbot | [wikis] http://www.voip-info.org |
| 18:04.44 | *** join/#asterisk saftsack (n=oliver@p54A7CC9B.dip.t-dialin.net) |
| 18:05.31 | AllanLima | understand? |
| 18:08.43 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 18:16.10 | tzafrir_home | AllanLima, generally yes. But what type of vps is it? What technology does it use? |
| 18:16.44 | *** join/#asterisk shareenergy (i=shareene@62.169.96.30.rev.optimus.pt) |
| 18:17.04 | shareenergy | Hello, i need some help with weird linksys 941 phone |
| 18:17.23 | shareenergy | i have in 1 phone the version of firmware 4.1.8 and it works with no problem |
| 18:17.51 | shareenergy | on other 8 phones i have the version 4.1.15 and everyone complains about coutings of the line |
| 18:18.11 | shareenergy | i've changed positions of the phones but i can't get the solution for this |
| 18:18.20 | shareenergy | can anyone help me? |
| 18:18.36 | AllanLima | tzafrir_home virtuozzo |
| 18:18.59 | shareenergy | is there any place to get the old firmware? |
| 18:19.31 | tzafrir_home | virtuozzo (or openvz - the same technology) is quite nice - very little overhead |
| 18:19.56 | tzafrir_home | However you can't really install your own kernel modules, if you need azptel timing |
| 18:21.06 | tzafrir_home | linux-vserver is also a similar technology in that sence (same kernel). Maybe this will help you in searches |
| 18:25.24 | *** join/#asterisk Weezey (n=ohno@206.210.109.232) |
| 18:26.57 | Weezey | I've finally got this Google Talk stuff working, i needed to do bindaddr= to make it work... Now, I want to be able to send messages to the asterisk box and interpret them as commands, I can't seem to find a way to have the commands received by * though, just how to send them out. Any clues? |
| 18:28.30 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
| 18:28.53 | [TK]D-Fender | Weezey, like? |
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| 18:35.53 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 18:37.03 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 18:37.18 | lirakis | ... the battle continues ... |
| 18:54.00 | lmadsen | just finished reading all of QC2 of TFoT2 |
| 18:54.11 | lmadsen | lets hope I caught most of the issues ;) |
| 18:54.43 | Strom_C | lmadsen: that chapter on cheesecake still mentions potatoes for some reason |
| 18:55.32 | lirakis | lmadsen: sweet! i cant wait to get my hands on it |
| 18:55.34 | lmadsen | mmmm... I love potato cheesecake |
| 18:57.52 | *** join/#asterisk ikey (i=ikey@219.64.67.29) |
| 19:08.03 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 19:08.04 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 19:09.13 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 19:09.29 | ikey | kk |
| 19:22.31 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 19:25.11 | lirakis | if i want to forkCDR and have the new cdr have a different src, dst etc. can i do that?? i dont see the function taking any params? |
| 19:36.18 | codefreeze | lirakis: forkCDR just attaches a new cdr to the end of the cdr list for the channel; then marks all the cdrs before it as "locked". Any cdr ops you do after that, work on the last cdr in the list |
| 19:36.43 | codefreeze | That CDR, is basically a copy of the cdr in front of it. |
| 19:37.16 | codefreeze | (start times are mod'd;) |
| 19:37.37 | codefreeze | and you can change the acctcode, userfield, vars, etc, just like normal |
| 19:38.06 | lirakis | codefreeze: yeah i know... i am home brewing a calling card app for fun... and im trying to figure how to best create a new CDR for the "actual call" the person would make |
| 19:38.47 | lirakis | .. the only way i can think is a "EXEC GOTO NXXXXXX|1" ... but that would return control to the dialplan from my AGI script |
| 19:38.50 | lirakis | .. and i dont want that |
| 19:38.50 | codefreeze | 1.4, or 1.2 |
| 19:38.52 | codefreeze | ? |
| 19:38.53 | lirakis | .. 1.2 |
| 19:39.14 | lirakis | .. i wish you could just set the stupid CDR variables.. like src,dst etc... but you can |
| 19:39.34 | lirakis | codefreeze: .. well its for no purpose other than to learn... so (shrug) |
| 19:40.08 | lmadsen | lirakis: yer doing stuff for fun on 1.2? get with the times :) |
| 19:40.08 | codefreeze | lirakis: you might like the "new approach" I'll be taking in trunk soon. |
| 19:41.11 | lirakis | lmadsen: .. well i use it for my everyday pbx too .. sooo |
| 19:41.46 | lirakis | lmadsen: i tried a 1.4 upgrade about 6 mo back when i had a hard drive die.. but it was not working out for me |
| 19:42.34 | codefreeze | lirakis: just keep in mind, that cdrs in 1.2 are ok, as long as no parking, or xferring occur. |
| 19:44.47 | lirakis | codefreeze: .. well .. i do "transfer" .. in that they call in .. and they dail out through to another number... and thats what i want im my cdr .. not thier # and my DID |
| 19:45.15 | codefreeze | lirakis: that's not an xfer situation, so you should be ok |
| 19:46.00 | codefreeze | xfers are where someone hits "#" and a number during the call (that's just one way, of course, there are others) |
| 19:47.32 | lirakis | codefreeze: .. okay.. at any rate.. im not sure how to get the cdr right without leaving AGI |
| 19:48.02 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 19:48.57 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 19:50.23 | kombi | does running mysql on the same box eat away overly many resources needed by *? should one rather run sqlite? |
| 19:51.18 | lirakis | kombi: i dont believe so .. i run mysql on the same machine (p3 750mhz) and i have tested up to 150 concurrent calls with media (not transcoded) strait ulaw. |
| 19:51.45 | kombi | lirakis: sounds convincing! |
| 19:51.55 | lirakis | kombi: maybe if you are running on an embedded appliance .. like a wrt54g or some thing |
| 19:52.47 | kombi | lirakis: actually, the box has some muscle (amd dual core, 1 Gig), so I think it should be fine then |
| 19:53.24 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:53.41 | kombi | wrt54g, is that comparable to soekris? |
| 19:54.03 | kombi | oh, Linksys, ok.. |
| 19:54.50 | lirakis | kombi: yeah soekris > wrt54g |
| 19:55.21 | lirakis | kombi: the early hw rev. wrt's are pretty good though.. with an SD hack.. they are sweet little <$40 units |
| 19:55.46 | kombi | no way to connect wrt54g straight to pstn though, is there? |
| 19:56.51 | LoneShadow | where can I find app_flite ? |
| 19:56.59 | LoneShadow | dont want to install trixbox |
| 19:57.18 | lirakis | i wonder if i can use set extension... hrmm /me thinking |
| 19:58.09 | LoneShadow | wrt54g is good for simple asterisk setup as long as you dont do transcoding |
| 19:58.31 | LoneShadow | I use it since spa3k dosnt support gsm |
| 19:59.48 | *** join/#asterisk saftsack (n=saftsack@pD9E07726.dip.t-dialin.net) |
| 20:03.04 | *** join/#asterisk decobbb (n=bbradley@66.255.125.146) |
| 20:05.03 | tzafrir_laptop | LoneShadow, what is it used for? |
| 20:05.14 | LoneShadow | flite ? |
| 20:05.28 | LoneShadow | which wrt54g or app_flite ? |
| 20:05.57 | LoneShadow | flite is used for text to speech |
| 20:07.08 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-200.shs.nyc.wayport.net) |
| 20:07.23 | LoneShadow | tzafrir_laptop: ? |
| 20:08.20 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 20:09.24 | tzafrir_laptop | LoneShadow, what distro di you use? |
| 20:09.30 | LoneShadow | ubuntu |
| 20:09.44 | LoneShadow | trying to extract an rpm file from nerdsvittle |
| 20:10.15 | decobbb | got any debugging gurus out there? |
| 20:10.37 | decobbb | im not able to hear any of mye * voice menus |
| 20:10.51 | tzafrir_laptop | flite itself is a package, at least in Debian |
| 20:10.53 | decobbb | yet i can make calls |
| 20:11.03 | tzafrir_laptop | Not sure about app_flite |
| 20:11.03 | LoneShadow | tzafrir_laptop: yea I installed that |
| 20:11.05 | lirakis | so.. when i use the L option for dial... it cuts the call at the max ms, but it does not give any warning.. i can not figure out why .. i am passing the parameters for warning and announce freq. |
| 20:11.19 | LoneShadow | tzafrir_laptop: but thats a generic package, for asterisk to use it, we need the app_flite |
| 20:11.25 | LoneShadow | just like app_festival |
| 20:11.29 | decobbb | [Aug 11 16:05:11] WARNING[4868]: app_voicemail.c:6194 vm_authenticate: Couldn't read username |
| 20:11.35 | decobbb | thats the error |
| 20:11.41 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 20:14.26 | foo | What's the one way to know whether or not my asterisk system is online / working or not? eg. something to monitor it? I'm thinking show sip connections needs to always show the 2 connections ... and if not, then it's not working |
| 20:15.10 | tzafrir_laptop | http://dialogpalette.sourceforge.net/download.html |
| 20:16.09 | LoneShadow | thanks :) |
| 20:16.29 | tzafrir_laptop | Look a bit old |
| 20:18.46 | tzafrir_laptop | it won't build with asterisk 1.4, I guess |
| 20:20.33 | tzafrir_laptop | http://yum.trixbox.org/centos/5/devel/RPMS/repodata/repoview/asterisk-flite-0-0.5-1.html |
| 20:20.55 | tzafrir_laptop | Lists verson dialogpallete as the URL, but has a newer version |
| 20:20.58 | tzafrir_laptop | strange |
| 20:22.00 | tzafrir_laptop | http://yum.trixbox.org/centos/5/devel/SRPMS/repodata/repoview/asterisk-flite-0-0.5-1.html - the srpm |
| 20:24.12 | elixer | ok, now i have a question. i am running 1.4.8 (will upgrade to 1.4.10.1 relatively soon, but no matter) |
| 20:24.48 | elixer | i have about 10 hard phones that are always registered, and around 90 soft phones that are registered when users login to their workstations |
| 20:24.51 | elixer | all SIP |
| 20:25.24 | *** join/#asterisk adker (n=chatzill@74-33-216-174.br1.glv.ny.frontiernet.net) |
| 20:25.28 | elixer | when a user dials one of the soft phones, say 1234 is the extension, it should try to dial them, and if they are not there, go to their VM |
| 20:25.54 | elixer | if a user dials a non-existant extension, say 9999, i want them to get a "we're sorry, the number cannot be completed as dialed" |
| 20:26.18 | elixer | but i'm in a position (i think) where i can't tell if a given extension is non-existant vs. currently unregister |
| 20:26.21 | elixer | unregistered |
| 20:26.32 | elixer | does that make any sense whatsoever |
| 20:26.32 | elixer | ? |
| 20:27.24 | elixer | like, if the user hasn't logged into their workstation, the dial will just fail with a "no route to host" |
| 20:28.20 | elixer | i guess my question is "is there a way to determine (before dialing) if a given SIP registration exists, and if so, if that phone is currently registered" |
| 20:29.25 | lirakis | elixer: why dont you just react to the responce code given from dial? |
| 20:30.16 | elixer | because for either case, its always CHANUNAVIL |
| 20:30.59 | elixer | either there is no such SIP account in sip.conf or the soft phone hasn't registered (because the user's machine is off or something) |
| 20:31.10 | elixer | there is no way that i can tell to differentiate |
| 20:31.37 | lirakis | elixer: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail |
| 20:31.53 | elixer | tried that too |
| 20:32.14 | lirakis | {AVAILSTATUS} from that will get what you want ... |
| 20:32.20 | elixer | i've tried that |
| 20:32.23 | lirakis | it waill tell you if it is an invalid device |
| 20:32.30 | elixer | i will try it again though, just for you |
| 20:32.33 | elixer | back momentarily |
| 20:34.00 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:34.03 | [TK]D-Fender | elixer, you can tell if you stop using pattern matches for your internal extensions and hard-number them. |
| 20:34.10 | decobbb | [Aug 11 16:05:11] WARNING[4868]: app_voicemail.c:6194 vm_authenticate: Couldn't read username |
| 20:34.15 | lirakis | foo: nagios |
| 20:34.19 | [TK]D-Fender | decobbb, taht is NOT an audio problem. |
| 20:34.27 | decobbb | anybody know how to solve that one for not hearing voice prompts |
| 20:34.28 | _ShrikE | elixer: and then have an s extension for failures. |
| 20:34.28 | [TK]D-Fender | decobbb, we covered this already |
| 20:34.32 | elixer | [TK]D-Fender: ah. not a bad plan. |
| 20:34.41 | decobbb | u told me to check late |
| 20:34.47 | decobbb | so i am |
| 20:35.09 | [TK]D-Fender | decobbb, Yes, but I didn't say to continue barking upt he wrong tree for things I DID confirm for you. |
| 20:35.19 | decobbb | sheesh |
| 20:35.35 | decobbb | what tree would u suggest i bark up then ? |
| 20:35.49 | foo | lirakis: Thanks, but that's not really an answer :) Yes, I'm going to use nagios, but I need to know what to monitor ;) |
| 20:39.04 | lirakis | foo: .. uhh.. it is an answer.. you can write a plugin for nagios that will check sip registrations using the asterisk manager, or you could check channels.. all kinds of stuff |
| 20:40.40 | *** part/#asterisk Garwezys (n=gwz@ctv-86-100-202-48.ip.rygveda.lt) |
| 20:40.50 | lirakis | switching to laptop |
| 20:40.52 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 20:42.30 | elixer | [TK]D-Fender: i know you said to use numbered extensions, but could you check this out -> http://pastebin.com/d1f6cb372 |
| 20:42.47 | elixer | [TK]D-Fender: seems like i should get something different from both of those calls to ChanIsAvail() |
| 20:45.20 | elixer | [TK]D-Fender: and getting 3? that's BUSY, isn't it? |
| 20:50.01 | [TK]D-Fender | elixer, You've misunderstood. THIS is the problem -> exten => _XXXX,1,Macro(dial-simple) |
| 20:50.06 | elixer | no |
| 20:50.09 | elixer | i understood that |
| 20:50.10 | elixer | :) |
| 20:50.10 | [TK]D-Fender | elixer, XXXX can be one of 1000 valiues |
| 20:50.19 | elixer | hold on a sec before you go off on a tangent |
| 20:50.24 | elixer | i know what you were saying |
| 20:50.34 | elixer | if i do what you suggest that it would work |
| 20:50.43 | elixer | i guess i'm asking if my logic is off or something. |
| 20:51.04 | elixer | shouldn't ChanIsAvail() return something different for those 2 args? |
| 20:51.05 | [TK]D-Fender | elixer, I never suggested using ChanIsAvail. |
| 20:51.12 | elixer | sigh |
| 20:51.13 | elixer | i know |
| 20:51.20 | elixer | pretend i am asking a question |
| 20:51.23 | elixer | heh |
| 20:51.58 | elixer | shouldn't ChanIsAvail() return 2 different if being passed an existing sip peer one time, and a non-existant sip peer the second time? |
| 20:52.06 | [TK]D-Fender | elixer, no idea. |
| 20:52.09 | elixer | gotcha |
| 20:52.48 | elixer | i'll use the hard-numbers |
| 20:52.58 | elixer | just thought using pattern matching would make maintenance a lot easier |
| 20:57.38 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 20:58.07 | _ShrikE | elixer: You want it so that if you have extensions 1200 - 1300 assigned to phones you want them to go to vm is the phone is unavailable, if anything else is dialed you indicate the number is invalid? |
| 20:58.21 | _ShrikE | for example. |
| 20:58.22 | elixer | correct |
| 20:59.03 | *** join/#asterisk polerin (n=erin@c-71-228-222-87.hsd1.tn.comcast.net) |
| 20:59.06 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 20:59.54 | _ShrikE | So why not have exten => _1[23]XX,1,Macro(dial-simple) |
| 21:00.15 | _ShrikE | and exten => s,1,Playback(some invalid statement) |
| 21:00.22 | _ShrikE | in your from-internal context |
| 21:00.45 | elixer | also not a bad idea |
| 21:00.58 | _ShrikE | who cares what the registration status is in this case |
| 21:01.07 | elixer | well |
| 21:01.19 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 21:01.24 | elixer | i don't necessarily have all registrations defined for 1200-1300 |
| 21:01.37 | elixer | it might be 1200-1300 with the exception of 1234 and 1257 |
| 21:01.57 | elixer | we do sales and have a lot of turnover and we don't reuse extensions for a long time |
| 21:02.05 | *** join/#asterisk nirz (n=nirz@79.178.40.81) |
| 21:02.30 | _ShrikE | that could make your life hard. |
| 21:03.19 | _ShrikE | Like [TK]D-Fender said, just use static extensions then. |
| 21:04.12 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 21:06.31 | elixer | ahh... got it |
| 21:06.58 | *** join/#asterisk _Krieger_ (n=warsword@91.102.176.47) |
| 21:07.03 | elixer | exten => s,n,GotoIf($["${SIPPEER(CHX_1445:status)}" = ""] ? 10 : 5) |
| 21:07.23 | elixer | if its not a valid sip peer, SIPPEER returns "" |
| 21:07.41 | elixer | so i can happily use my patterns |
| 21:07.48 | elixer | hooray |
| 21:07.48 | elixer | :) |
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| 21:18.10 | *** part/#asterisk zerohalo (n=zeroHalo@h-74-2-90-66.cmbrmaor.covad.net) |
| 21:25.20 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 21:26.59 | Daviey | How could i go about running two asterisk servers NAT'd? |
| 21:27.03 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 21:27.23 | Daviey | Using SIP / IAX on a LAN - obv. i can't forward the ports to two hosts |
| 21:28.43 | drako | Daviey, using alternatives ports |
| 21:30.27 | Daviey | drako: cool.. how can i go about doing that? |
| 21:30.42 | drako | change ports on sip.conf and iax.conf |
| 21:30.48 | drako | on one server |
| 21:30.57 | drako | and redirect ports to others |
| 21:31.17 | Daviey | thanks! |
| 21:31.19 | foo | What's the one way to know whether or not my asterisk system is online / working or not? eg. something to monitor it? I'm thinking show sip connections needs to always show the 2 connections ... and if not, then it's not working. I'm going to use nagios and write a custom plugin, I just need to know what to have the plugin check for |
| 21:31.45 | elixer | asterisk -rx "show sip connections" |
| 21:31.46 | elixer | ? |
| 21:32.44 | foo | ah, sip show registry is what I meant |
| 21:32.56 | foo | Hm, elixer, so, is that one of the best ways of determining that my PBX is up and running? |
| 21:33.27 | elixer | well its probably not 100%, but its pretty damn close |
| 21:34.00 | foo | elixer: Yeah, I can have it check network / asterisk PID and stuff... but, if both of those don't exist, then sip show registry won't work properly |
| 21:34.03 | foo | elixer: ok, I'll do that. Thanks |
| 21:35.01 | drako | foo, core show uptime |
| 21:35.42 | foo | drako: hm, if it wasn't up, then there wouldn't be a SIP connection... but, that's good to include nonetheless |
| 21:36.08 | drako | sip show peers? |
| 21:37.19 | foo | drako: oh, I wasn't aware of that one. Hmm |
| 21:37.34 | foo | Ah, cool, that shows if there are phones connected |
| 21:38.10 | [TK]D-Fender | Daviey, Read this |
| 21:38.12 | [TK]D-Fender | ~sipnat |
| 21:38.13 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:39.35 | foo | drako: thanks |
| 21:42.08 | drako | np |
| 21:59.35 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 22:01.14 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 22:05.33 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:13.39 | elixer | if you have an underscore in your extension but its not the first character will it be interpreted literally? or will it turn on pattern matching? |
| 22:17.09 | [TK]D-Fender | elixer, literal |
| 22:17.17 | elixer | thanks |
| 22:27.04 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 22:30.04 | *** join/#asterisk tomcontr3 (n=tomcontr@248-76-246-201.adsl.terra.cl) |
| 22:30.29 | tomcontr3 | hi, does anyone knows where to find the modules app_rxfax.c and app_txfax.c for FAX support? |
| 22:30.53 | _ShrikE | soft-switch.org |
| 22:31.09 | tomcontr3 | I couldnt find those there |
| 22:31.15 | _ShrikE | they are part of spandsp |
| 22:31.42 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-186-68.dsl.irvnca.pacbell.net) |
| 22:31.51 | BSD_Tech | ok need input |
| 22:32.04 | BSD_Tech | brb |
| 22:32.05 | tomcontr3 | yeah, but there is no download |
| 22:32.26 | tomcontr3 | I remember that I download it once a long time ago from there |
| 22:32.48 | _ShrikE | look harder... http://www.soft-switch.org/downloads/spandsp/ |
| 22:32.48 | BSD_Tech | have a issue with call parking |
| 22:33.11 | tomcontr3 | I can only see the spandsp |
| 22:33.16 | tomcontr3 | but not the modules |
| 22:33.17 | BSD_Tech | do they plan to make call parking use context |
| 22:33.42 | BSD_Tech | so that with multi client boxes the parking does not over lap |
| 22:34.21 | tomcontr3 | also I need the apps_makefile.patch which is not there... is there any other alternative mirror that has this files? |
| 22:34.35 | _ShrikE | tomcontr3: What version of * |
| 22:34.50 | tomcontr3 | las branch of 1.2 |
| 22:34.52 | tomcontr3 | lat |
| 22:34.54 | tomcontr3 | last |
| 22:35.00 | _ShrikE | http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/ |
| 22:35.17 | tomcontr3 | thanks a lot!! |
| 22:35.23 | _ShrikE | np |
| 22:36.52 | engrxyz | i am connecting my sip phone to asterisk and i got this error "SIP/2.0 481 Subscription Does Not Exist". any tips is appreciated? |
| 22:38.09 | lirakis | <PROTECTED> |
| 22:38.37 | lirakis | engrxyz: what does your cli show? |
| 22:38.56 | lirakis | engrxyz: is this .. on registration, or when you try to make a call? |
| 22:39.07 | engrxyz | i just debug it and it got those. on registration of my sip phone |
| 22:39.37 | engrxyz | what does it mean then by 481 and how to fix it? |
| 22:40.36 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:41.51 | lirakis | engrxyz: you are being far too vague... is it defined it sip.conf? ... what does your cli show? ... use pastebin and give us a better picture of your setup ... saying "its broken" .. doesnt allow anyone to help you |
| 22:42.37 | elixer | another question, if this is immediately obvious, please answer. my hints don't seem to work when my SIP accounts start with alphabetic characters |
| 22:42.50 | elixer | doing this: |
| 22:43.00 | elixer | exten => 1234,hint,SIP/1234 |
| 22:43.13 | elixer | (with a user [1234] in sip.conf works fine) |
| 22:43.15 | elixer | doing this: |
| 22:43.27 | elixer | exten => FOO1234,hint,SIP/FOO1234 |
| 22:43.39 | elixer | (with a user [FOO1234] in sip.conf) |
| 22:43.40 | elixer | does not work. |
| 22:43.48 | elixer | meaning i don't get VM notifications. |
| 22:44.44 | *** join/#asterisk logicwrath (n=some@c-68-41-24-98.hsd1.mi.comcast.net) |
| 22:44.46 | lirakis | elixer: .. sorry :\ .. i cant think of anything right now |
| 22:45.19 | lirakis | elixer: you are hinting for VM .. not blf? |
| 22:47.21 | apardo | somebody with imapstorage running ? |
| 22:47.56 | apardo | i always receive Mailbox does not exist with Cyrus and all posible delimiters |
| 22:50.19 | elixer | lirakis: yeah |
| 22:50.37 | [TK]D-Fender | elixer, Hints have nothing to do with VM |
| 22:50.49 | elixer | ah |
| 22:50.52 | elixer | well |
| 22:51.04 | elixer | my soft phones display an envelope when i have VM waiting |
| 22:51.08 | [TK]D-Fender | elixer, and don't think that the exten for your hint has anything to do with the device that comes after. |
| 22:51.21 | [TK]D-Fender | elixer, and.....? |
| 22:51.54 | elixer | when i have a SIP peer 1234 and leave VM i get the envelope |
| 22:52.05 | elixer | when i have a SIP peer FOO1234 and leave VM i get no envelope |
| 22:52.12 | elixer | without any hinting |
| 22:52.16 | elixer | i get no envelope |
| 22:52.22 | lirakis | elixer: you get VM subscription in sip.conf through subscription |
| 22:52.22 | elixer | in either case |
| 22:52.43 | lirakis | elixer: check your foo1234 extension in sip.conf and see what it says for mailbox |
| 22:52.56 | elixer | it says FOO1234@default |
| 22:53.20 | lirakis | elixer: is that the context it's voicemail is in? |
| 22:53.33 | [TK]D-Fender | elixer, remove all association between "hints" and "voicemail" from your head, permanently. |
| 22:53.39 | elixer | yes sir |
| 22:53.57 | [TK]D-Fender | elixer, and last I checked the VM system does not like alpha in your box definitions. |
| 22:54.08 | elixer | well that explains it |
| 22:54.14 | elixer | thanks for your kind words |
| 22:54.26 | [TK]D-Fender | elixer, we try :) |
| 22:55.23 | elixer | and by "we try :)" you mean the exact polar opposite, yeah? |
| 22:56.00 | [TK]D-Fender | elixer, No, its a polar opposite to the OUTCOME ;) |
| 22:56.20 | [TK]D-Fender | elixer, either way, you've got your answer. Enjoy! |
| 22:57.02 | elixer | so to my next question |
| 22:57.04 | elixer | :) |
| 22:57.14 | elixer | eh. nevermind. |
| 23:01.18 | [TK]D-Fender | ah err... um... *grumbe*... huh? :) |
| 23:02.02 | elixer | i'd rather figure it out myself. don't think i need for someone to bend me over everytime i ask a question. |
| 23:07.29 | lirakis | elixer: .. as far as guru's go .. and ESPECIALLY guru's on IRC .. iv'e found [TK]D-Fender to be very reasonable/amiable. |
| 23:08.08 | elixer | oh, i definitely appreciate the knowledge transfer |
| 23:08.14 | elixer | i think the delivery sucks |
| 23:08.24 | elixer | but i guess you gotta take the bad with the good |
| 23:09.08 | lirakis | elixer: i disagree.. if you give an inch.. people take a mile... and some times you have to be blunt when people start using you as a reference manual. |
| 23:10.16 | lirakis | elixer: i believe d-fender is pretty selfless.. and often tries only to get people to think for themselves... (shrug) ... |
| 23:10.42 | *** join/#asterisk exvito (n=exvito@89-180-79-126.net.novis.pt) |
| 23:11.32 | lirakis | i had to go out and get a different chair... i have been using my kitchen chairs at my computer desk since i got a new apt... my back was just killing me though.. lol .. i had to upgrade |
| 23:11.42 | lirakis | soooooo much better now... |
| 23:11.43 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:11.49 | *** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com) |
| 23:12.34 | lirakis | exvito: ... |
| 23:12.49 | *** join/#asterisk Jan-pp (n=Jan@frnk-590c421a.pool.einsundeins.de) |
| 23:14.03 | BSD_Tech | hey TK |
| 23:14.39 | BSD_Tech | is there a plan to make callparking work with context for a multicompany pbx |
| 23:14.46 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 23:14.57 | Jan-pp | hi, i need help with an asterisk set up. I have an asterisk on a local linux box which connects to the internet by ways of a nat router. this nat router also has an integrated voip function, so it uses some important ports and causes additional trouble. i want asterisk to log into a sipgate.de account and be able to receive calls (no outgoing calls necessary) - for the beginning, a message playback should be set up |
| 23:15.11 | Jan-pp | can anyone help me on this please, i am getting registration timeouts |
| 23:16.07 | lirakis | Jan-pp: |
| 23:16.09 | lirakis | ~sipnat |
| 23:16.09 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:17.08 | lirakis | Jan-pp: also .. turn off virtually ANY crap related to voip on the router ... if its "QOS" maybe leave it.. but it probably wont make things better enough for you to try and figure out what to leave on/off |
| 23:18.18 | Jan-pp | lirakis: voip on the router is absolutely needed. there are phones connected to it which must continue to work normally. asterisk is just a low-priority thing. i can do port forward and change the asterisk ports, however (which i already started with) |
| 23:18.58 | lirakis | Jan-pp: .. good luck |
| 23:20.06 | Jan-pp | after a sip reload i get the message "sip_reg_timeout: -- Registration for 'xxxxx@sipgate.de' timed out" |
| 23:21.27 | lirakis | Jan-pp: then it is not reaching * for registration |
| 23:21.43 | lirakis | Jan-pp: read the link.. open ports |
| 23:22.22 | lirakis | Jan-pp: change ports.. if you have to .. but it will add confusion.. i.e. if you change default sip to 6732 from 5060 you will have to change it on your phone too |
| 23:23.14 | Jan-pp | no phones will connect to asterisk, ever. asterisk connects to sipgate account, listens for incoming calls and plays back a message (and later does more stuff) |
| 23:24.59 | lirakis | <PROTECTED> |
| 23:25.22 | Jan-pp | the sip port is changed and forwardet, i did not care about rtp, as the problem is with the sip registration (am i right?) |
| 23:26.03 | Jan-pp | lirakis: thx, maybe found something |
| 23:26.05 | lirakis | Jan-pp: yes.. rtp is media.. thats one way audio etc. ... your not there yet |
| 23:26.06 | lirakis | ;) |
| 23:27.24 | Jan-pp | Via: SIP/2.0/UDP 192.168.50.50:5060; looks like it does not get the external ip right |
| 23:27.40 | Jan-pp | externip=89.12.66.26 is in sip.conf, however.. |
| 23:29.26 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 23:30.57 | *** part/#asterisk exvito (n=exvito@89-180-79-126.net.novis.pt) |
| 23:31.18 | [TK]D-Fender | Jan-pp, You saying that the phones connected to your router use an outside resource and mst continue to work seperate of *? |
| 23:31.22 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 23:31.45 | [TK]D-Fender | bkruse_home, MokoIAX :D |
| 23:32.09 | Jan-pp | phones on router are normal cheap analog phones (no ip phones), router (fritz!box) has a phone jack and allows voip calls to be made from cheap phones |
| 23:32.15 | bkruse_home | [TK]D-Fender: yes ;] |
| 23:32.19 | bkruse_home | so youve seen? |
| 23:32.22 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 23:32.27 | [TK]D-Fender | bkruse_home, You like... totally rock :) |
| 23:32.44 | bkruse_home | [TK]D-Fender: hehe, thanks! Its going pretty well, just getting used to the build project |
| 23:32.48 | bkruse_home | build process* |
| 23:32.50 | lirakis | [TK]D-Fender: .. your not talking about openmoko are you? |
| 23:32.52 | Jan-pp | router is registered with some ohter provider. asterisk is going to register independently with sipgate |
| 23:33.01 | [TK]D-Fender | bkruse_home, I'm waiting for P1 to come out and from a reveiw or two to solidifyt he choice, but the preliminaries scream "awesome" |
| 23:33.13 | bkruse_home | itll be out before the hardware is released for consumers, sean said he would integrate it into the release if I wanted/it was ready |
| 23:33.35 | [TK]D-Fender | Jan-pp, see if you can get your router to sue something other than 5060 for its own purposes and then forward *'s normal ranges on to your server |
| 23:33.38 | bkruse_home | ya, I should be getting a couple neos in maybe 3 weeks or so |
| 23:33.38 | tomcontr3 | what ports do I need to open so Asterisk can work normally? |
| 23:33.51 | bkruse_home | 4569; 5060; 10000-20000 |
| 23:33.56 | [TK]D-Fender | all UDP |
| 23:33.59 | bkruse_home | thats iax, sip, rtp |
| 23:34.00 | bkruse_home | right |
| 23:34.02 | tomcontr3 | thanks |
| 23:34.13 | bkruse_home | [TK]D-Fender: yes, should be awesome, looking for devs now, but I kind of want to do it on my own too |
| 23:34.26 | Jan-pp | sip should be enough for me for the beginning (as soon as i want calls to work i will need rtp too, of course) |
| 23:34.29 | Jan-pp | right? |
| 23:34.45 | bkruse_home | Jan-pp: i guess? why not just do both |
| 23:34.48 | [TK]D-Fender | bkruse_home, Oh I wasn't volunteering.... I'm not qualified.. jsut wanted to pass on a pre-emptive "thanks" |
| 23:34.50 | bkruse_home | save yourself troulbe and confusion |
| 23:34.57 | bkruse_home | [TK]D-Fender: oh, awesome, well thank you :] |
| 23:35.03 | bkruse_home | I will keep you updated with screenies :D |
| 23:35.23 | lirakis | [TK]D-Fender: have you actually got a openmoko/Neo1973 |
| 23:35.33 | bkruse_home | i would bet no |
| 23:35.36 | [TK]D-Fender | bkruse_home, no need for pretty pictures. Your telling me that it works pretty well will do. The means isn't so relevant :) |
| 23:35.53 | Jan-pp | no way to change router port, sorry. going to do a etheral sniff at DSL level |
| 23:35.58 | bkruse_home | [TK]D-Fender: cool :] btw, did you see it on the projects list? didnt see it get accepted |
| 23:36.09 | bkruse_home | and my gmail box is crazy full because of the moko mailing list |
| 23:36.18 | [TK]D-Fender | bkruse_home, Though I'm sure that will be awesome too :p.... just borrow from the primary dialer app and all is good... heck I can imagine the addressbook integration you could have.... |
| 23:36.49 | bkruse_home | [TK]D-Fender: I am first doing a standalone app right? like a whole application for it, then I will integrate with dialer as just an outbound gateway option |
| 23:36.57 | [TK]D-Fender | Jan-pp, can you get into its config? |
| 23:37.24 | Jan-pp | router? webinterface only and it does not allow port changes. |
| 23:37.48 | Jan-pp | well, looks like the wireshark shows something: "unauthorized" |
| 23:38.15 | [TK]D-Fender | bkruse_home, Just saying you could use the Dialer base for your test interface, and that depending how the "core" contacts work, you could maybe use field in it from there to avoid creating a parallel database. |
| 23:38.41 | [TK]D-Fender | Jan-pp, Ok, forget port, what about reg, etc? Do you hav all of your connection ifo or do they hide it from you? |
| 23:39.04 | Jan-pp | "SIP/2.0 401 Unauthorized" means i probably messed the pw or username up, right? |
| 23:39.20 | [TK]D-Fender | Jan-pp, jsut seeing if you can have * connect to your provider instead and then point the router INWARDS for your *. |
| 23:39.21 | Jan-pp | asterisk is using sipgate, while router is using 1und1, so different providers, different accounts |
| 23:39.27 | [TK]D-Fender | Jan-pp, correct |
| 23:40.18 | bkruse_home | [TK]D-Fender: exactly, and it will just be like "dialout gsm/provider or voip/iax" |
| 23:40.23 | Jan-pp | the asterisk is on a box that is not running 24/7 - its just an experiment at the moment. router must stay 100% reliable, so no changes there (except port forward and the like) |
| 23:40.23 | bkruse_home | but will first be its own application |
| 23:41.15 | [TK]D-Fender | bkruse_home, Sure, just saying that borrowing their interface code could "speed-launch" you so you can concentrate on the IAX stack / hardware interfaces (audio, etc) |
| 23:41.16 | Jan-pp | is there a way to find out why I am getting the "unauthorized" from sipgate? |
| 23:41.28 | [TK]D-Fender | Jan-pp, 401 = bad user / pass |
| 23:41.35 | *** join/#asterisk nath0099_ (i=skylark_@82-34-164-64.cable.ubr02.maid.blueyonder.co.uk) |
| 23:42.52 | bkruse_home | [TK]D-Fender: yes, its going to be a good challenge, and I see exactly what you mean |
| 23:43.12 | bkruse_home | I often get frustrated and stop for the night when I cannot get a simple gtk thing working in the qemu |
| 23:43.49 | Jan-pp | i copied user/pass from the sip.conf to another voip program and the other program connected without any trouble! |
| 23:43.52 | [TK]D-Fender | bkruse_home, 3 "R"'s! |
| 23:44.17 | tomcontr3 | does anyone here uses Spanish sounds for asterisk? |
| 23:44.18 | [TK]D-Fender | Jan-pp, could be the "fromdomain", or rejected because they trakck UA's, etc |
| 23:44.30 | Jan-pp | ok, what to set the fromdomain to? |
| 23:48.04 | Jan-pp | if i want incoming calls only, what do i have to specify in sip.conf? is [general] with port bindaddr nat externhost and registern enough? |