| 00:03.09 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 00:03.54 | *** join/#asterisk jets (n=jets@root.ownsu.com) |
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| 00:06.53 | Lyfe | I haven't had to do this yet, but is there a function to read digits input from a phone? once the call's in place, anyway (i swore i'd read something about it, but not sure what nor where) |
| 00:07.15 | Lyfe | nevermind, i might be nuts. |
| 00:07.22 | russellb | show application Read |
| 00:07.57 | Lyfe | oh, guess i'm not nuts. |
| 00:09.51 | hads | Can anyone confirm if you do a NoOp(${STRFTIME(,,)})) or SayUnixTime(,) in your dialplan that it gives GMT time instead of the systems local timezone? |
| 00:10.42 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 00:11.41 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
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| 00:16.39 | infinity1 | i'm having dtmf issues when dialing out via voipjet and teliax. teliax does work better, but both are troublesome |
| 00:16.47 | infinity1 | anyone know what i can do to troubleshoot |
| 00:16.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:18.28 | MACscr | what file do i set the caller id in? |
| 00:18.41 | MACscr | sip.conf? |
| 00:21.39 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-8.getnet.net) |
| 00:23.09 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 00:25.06 | Skyelar | infinity1: IAX links with jitterbuffering? |
| 00:25.19 | Amilcar_ | MACscr: if it's a sip channel, yes, sip.conf |
| 00:26.19 | MACscr | basically i am running two companies on one asterisk system, i have them broken up into two different auto attendents |
| 00:26.30 | MACscr | i have the same staff monitoring both companies |
| 00:26.42 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:26.52 | MACscr | i want them to know which company was called when the phone rings at their extension |
| 00:28.48 | Skyelar | MACscr: callerid per call, not per phone. I see. You'll need to set it in your dialplan as the call comes in |
| 00:28.53 | intralanman | MACscr: in that case you probably wanna play with the callerid variable in the dialplan.... maybe set the calleridname to the co name |
| 00:30.10 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:30.46 | MACscr | someone else set this up, so im definately going to be hacking it up. Doh. Anyway, i know this sounds dumb, but where is the dialplan set |
| 00:31.52 | Amilcar_ | MACscr: you better start googling about asterisk. :-) extensions.conf is the dialplan. |
| 00:32.38 | Amilcar_ | I'll start playing with voicemail-imap.... Anyone here using it?? |
| 00:32.48 | intralanman | MACscr: just let us know if you need to hire a consultant or two ;) |
| 00:33.17 | Amilcar_ | :) |
| 00:33.22 | MACscr | lol, i have found that to be the common response around here. your vulchers =P |
| 00:33.24 | groogs | So I take it this is probably a bad result from zttest: --- Results after 11060 passes --- Best: 100.000000 -- Worst: 99.841309 -- Average: 99.993931 |
| 00:34.31 | intralanman | MACscr: you have us all wrong:) we're happy to point you in the right direction.... but if it's mission critical and you need it to work right now....we can probably get it done faster than we can tell you how to do it |
| 00:34.36 | infinity1 | Skyelar: elaborate on the iax and jitterbuffering |
| 00:34.43 | infinity1 | i just googled some more stuff and saw something about it |
| 00:34.43 | groogs | Amilcar_: oooh, where is that from? |
| 00:34.57 | *** join/#asterisk rikstah (n=rick@c-71-227-234-92.hsd1.or.comcast.net) |
| 00:34.59 | infinity1 | i'm using iax with both voipjet and teliax so... |
| 00:35.03 | Skyelar | infinity1: I was wondering whether you were getting bitten by the following: http://bugs.digium.com/view.php?id=6011 |
| 00:35.28 | MACscr | throw out your hourly rates people, id like to see what my options are =P |
| 00:35.36 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:35.40 | infinity1 | i put jitterbuffer=no in both of the teliax/voipjet contexts in iax.conf |
| 00:35.46 | infinity1 | so now i'll text more i guess |
| 00:36.13 | Amilcar_ | groogs: get the trunk, make menuselect, and you will see a beautiful and misterious option! ;-) |
| 00:36.42 | Amilcar_ | :-) |
| 00:36.44 | groogs | ah, cool |
| 00:36.49 | intralanman | MACscr: i usually work by the job.... pm me if you're serious though |
| 00:38.02 | Amilcar_ | groogs: imap support has been introduced in rev. 39404. |
| 00:38.58 | groogs | interesting, i was actually considering doing something with that as an AGI, just haven't had time yet |
| 00:39.10 | groogs | thanks, i'll have to look and see whats happening |
| 00:39.43 | Amilcar_ | groogs: me too! :-) |
| 00:40.29 | groogs | mostly it annoys me that 99% of the time, i listen to my voicemail in email, but then i still have to log in from my phone to get the red light to go off ;p |
| 00:41.08 | Skyelar | groogs: you can tell voicemail to delete it from your mailbox after emailing it |
| 00:41.57 | groogs | i know, but i do still use regular voicemail sometimes |
| 00:42.27 | Skyelar | groogs: fair enough |
| 00:42.31 | Amilcar_ | Skyelar: that's the problem. I want to use email AND regular phone |
| 00:42.34 | Amilcar_ | :) |
| 00:43.54 | Skyelar | Amilcar_: the other option is installing a script, accessable via the web, that deletes a specified message from a mailbox, and put a link in each email to the delete app. But I must admit, if you're willing to go SVN, IMAP sounds much nicer :) |
| 00:46.30 | Amilcar_ | :-) |
| 00:46.47 | groogs | does it actually store in imap, or just sync imap with file-based storage? |
| 00:47.01 | Amilcar_ | Skyelar: sure we can think many other ways to do it, but native imap support for vm storage is veeery cool! ;-) |
| 00:47.57 | hads | groogs: It stores in IMAP AFAIK |
| 00:48.01 | Amilcar_ | groogs: take a look - http://svn.digium.com/view/asterisk/trunk/doc/imapstorage.txt?rev=39404&view=markup |
| 00:51.44 | groogs | hm, yeah, that would work well if you have a separate imap server on your asterisk box.. |
| 00:52.00 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 00:53.24 | groogs | my imap is actually on our webserver in another city.. so it would be bad to use (eg, if internet is down, people can't leave vm.. not to mention bandwidth/speed issues). i also like to store basically all my non-junk mail.. i have messages going back to like 2002.. |
| 00:55.06 | groogs | so in my situation, assuming i wanted a combined email+voicemail inbox, doing a sync-type setup would be preferable (basically, poll the imap server ocasionally to see if the specific VM messages are read/deleted.. and mark them as read if i listen to them via *, and optionally delete them if i delete from *) |
| 00:55.10 | Lyfe | so, apparently if i park a call, the asterisk CLI starts spamming me with stuff about " -- Attempting native bridge of SIP/joe2-0874c000 and SIP/joe-08760000" |
| 00:55.48 | groogs | just food for thought, anyways. |
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| 01:37.31 | pyrom | Using the spa3102, can i call out via the Connected landline? |
| 01:37.38 | [TK]D-Fender | pyrom : Sure |
| 01:37.43 | pyrom | via, asterisk. |
| 01:37.55 | [TK]D-Fender | pyrom : You can use it in a ton of different ways |
| 01:38.03 | pyrom | Great! |
| 01:38.35 | [TK]D-Fender | pyrom : Use as an FXO gateway for * to bridge other calls to, use as a failover for the FXS port if the server isn't repsonding, or just pump local suff out direct etrc.... |
| 01:38.43 | [TK]D-Fender | pyrom : Its a remarkable value. |
| 01:39.17 | suma | Is there is IAX FXO Device? My sipura 3000 is not working fine with NAT |
| 01:39.22 | pyrom | Any guides/pointers on how i can use it as an FXO gateway?, been trying for hours :-=) |
| 01:39.52 | suma | pyrom: just give the asterisk sip username and password and register |
| 01:40.01 | [TK]D-Fender | pyrom : www.voxilla.com . Go check out the sipura/Linksys forums on them and there are "stickied" threads on the topic |
| 01:40.06 | [hC] | would the spa3102 be a good replacement for single analog failover sites of mine instead of using an a200? |
| 01:40.25 | [TK]D-Fender | [hC] : Its a failover, not the full time solution so sure. |
| 01:40.32 | suma | pyrom: in the dialplan (xx.) should be there , then whatever number you pass you will get the sipura to dial for you |
| 01:40.42 | [TK]D-Fender | [hC] : They're actually pretty decent, but A200 is a class of its own. |
| 01:40.49 | file | [TK]D-Fender: guess what just shipped |
| 01:41.12 | file | [TK]D-Fender: everything except the case |
| 01:41.13 | [TK]D-Fender | suma : He's talking about use as an FXO gateway, not the FXS for handset... |
| 01:41.22 | [TK]D-Fender | file : LOLZ! |
| 01:41.48 | suma | [TK]D-Fender, yes i'm also speaking about the FXO gateway sir ! |
| 01:41.52 | pyrom | suma, been struggling with this device all day, but after upgrading firmware things seems to get a little better :-) |
| 01:42.17 | *** join/#asterisk LoneShadow (n=duh@59.92.169.215) |
| 01:42.33 | [TK]D-Fender | suma : Most implementations shouldn't HAVE a dialplan for the FXO port but rather just pass on all calls in/out direct. |
| 01:42.39 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 01:42.45 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 01:42.47 | suma | [TK]D-Fender, SIPURA has |
| 01:42.57 | [TK]D-Fender | suma : Dial-plan's belong in the PBX not the phone. |
| 01:43.03 | suma | [TK]D-Fender, you can configure what number to go and what not |
| 01:43.22 | pyrom | suma, this is all setup in the PSTN Line tba? |
| 01:43.23 | pyrom | tab |
| 01:43.25 | suma | [TK]D-Fender, ha ha, SIPURA has a dial plan, check the faq sir |
| 01:43.34 | *** join/#asterisk fluxbuntu (n=joejaxx@unaffiliated/joejaxx) |
| 01:43.46 | suma | [TK]D-Fender, If you have a SIPURA check it out |
| 01:43.54 | suma | pyrom: yes |
| 01:43.59 | [TK]D-Fender | suma : I know they have them, just in an * environment the ATA shouldn't start thinking it has a brain :) |
| 01:44.29 | [TK]D-Fender | suma : And I've owned a half-dozen different models, including the 3102 and 3000 :) |
| 01:44.41 | suma | [TK]D-Fender, come on, we need to security at that end also, otherwise someone will misuse the Sipura |
| 01:44.46 | pyrom | So i dont even need to use the "PSTN USER" & "User 1 " Tab's? |
| 01:45.09 | [TK]D-Fender | pyrom : Yes you certainly do. |
| 01:45.21 | LoneShadow | suma where are you using your spa3k ? |
| 01:45.34 | [TK]D-Fender | pyrom : its just that you won't be setting a dialplan for the ATA to choose what and where to send calls to/from |
| 01:45.43 | LoneShadow | in US ? |
| 01:45.49 | suma | pyrom: you don't need those tabs to configure, just pstn line alone is enough |
| 01:45.55 | [TK]D-Fender | suma : Misuse how? |
| 01:46.50 | suma | [TK]D-Fender, If you have your Sipura Online, if I can sniff that I can make IDD calls on your sipura, then i will call sipura directly and make outgoing call |
| 01:47.05 | pyrom | suma, so then it's only " |
| 01:47.05 | pyrom | Proxy and Registration " & !" |
| 01:47.05 | pyrom | SIP Settings" !" |
| 01:47.06 | pyrom | Subscriber Information" ? |
| 01:47.10 | pyrom | Sorry for that. |
| 01:47.20 | [TK]D-Fender | suma : Not if you set it up right. Have it register and only accept authed calls. |
| 01:47.38 | pyrom | Back to the forum. |
| 01:48.08 | LoneShadow | pyrom you trying to configure your spa3k ? |
| 01:48.18 | suma | [TK]D-Fender, Even authed calls easy to break though |
| 01:48.25 | pyrom | LoneShadow, yeah, not going quit well |
| 01:48.33 | pyrom | LoneShadow, Feel like throwing it on the freeway |
| 01:48.40 | LoneShadow | no no |
| 01:48.45 | LoneShadow | just ship it to me :P |
| 01:48.52 | LoneShadow | I can take good care of it ;) |
| 01:49.05 | LoneShadow | so which part dosnt work for you ? |
| 01:49.29 | pyrom | First i had callerid problems, those seems to be sorted with firmware upgrade, after checking every setting out there :-) |
| 01:49.31 | [TK]D-Fender | suma : Well at that point you're screwed with ANY solution! ;) |
| 01:49.42 | pyrom | Now i'm trying to fix the FXO gateway |
| 01:50.00 | LoneShadow | suma, [TK]D-Fender: did you folks had to change the impedance or GAIN ? |
| 01:50.49 | suma | LoneShadow, I'm going to, and going to increase the GAIN and reduce impedence |
| 01:51.05 | LoneShadow | suma: you using the box in US or some other country ? |
| 01:51.22 | suma | LoneShadow, I'm using one in india and one in singapore |
| 01:51.27 | LoneShadow | aah nice :D |
| 01:51.50 | LoneShadow | my US box spa3k works fine, having issues with the India one |
| 01:51.59 | LoneShadow | echo and noise issues |
| 01:52.23 | suma | LoneShadow, You might need to change the line impedence |
| 01:52.38 | suma | LoneShadow, India is bit complex, which part of india you are using it though? |
| 01:52.43 | LoneShadow | Bangalore |
| 01:53.23 | LoneShadow | my PSTN to VOIP gateway will be turned off, just patching incoming pstn line calls to ring the phones |
| 01:53.55 | LoneShadow | and still the caller gets echos |
| 01:54.43 | suma | sipura is not good with echo cancellation |
| 01:54.51 | [hC] | Qwell: Yo |
| 01:54.58 | Qwell | [hC]: hey |
| 01:55.02 | suma | you can try with any of the TDM cards |
| 01:55.27 | [hC] | Qwell: You are hired full time with digium now? |
| 01:55.35 | Qwell | [hC]: I am :) |
| 01:55.43 | [hC] | Qwell: Congrats man :) |
| 01:55.46 | Qwell | thanks |
| 01:56.01 | [hC] | Are you gonna move to alabama??? |
| 01:56.06 | [TK]D-Fender | LoneShadow : I didn't. |
| 01:56.41 | Qwell | [hC]: yep, in a couple months |
| 01:57.14 | [hC] | Qwell: wow. Crazy. Thats gonna be such a change from cali :) |
| 01:57.22 | Qwell | yeah, heh |
| 01:58.27 | [hC] | OooOo... Astricon schedule posted finally! |
| 01:59.07 | [hC] | Time to buy my tickets :) |
| 01:59.23 | Qwell | buy my plane ticket too |
| 01:59.44 | [hC] | Done! |
| 01:59.46 | Qwell | :p |
| 02:00.38 | [hC] | Im mega excited, my company is gaining some great momentum, and hit the obvious first milestone today, we acquired our own office space, complete with a drive in garage warehouse area! |
| 02:00.45 | [hC] | And now we're taking on financing, which rules. |
| 02:00.55 | harryvv | ohh sweet |
| 02:01.01 | file | moneyz! |
| 02:02.01 | [hC] | It was pretty neat, I was digging thru stuff the other day and found the first grandstream bt-100 that started my asterisk foray... |
| 02:02.23 | [hC] | What a piece of shit that thing was.. |
| 02:02.25 | [hC] | is? |
| 02:02.26 | file | did you set it aflame? |
| 02:02.34 | [hC] | Actually I configured it to register |
| 02:02.37 | [hC] | it didnt (shocker) |
| 02:02.48 | [hC] | tried dialing and it went thru, but the speaker just went BNZZZZZZZzzzzzzzzzzzzzzZZZ |
| 02:02.51 | [hC] | until i unplugged it |
| 02:02.55 | [hC] | so i just put it back in the closet. |
| 02:02.55 | [hC] | :) |
| 02:07.37 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 02:09.58 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 02:11.20 | ManxPower | Good god man! At least put it inside a pentagram! |
| 02:11.29 | ManxPower | A ritual burning would be best, of course. |
| 02:13.32 | ManxPower | The cleanse the area by burning some sage. |
| 02:13.42 | ManxPower | You can't be too careful with those GS phone. |
| 02:14.20 | Kerry_G | In case anyone cares, the new Linksys SPA400 (4 port FXO) took all of 2 settings to get it to work with * |
| 02:14.35 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 02:14.55 | ManxPower | Kerry_G, The other SPAs usually take three. host to connect to, userid, and password |
| 02:16.21 | Kerry_G | this took host to connect to and changing the user id to the DID number |
| 02:16.36 | Kerry_G | so its 33% easier |
| 02:17.06 | Kerry_G | there are hardly any settings on it to begin with |
| 02:18.24 | file | 33.33% |
| 02:18.33 | tengulre | Hi,all |
| 02:18.53 | [TK]D-Fender | Kerry_G : What does the Voicemail server part do really? USB for storage? |
| 02:19.05 | tengulre | I got many characters in /var/log/messages, 'FXO PCI Master abort'? why? |
| 02:19.26 | Kerry_G | yes, you can use the USB port with a flash drive or USB hard drive for storage |
| 02:19.42 | [hC] | Thats kinda cool. |
| 02:19.51 | [TK]D-Fender | Kerry_G : I meant it doesnt' have any interal stoarge capacity does it? |
| 02:19.58 | [hC] | What sort of read/write limitations are there on USB flash drives? |
| 02:20.01 | Kerry_G | no it does not |
| 02:20.02 | [hC] | just like compact flash? |
| 02:20.18 | [hC] | "limited" |
| 02:20.25 | Kerry_G | big debate on that, some say 1000 and others say millions |
| 02:21.08 | [hC] | im in the process of building a CF based astlinux box, but need to figure out the best solution for a keydisk for VM storage and such. Debating either laptop drive or microdrive at the moment. |
| 02:21.21 | ManxPower | tengulre, that means there is a significant issue with the card getting interripts. check for IRQ conflicts |
| 02:21.22 | [hC] | I want something thats cost effective and least likely to shit the bed, so to speak. |
| 02:21.24 | tengulre | hi, anybody know why? |
| 02:23.03 | Kerry_G | I know guys running PFSense on CF cards for ages without any problems |
| 02:23.57 | tengulre | ManxPower, Thanks for reply, but how to view all interripts messages in system. |
| 02:25.02 | tengulre | MaxPower, have not conflicts in my linux box, I using 'lspci -v' to view, all device using different interript number. |
| 02:28.57 | *** join/#asterisk pcm (n=pcm@72.146.59.132) |
| 02:30.47 | ManxPower | tengulre, cat /proc/interrupts |
| 02:31.22 | tengulre | <PROTECTED> |
| 02:31.45 | ManxPower | If you do not have an IRQ conflict, then the only other things that could cause that are difficult to fix. SATA controllers can cause that, RAID controllers can cause that, GigabitEthernet controllers can cause that |
| 02:31.51 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 02:32.50 | tengulre | crying....! |
| 02:33.22 | [TK]D-Fender | X100's cause that.... |
| 02:33.41 | [TK]D-Fender | :O |
| 02:34.50 | tengulre | wuwu.... |
| 02:35.22 | harryvv | What is the GDM designation in front of alot of digium cards mean? |
| 02:35.53 | JT | TDM? |
| 02:35.57 | JT | you mean tdm? |
| 02:37.27 | JT | harryvv: ? |
| 02:37.30 | tengulre | http://rafb.net/paste/results/xH29K994.html |
| 02:39.01 | harryvv | Jt yea sorry was looking at voipsupply and this is a new designation. |
| 02:39.05 | JT | Time Division Multiplexing |
| 02:39.15 | harryvv | no GDM |
| 02:39.24 | JT | pretty standard concept used in telcoms |
| 02:39.25 | JT | oh |
| 02:39.28 | harryvv | as in DGM-TDM01B |
| 02:39.42 | JT | then i wouldn't say it's in front of "a lot" of digium cards :P |
| 02:40.10 | *** join/#asterisk mds2 (n=mds@thewife.inspirednetworks.co.nz) |
| 02:40.29 | mds2 | morning |
| 02:40.39 | harryvv | [TK]D-Fender would know |
| 02:40.51 | tengulre | mds2: where are you from? |
| 02:41.16 | [TK]D-Fender | DGM - sounds like DiGiuM to me..... |
| 02:41.25 | JT | indeed |
| 02:41.43 | JT | harryvv is dyslexic, saying GDM instead of DGM :P |
| 02:42.40 | *** join/#asterisk Skarmeth (n=Skarmeth@201009084250.user.veloxzone.com.br) |
| 02:43.17 | harryvv | Tk, yea that is a little strange because in the past I do not recall seeing that and JT, hell no I am not dyslexic :) |
| 02:43.44 | mds2 | tengulre: New Zealand, why's that? |
| 02:44.39 | JT | harryvv: i just checked out the voipsupply site, you sid GDM the first 2 times, heh |
| 02:45.00 | tengulre | here is morning too! :) |
| 02:45.22 | JT | it's not morning in new zealand |
| 02:45.27 | mds2 | anyone know what causes a Cisco 79xx to say "-- Got SIP response 400 "Bad Request" back from x.x.x.x" when * tries to light its message waiting light? |
| 02:45.30 | JT | "morning" is more a figure of speech on irc |
| 02:45.57 | [TK]D-Fender | Dyslexics of the world untie! |
| 02:45.58 | mds2 | <- not paying much attention to the clock |
| 02:46.50 | *** join/#asterisk niter3 (n=niter3@d57-102-239.home.cgocable.net) |
| 02:47.31 | niter3 | I've installed Festival and set it up in asterisk. I put a test extension in my extensions.conf and I dial the number it connects and Festival says it accepts the connection but I hear no sound. |
| 02:47.40 | niter3 | I've installed a festvox |
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| 03:01.47 | delmar | how do you upgrade the firmware on the linksys/sipura spa2102/1002? I can't see anything on the web-ui as to how you do it... |
| 03:04.04 | ManxPower | delmar, the linksys firmware you download has an installer. |
| 03:04.21 | [hC] | Yarr, It be a pokin' match! |
| 03:04.43 | delmar | ManxPower, ah ok. im not local to the devices so I was looking into it via remote on the webconsole.. |
| 03:04.59 | delmar | ManxPower, i will advise the person onsite to run the installer.. and see what happens |
| 03:05.51 | mds2 | anyone know how to turn on mwi_status (Message Waiting Indicator) on a Cisco 79xx? |
| 03:06.04 | delmar | ah ok. now i am told there is a .bin file and a .exe :P. makes sense now. |
| 03:06.27 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:06.29 | [hC] | Hmm. Can I use a regex match like this? |
| 03:06.30 | [hC] | exten => _NXXNXXXXXX/101|102|201|202|301|302,1,Congestion |
| 03:06.33 | [hC] | for incoming CID? |
| 03:07.06 | crochat | Hello ! |
| 03:07.06 | delmar | I think both of these SPA's might be poked. line2 on the spa2102 doesnt ring. line1 on the spa1001 doesnt ring. line1 on the spa2102 rings fine. |
| 03:07.30 | crochat | I have a problem with MP3Player application... I can't make it work :-( |
| 03:07.42 | delmar | crochat, dont use it !! |
| 03:07.51 | crochat | But I have mpg123 0.59r |
| 03:08.01 | crochat | delmar: What should I use ? |
| 03:08.10 | delmar | crochat, use asterisk native music on hold. mpg123 is a pig. you are talking about mpg123 yes? |
| 03:08.23 | ManxPower | [hC], no |
| 03:09.10 | crochat | delmar: Yes, but if I want to hear a streaming mp3 from an online radio, does music on hold work too ? |
| 03:09.37 | [hC] | ManxPower: is there an effective way to do that? or do i have to list each one by hand? |
| 03:09.44 | delmar | crochat, that i cant say. never done that before. sounds dodgy :P |
| 03:10.13 | ManxPower | exten => _NXXNXXXXXX/10[1-2],1,Congestion |
| 03:10.39 | ManxPower | exten => _NXXNXXXXXX/[1-3]0[1-2],1,Congestion |
| 03:10.43 | ManxPower | that would work too |
| 03:11.02 | [hC] | Yeah :) I was just curious if it was able to evaluate OR or take a list.. ok.. Thanks! |
| 03:11.22 | ManxPower | [hC], if you want REAL regexs you need a GotoIf or something like that |
| 03:11.31 | ManxPower | oh! |
| 03:11.38 | ManxPower | exten => _NXXNXXXXXX/_[1-3]0[1-2],1,Congestion |
| 03:11.50 | ManxPower | since the CID is a pattern match..... |
| 03:12.12 | justinu | if you want real regex in dialplans, check out freeswitch |
| 03:13.00 | *** join/#asterisk num000 (n=numerobi@e177185080.adsl.alicedsl.de) |
| 03:13.18 | [hC] | Ah yes of course! |
| 03:15.21 | num000 | i've no space on my device, could i delete those asterisk modules which i will not use? |
| 03:16.23 | [hC] | Any of you guys ever mucked with adjusting the ringer gains on the polycom 501? |
| 03:16.34 | [hC] | I have one unit in a noisy area and on full ring they still have a hard time hearing it |
| 03:16.38 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 03:16.40 | [hC] | im looking in sip.conf and noticing a lot of gain settings. |
| 03:16.43 | [hC] | er sip.cfg |
| 03:17.03 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 03:17.03 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:20.29 | *** join/#asterisk cybertrickle_ (n=cybertri@ip70-190-74-204.ph.ph.cox.net) |
| 03:23.43 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 03:28.52 | crochat | delmar: moh does not work too :-( |
| 03:29.37 | crochat | delmar: Asterisk said : -- Started music on hold, class 'default', on channel 'SIP/..... |
| 03:30.11 | crochat | delmar: But there was just a tone like if it was ringing... |
| 03:31.05 | crochat | delmar: And my mp3 file is CBR 128kbps without any ID3 tag |
| 03:31.07 | delmar | crochat, http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 03:32.28 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 03:33.07 | CunningPike | crochat: You should try to make it 8KHz mono - you should also use native MOH (if you're not already) and match your codec to what you are using for your calls (ulaw or whatever) |
| 03:33.37 | CunningPike | ~nativemoh |
| 03:33.57 | CunningPike | Yo, jbot |
| 03:34.09 | CunningPike | ~moh |
| 03:34.10 | jbot | moh is probably Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
| 03:36.59 | *** join/#asterisk }btorch{ (n=btorch@c-66-176-87-59.hsd1.fl.comcast.net) |
| 03:37.42 | *** join/#asterisk SwK (n=Silik0nJ@c-24-99-246-180.hsd1.ga.comcast.net) |
| 03:42.02 | [TK]D-Fender | [hC] : You know my solution already ;) |
| 03:42.22 | [hC] | [TK]D-Fender: ? :) |
| 03:42.38 | [TK]D-Fender | [hC] : To the gain issue.... |
| 03:42.52 | [hC] | You are going to tell me something smart assed, I can feel it. :) |
| 03:43.07 | [TK]D-Fender | [hC] : You gain wisdom child! |
| 03:43.18 | [hC] | [TK]D-Fender: Haha |
| 03:43.24 | Skyelar | crochat: had the channel been answered before you attempted to start the MOH? |
| 03:43.39 | [hC] | Maybe I will rig a megaphone taped to the desk, with an AC Adapter |
| 03:43.45 | [TK]D-Fender | [hC] : sample up your own ringer and crank it at source. |
| 03:43.49 | [hC] | or even better yet, batteries that have a battery charger in line. |
| 03:44.15 | [hC] | [TK]D-Fender: Yeah, Im just using all the builtins right now. I'll put like high pitched chimes or something on there, custom, first. |
| 03:45.30 | [TK]D-Fender | [hC] : or as someone else here said "one saying out loud 'answer the damn phone bitch'" |
| 03:45.36 | crochat | Skyelar: Aaaargggghhhhh !!!!! My dialplan is so big that I omitted to check that in my test :-( |
| 03:46.02 | Skyelar | crochat: that'd likey cause your ringing sound then :-) |
| 03:46.10 | crochat | Skyelar: It's passed my bedtime ;-) |
| 03:46.20 | Skyelar | s/likey/likely/ |
| 03:46.46 | crochat | jbot: Sure, I'll test it now ;-) |
| 03:47.42 | Skyelar | crochat: staying up overly late hacking code/dialplan/whatever tends to just mean you have to work half the rest of the day fixing all the problems you created whilst tired... although I must admit to having 2am "inspirations" sometimes |
| 03:48.07 | crochat | Great ! It works fine ! At last... |
| 03:48.17 | crochat | I'm soooo tired :-( |
| 03:48.36 | }btorch{ | do you guys ever hear some sort of very low garbage noise on the background when either calling from IAX<->IAX or IAX<->ZAP |
| 03:48.45 | crochat | 05:48 AM here... and still not in bed |
| 04:01.02 | JT | is AEL still considered too experimental for production use? |
| 04:01.45 | [TK]D-Fender | JT : AEL is for the most-part a waste of time. Few people use it and it offers nothing you can't do with standard extension logic. |
| 04:01.55 | mds2 | anyone got a Cisco 79xx that actually displays date/time on its LCD? |
| 04:01.57 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 04:02.10 | crochat | Skyelar: I'm still trying to play http://broadcast.infomaniak.ch/rtn-low.mp3 but it doesn't work :-( |
| 04:03.07 | JT | [TK]D-Fender: fair enough |
| 04:03.14 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 04:04.04 | [TK]D-Fender | JT : Its a nice idea, and AEL2 is going to be considered more "stable" and usable in 1.4. Mind you us old-schoolers are unlike to switch in droves... |
| 04:04.22 | *** join/#asterisk gkemper (n=me@S0106000f6636e18c.ed.shawcable.net) |
| 04:05.06 | JT | heh |
| 04:07.59 | *** join/#asterisk apardo (n=apardo@87.217.146.143) |
| 04:09.36 | gkemper | Hello. Could anybody tell me if just a standard modem could be used to connect to PSTN instead of a FXO card? |
| 04:09.48 | [hC] | Crap. The problem with astricon is i want to take part in ore than one track per day |
| 04:09.49 | [hC] | :) |
| 04:09.55 | [hC] | [TK]D-Fender: you going to astricon this year? |
| 04:10.05 | *** join/#asterisk tdonahue-laptop (n=tdonahue@seymour-cuda1-69-173-87-106.albyny.adelphia.net) |
| 04:11.57 | CunningPike | gkemper: No |
| 04:12.14 | CunningPike | gkemper: If it could, we all be doing it |
| 04:12.35 | CunningPike | [hC]: I am |
| 04:12.46 | gkemper | SHIT |
| 04:13.02 | *** join/#asterisk TrickFinlay2 (n=Trickste@71-10-242-220.dhcp.oxfr.ma.charter.com) |
| 04:13.42 | TrickFinlay2 | anyone using asterisk/mythtv on the same box? |
| 04:13.55 | [hC] | CunningPike: woo, me too :) This your first year, or did you go last year? |
| 04:14.16 | [TK]D-Fender | [hC] : Nope |
| 04:14.29 | CunningPike | I went last year - it was really good, especially as I was a relative newbie at the time (and maybe now!) |
| 04:14.34 | [hC] | Doh :/ |
| 04:14.48 | CunningPike | [TK]D-Fender: How come? |
| 04:14.48 | [hC] | CunningPike: I probably met you and I forget :S |
| 04:14.52 | russellb | TrickFinlay2: i do |
| 04:14.53 | [TK]D-Fender | Poor economic choice for me... |
| 04:15.03 | [hC] | CunningPike: wait.. you're from vancouver, thats why i recognize your name |
| 04:15.12 | CunningPike | [hC]: Indeed I am |
| 04:15.18 | [hC] | CunningPike: Me too |
| 04:15.20 | [TK]D-Fender | CunningPike : time off work, travel & stay costs, conf costs.... |
| 04:15.24 | [hC] | CunningPike: who do you work for again? |
| 04:15.35 | CunningPike | [hC]: Cool - District of North Vancouver |
| 04:15.39 | russellb | you guys coming to hear file and I talk about Asterisk coding foo?! |
| 04:15.40 | CunningPike | [TK]D-Fender: Too bad |
| 04:15.54 | [TK]D-Fender | CunningPike : Yup, if it were next door and cheap I'm in! |
| 04:15.57 | CunningPike | [TK]D-Fender: No employer to pay for you, eh |
| 04:16.11 | [hC] | CunningPike: Ahh yes yes. Do you know of anyone who's into asterisk who's looking for work? After our round of financing we're pulling in this week, im going to be looking for talented people to join us |
| 04:16.12 | [TK]D-Fender | CunningPike : I struggled to get * in the door there :) |
| 04:16.31 | CunningPike | [TK]D-Fender: I hear you |
| 04:16.38 | harryvv | CunningPike so your in Vancooover :) |
| 04:16.47 | file | coding foooooo |
| 04:16.49 | harryvv | CunningPike what brings your interest here |
| 04:16.50 | [TK]D-Fender | CunningPike : and last year barely convinced them at the start to send me to a Meet Asterisk conf here where I met a bunch of guy's include file and JunK-Y |
| 04:17.03 | CunningPike | [hC]: Not as such, although I've been known to moonlight after hours ;) |
| 04:17.18 | CunningPike | harryvv: Yes, indeedy. You too? |
| 04:17.21 | harryvv | yea |
| 04:17.35 | TrickFinlay2 | russellb :any tips for setting one up,i plan to use the mythtv/asterisk in my dorm next year |
| 04:17.45 | [hC] | CunningPike: Right on.. Well, maybe we can talk sometime about it, if you want some extra stuff to do :) |
| 04:17.59 | CunningPike | There's a few of us - I'd like to get a Vancouver AUG off the ground, but haven't had much response |
| 04:18.05 | CunningPike | [hC]: Sure |
| 04:18.16 | russellb | TrickFinlay2: you should have no conflicts. The only potential issue is if you use mythphone on the same box as asterisk |
| 04:18.18 | harryvv | CunningPike that may mean thats good or bad |
| 04:18.41 | russellb | TrickFinlay2: if so, you will need to set one of the two to use a different port |
| 04:19.06 | TrickFinlay2 | yeah in this case i wont be using mythtvphone |
| 04:19.15 | harryvv | I guess I could ask what is your mission statment or bussiness plan if thats how far you have taken it :) |
| 04:19.27 | russellb | TrickFinlay2: then you should have no problems :) |
| 04:19.35 | TrickFinlay2 | nice |
| 04:20.29 | TrickFinlay2 | russellb: pretty much any generic phone should work correct? |
| 04:20.57 | russellb | TrickFinlay2: well, yeah ... given that you have something to plug the phone into |
| 04:21.32 | TrickFinlay2 | alrighg well i was just going over this list |
| 04:21.33 | TrickFinlay2 | http://www.voip-info.org/wiki/view/Asterisk+phones |
| 04:21.57 | *** join/#asterisk lmpbzktwn5 (n=lmpbzktw@24-151-139-231.dhcp.oxfr.ma.charter.com) |
| 04:29.02 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:37.26 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 04:37.30 | redder86 | russel are you here? |
| 04:37.38 | russellb | no |
| 04:37.46 | russellb | what's up? |
| 04:37.57 | redder86 | give the people a chance to actually respond before you go closing that thihg |
| 04:37.59 | redder86 | thing |
| 04:38.18 | redder86 | http://bugs.digium.com/view.php?id=7742 |
| 04:38.28 | russellb | if you file a bug, you have permission to reopen it, right? |
| 04:38.44 | russellb | in any case ... |
| 04:38.54 | redder86 | I don't see how to reopen it. |
| 04:38.59 | russellb | the code is not disclaimed, and the person filing the report is not in a position to disclaim it |
| 04:39.17 | russellb | i assume that is you? |
| 04:39.28 | *** join/#asterisk pengyong (n=lala@222.188.135.252) |
| 04:39.30 | redder86 | I'm not disclaiming it, and they will likely public-domain it. But you got to give them a moment to actually do that before you go closing the report. |
| 04:39.54 | lmpbzktwn5 | russellb: in order to make a answering machine with asterisk and standard telephone service, do i just need a comp with asterisk, an FXO interface card and a POTS? |
| 04:40.13 | russellb | redder86: let them know that they will need to file a new report once a disclaimer is in place ... |
| 04:40.29 | redder86 | I'm doing some leg-work for them in opening the bug report ... because they doubted that you/Digium would even let the patch in at all ... even after jumping through all the hoops. So I'm trying to convince them that you're not so unreasonable as your reputation holds. |
| 04:41.02 | russellb | well I didn't close it saying "no, never." I said that we can't accept it, and I said what has to happen for it to be considered |
| 04:41.10 | CunningPike | lmpbzktwn5: Pretty much...... |
| 04:41.25 | redder86 | nobody can comment on that bug any further because it's closed now |
| 04:41.45 | CunningPike | lmpbzktwn5: Either a card, or an external gateway |
| 04:41.54 | russellb | they will need to open their own report once a disclaimer is on file. |
| 04:42.39 | redder86 | ah look, I found the reopen button |
| 04:42.39 | lmpbzktwn5 | cunningpike: the discontinued x100p should suit my needs if i'm just using it for an answering machine (yes, overkill i know) etc.? |
| 04:42.45 | russellb | don't reopen it. |
| 04:43.09 | CunningPike | lmpbzktwn5: It should - it's not the best card (so I've heard) but it should suffice |
| 04:43.21 | CunningPike | lmpbzktwn5: Especially if you already have one :) |
| 04:43.29 | redder86 | your making this difficult is only confirming the negative opinion of the patch submission process. |
| 04:43.50 | lmpbzktwn5 | cunningpike: any suggestions on better FXO PCI cards? |
| 04:43.51 | harryvv | its overkill but who cares. |
| 04:44.43 | CunningPike | lmpbzktwn5: I have no personal experience with them, but Digium and Sangoma cards are supposed to be good |
| 04:44.46 | russellb | redder86: I told you exactly what has to happen to get this code considered, *less than 30 minutes* after you started this process |
| 04:44.52 | russellb | what is the problem? |
| 04:44.56 | russellb | go make it happen |
| 04:45.08 | russellb | and how about i get back to fixing asterisk bugs |
| 04:45.24 | lowlevel | joy! |
| 04:45.51 | lowlevel | lmp: just bite the bullet and get the digium cards |
| 04:45.59 | redder86 | sigh |
| 04:46.01 | lowlevel | rather one card |
| 04:46.11 | TrickFinlay2 | CunningPike: me and lmpbzktwn5 were trying to create the best possible setup using both mythtv and some sort of pci phone card,another other tips/suggestions? |
| 04:46.37 | JT | lmpbzktwn5: there is pretty much nothing else in price range |
| 04:46.48 | CunningPike | lmpbzktwn5: Ummm - what's mythtv? |
| 04:46.59 | hads | Can someone confirm either way, if they do a NoOp(${STRFTIME(,,)})) in their dialplan that asterisk returns GMT time rather than the systems localtime? I'll starting to think I'm going crazy. |
| 04:47.00 | harryvv | TVDVR |
| 04:47.04 | JT | linux PVR app, CunningPike |
| 04:47.06 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:47.06 | lmpbzktwn5 | cunningpike: DIY tivo |
| 04:47.15 | harryvv | tivo |
| 04:47.16 | harryvv | :) |
| 04:47.21 | CunningPike | Ah - pardon my ignorance :) |
| 04:47.23 | lmpbzktwn5 | but more kick-ass |
| 04:47.37 | TrickFinlay2 | CunningPike: we're trying to setup hte ulimate dorm room |
| 04:48.00 | CunningPike | So you're looking to send RTP video? |
| 04:48.02 | *** join/#asterisk constfilin (n=cf@ppp-71-139-98-220.dsl.snfc21.pacbell.net) |
| 04:48.15 | lmpbzktwn5 | we were actually looking for a way to run mythTV on our linux (fedora core 5) box, but have it double as an answering machine running asterisk, and possibly using an OSD with CID info |
| 04:48.48 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 04:49.04 | lmpbzktwn5 | the only thing we don't know is how the hell to get it all together |
| 04:49.37 | CunningPike | lmpbzktwn5: Hmmmmm - I've always kept other processes away from my asterisk servers, but ymmv |
| 04:49.38 | lmpbzktwn5 | as of now, we have two separate machines for each. one for mythTV, another for asterisk... kind of want to combine the two into one machine |
| 04:49.49 | *** part/#asterisk Skarmeth (n=Skarmeth@201009084250.user.veloxzone.com.br) |
| 04:49.59 | joe | http://www.voip-info.org/wiki/view/Asterisk+tips+MythTV+integration |
| 04:50.03 | lmpbzktwn5 | take up that many resources? or simply for quality issues? |
| 04:50.07 | joe | seen that? |
| 04:50.23 | TrickFinlay2 | checking it now,thanks joe |
| 04:50.38 | constfilin | Hello, has anybody tried to use iptables for efficient routing of RTP packets within asterisk ? |
| 04:50.54 | hads | Quality issues normally as when Asterisk wants resources it wants resources. If it's just a glorified answering machine then it shouldn't be a problem. |
| 04:50.59 | joe | constfilin: efficient routing? |
| 04:51.00 | lmpbzktwn5 | joe: i was looking for something like that... |
| 04:51.05 | joe | lmpbzktwn5: :) |
| 04:51.32 | lmpbzktwn5 | joe: any experience with that? |
| 04:52.12 | joe | lmpbzktwn5: I run both * and mythtv but haven not made them do anything described as such just read about it |
| 04:52.16 | joe | been too busy... |
| 04:53.02 | lowlevel | what is everyones fascination with TV!? ;) |
| 04:53.11 | joe | constfilin: do you mean like QOS sort of thing? |
| 04:53.22 | lowlevel | what is with rather. |
| 04:53.26 | lmpbzktwn5 | lowlevel: who knows. it's just cool to have. |
| 04:53.35 | lowlevel | I guess... |
| 04:53.40 | CunningPike | lowlevel: They're students ;) |
| 04:53.50 | lowlevel | ohhhhhhhhhhhhhhhh students. |
| 04:53.50 | constfilin | I mean setup source nat and destination nat in iptables so that RTP packets are routing right in the kernel instead of going through the user space and ast_channel_bridge. |
| 04:54.01 | TrickFinlay2 | hey whats wrong with being students :p |
| 04:54.11 | CunningPike | TrickFinlay2: Nothing at all |
| 04:54.23 | lowlevel | nothing, just they only got room for 1 box... have no money to buy good hardware, etc. |
| 04:54.52 | CunningPike | What do you study? |
| 04:55.10 | lmpbzktwn5 | joe: so it would be better to have two separate boxes and have the * system send it to the myth box |
| 04:55.20 | lmpbzktwn5 | cunningpike: i'm going for CS |
| 04:55.28 | russellb | yay students |
| 04:55.32 | TrickFinlay2 | CunningPike: ill be a freshmen at RIT in computer networkind and sec. |
| 04:55.38 | TrickFinlay2 | *networking |
| 04:55.43 | CunningPike | TrickFinlay2: Nice |
| 04:55.54 | russellb | <-- senior in computer engineering ... |
| 04:55.54 | TrickFinlay2 | thanks |
| 04:55.58 | joe | lmpbzktwn5: depends on the box, and what you'll be doing, sorta hard to answer but imho having different systems tends to be easier... |
| 04:56.29 | CunningPike | lmpbzktwn5: CS? |
| 04:56.36 | TrickFinlay2 | comp sci |
| 04:56.45 | lmpbzktwn5 | ^^ beat me to it |
| 04:56.49 | CunningPike | Ah - that's what they called it on the old days ;) |
| 04:56.49 | joe | lmpbzktwn5: ie if you break one you still have the other :) |
| 04:57.14 | constfilin | Hey, so anybody played with iptables? |
| 04:57.19 | lmpbzktwn5 | joe: yeah that's true. just figured it might be easier and less space consuming had it been in one box |
| 04:57.34 | TrickFinlay2 | would a pII 350/384ram be enough to run Asterisk? |
| 04:57.43 | lmpbzktwn5 | joe: although, no big deal if it's in two... we'll just hide the asterisk box somewhere |
| 04:58.29 | lmpbzktwn5 | joe: no monitor needed for the * box though right? can it all be managed through a web-interface or ssh or something like that? |
| 04:58.39 | joe | yes |
| 04:58.40 | JT | you could build mini-itx computers into your furniture |
| 04:58.45 | joe | hehe |
| 04:58.46 | TrickFinlay2 | ahaha |
| 04:58.52 | lmpbzktwn5 | haha |
| 05:00.37 | lmpbzktwn5 | http://base.google.com/base/a/482424/D9722739606059606145 |
| 05:00.41 | lmpbzktwn5 | good buy? |
| 05:01.39 | *** join/#asterisk bintut (n=bintut@cable-202-8-251-159.d-one.net) |
| 05:01.51 | bintut | hello all.. |
| 05:02.19 | bintut | i'm thinking to get one unit of this product ==> http://www.mediatrix.com/products_devices.php?prodid=3 |
| 05:02.33 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 05:02.43 | [TK]D-Fender | bintut : pretty decent unit |
| 05:03.05 | [TK]D-Fender | bintut : easy enough to set up. Does all the basics. |
| 05:03.05 | bintut | but i can't find a local reseller in our country |
| 05:03.28 | bintut | that's why, i might look for an alternative product which is available in our place |
| 05:04.08 | bintut | any idea how much does it cost? |
| 05:05.52 | bintut | anyone here was able to purchase the Mediatrix 1124 - 24-port FXS VoIP Access Device? how much does it cost? |
| 05:06.21 | [TK]D-Fender | bintut : Look for the AudioCodes MP-124 as well for comparison |
| 05:06.35 | *** part/#asterisk jake1932 (n=Administ@pool-70-16-129-225.phil.east.verizon.net) |
| 05:07.56 | bintut | ok |
| 05:08.56 | bintut | i believe that the mediatrix 1124 is the device i'm looking for.. i'm planning to connect my existing analog connections to that device but the fxo side is now ethernet going to an asterisk box |
| 05:10.09 | CunningPike | Is there a good device for connecting multiple FXO ports to ethernet? |
| 05:10.20 | CunningPike | ~fxosfxa |
| 05:10.25 | CunningPike | ~fxofxs |
| 05:10.27 | jbot | somebody said fxofxs was An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 05:10.52 | CunningPike | Is there a good device for connecting multiple FXS ports to ethernet? |
| 05:10.54 | CunningPike | :D |
| 05:10.59 | *** join/#asterisk ANTILOCAS (i=wewrwe@200.87.89.209) |
| 05:11.16 | Kerry_G | has anyone used Hamachi on an Asterisk box? |
| 05:11.18 | hads | How many is multiple |
| 05:11.25 | x86 | CunningPike: asterisk box? :) |
| 05:11.37 | CunningPike | hads: About 8 |
| 05:11.48 | Kerry_G | Astribank |
| 05:12.11 | CunningPike | x86: I would like to connect a number of devices to Asterisk without a single port ATA for each one |
| 05:12.11 | lmpbzktwn5 | hey thanks all for your help-- we'll be back once we get it all together and let you all know how it went ;) |
| 05:12.15 | lmpbzktwn5 | thanks again |
| 05:12.19 | bintut | CunningPike: isn't it like the mediatrix 1124 device that i'm looking for? |
| 05:12.31 | CunningPike | bintut: Yes - but a little smaller |
| 05:12.38 | CunningPike | That's a 24-port, right? |
| 05:13.08 | hads | http://www.voipsupply.com/index.php?cPath=96_120 |
| 05:13.12 | hads | There's some |
| 05:13.42 | CunningPike | hads: Thanks |
| 05:13.43 | hads | Acutally, there are some 8 port ones; http://www.voipsupply.com/index.php?cPath=96_121 |
| 05:13.59 | CunningPike | hads: Cool - thanks |
| 05:14.04 | hads | Is anyone here running trunk? |
| 05:14.08 | bintut | CunningPike: yes |
| 05:14.19 | russellb | i'm running trunk!!! |
| 05:14.50 | hads | russellb: I don't want to bug you though, you do important stuff. :) |
| 05:15.02 | CunningPike | russellb: You are? :O I thought you were the stable king....... ;) |
| 05:15.16 | russellb | heh |
| 05:15.22 | russellb | i run 1.2 on stuff that matters |
| 05:15.38 | bintut | [TK]D-Fender: still there? i think, the audiocodes mp-124d is an alternative device |
| 05:15.40 | russellb | but most of the stuff I have my hands in administrating are development platforms |
| 05:16.24 | bintut | [TK]D-Fender: do you have a personal experience with audiocodes mp-124d? is it good enough to support 24port fxs? |
| 05:16.56 | x86 | CunningPike: get a channel bank and get a T1 card, run a T1 to the channel bank |
| 05:17.17 | CunningPike | x86: We thought of that, but our PRI card is full |
| 05:17.33 | CunningPike | x86: Actually, we have one free port, now that I think..... |
| 05:17.37 | JT | get another card? :) |
| 05:17.45 | *** join/#asterisk vlrk (n=root@202.65.134.119) |
| 05:18.18 | x86 | get another quad port card, you know, to make room for "future expansion" :P |
| 05:18.42 | [TK]D-Fender | bintut : Both the AudioCodes & MediaTrix work pretty well. |
| 05:19.34 | [TK]D-Fender | Audiocodes is more confusing to set up, but easier to expand with and has internal redundancy features the MediaTrix doesn't do as well. But for a basic * install I'd prefer Mediatrix. |
| 05:19.40 | vlrk | my asterisk crashed and while doing gdb the last frame it showed is at ast_set_write_format (chan=0x9d96db0, fmts=64) at channel.c:1710 |
| 05:20.31 | hads | russellb: I'm guessing that stdtime/localtime.c is where Asterisk is meant to pickup your local timezone? |
| 05:20.54 | russellb | no, asterisk gets your local timezone from the system |
| 05:20.58 | russellb | your /etc/localtime setting |
| 05:21.12 | bintut | [TK]D-Fender: ok. thanks. we have a local reseller for audiocodes mp-124d. we'll try to ask for a demo unit first before we decided to purchase one. thanks again. :) |
| 05:21.25 | [TK]D-Fender | bintut : Np. |
| 05:22.29 | TrickFinlay2 | guys, is there a recommended OS for asterisk? |
| 05:22.30 | hads | Hmm... I'm going bonkers trying to figure this out. logs and crds are using the systems local timezone but STRFTIME and SayUnixTime and Voicemail are using GMT. The only difference I can see is that logs and cdrs use localtime_r directly where the others use stdtime/localtime.c |
| 05:23.06 | *** join/#asterisk lmpbzktwn5 (n=lmpbzktw@24-151-139-231.dhcp.oxfr.ma.charter.com) |
| 05:23.15 | CunningPike | TrickFinlay2: Linux |
| 05:23.16 | CunningPike | :D |
| 05:23.28 | *** join/#asterisk num000 (n=numerobi@e177185080.adsl.alicedsl.de) |
| 05:23.53 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 05:23.56 | TrickFinlay2 | CunningPike: haha thanks,any spec. distro? |
| 05:24.16 | CunningPike | TrickFinlay2: :D Your favorite one - seriously |
| 05:24.36 | CunningPike | TrickFinlay2: The more comfortable you are with your distro, the easier it will be |
| 05:24.37 | russellb | hads: you have to configure the timezone in voicemail |
| 05:24.42 | num000 | what das the 404 Not Found mean? |
| 05:24.47 | russellb | hads: as for the others ... i'd have to look at the code |
| 05:24.50 | TrickFinlay2 | CunningPike:alright thanks |
| 05:24.52 | CunningPike | num000: Exactly what it says |
| 05:24.58 | [TK]D-Fender | TrickFinlay2 : I'd suggest something relatively standard that you can easily get with the common devel packages. Debin, Slackware, RHEL, CentOS.... |
| 05:25.09 | lmpbzktwn5 | hhahaha |
| 05:25.21 | CunningPike | TrickFinlay2: What is your favorite distro (asks just in case it's mandrake or something) |
| 05:25.21 | TrickFinlay2 | maybe Xubuntu? |
| 05:25.33 | num000 | i was trying to install an echo service but i do get this 404 Not found error |
| 05:25.44 | TrickFinlay2 | CunningPike: how about xunbuntu |
| 05:25.51 | [TK]D-Fender | TrickFinlay2 : I wouldn't suggest it personally.... way too many packages to download, plus the way it changes root access..... |
| 05:25.52 | TrickFinlay2 | * |
| 05:25.54 | TrickFinlay2 | Xubuntu |
| 05:26.22 | joe | TrickFinlay2: CentOS works great |
| 05:26.22 | CunningPike | TrickFinlay2: Actually, I'm not a *buntu fan - we use RHEL, some people like Debian |
| 05:26.25 | hads | russellb: Nope, I haven't configured the time in voicemail, I'm guessing that that would work but I figured that it should fall back to the systems localtime by default. I was going to figure that out after I figured out why it's not working for the dialplan functions. |
| 05:26.30 | TrickFinlay2 | so in that case prob slack,wheres its a p2/384 ram |
| 05:26.44 | CunningPike | TrickFinlay2: If you must :P |
| 05:26.51 | [TK]D-Fender | TrickFinlay2 : Slack = 0 troubles for me. |
| 05:27.02 | num000 | CunningPike what does the 404 not found mean in the contents of a echo-service? |
| 05:27.06 | TrickFinlay2 | alright ill look into CentOS/slack |
| 05:27.18 | russellb | hads: gotcha ... |
| 05:27.19 | ANTILOCAS | hey wich linux is better for Asterisk? red hat, mandrake, suse? |
| 05:27.26 | [TK]D-Fender | ok, check-out time for me here, later all. |
| 05:27.29 | CunningPike | num000: pastebin your extensions.conf |
| 05:27.31 | TrickFinlay2 | [TK]D-Fender: when is ver 11 due out for slack |
| 05:27.33 | CunningPike | ~pb |
| 05:27.34 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 05:27.47 | [TK]D-Fender | ANTILOCAS : Avoid SUSE, it make compiling kernal modules hell.... |
| 05:28.01 | ANTILOCAS | is Red Hat still alive? |
| 05:28.06 | hads | russellb: I've been going through and putting some debugging statements in localtime.c and figured that ast_tzset_basic is getting called with an empty string so is executing the block under * Name is set, but set to the empty string == no adjustments */ |
| 05:28.06 | [TK]D-Fender | TrickFinlay2 : "When its done" |
| 05:28.06 | russellb | hads: STRFTIME takes an argument for the timezone |
| 05:28.13 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 05:28.22 | CunningPike | ANTILOCAS: Very much so |
| 05:28.28 | TrickFinlay2 | is that from the site? |
| 05:28.29 | ANTILOCAS | TKD FEnder which linux u use and what version then |
| 05:28.34 | [TK]D-Fender | ~8ball Will Slackware 11 be released really soon? |
| 05:28.36 | jbot | I'm not sure. |
| 05:28.49 | russellb | hads: are you providing a timezone arg to the function? |
| 05:28.53 | hads | russellb: I know, and that _would_ be the obvious thing to do... but it says that it's optional and I was trying to find out why it doesn't work :) |
| 05:28.55 | TrickFinlay2 | haha |
| 05:29.01 | [TK]D-Fender | ANTILOCAS : I run mine on Slackware 10.2, Work is on FC3, most of my customers are on CentOS. |
| 05:29.16 | russellb | hads: optional, and if not provided, you get GMT :) |
| 05:29.20 | harryvv | slackware was my first linux os in 97 |
| 05:29.32 | ANTILOCAS | cunninpike and wichi linux u use? |
| 05:29.46 | CunningPike | ANTILOCAS: RHEL or CentOS |
| 05:29.48 | hads | russellb: It works if you specify the timezone, but gives GMT if you don't. I think it's meant to give localtime if you don't. |
| 05:30.02 | russellb | hads: well the documentation doesn't say either way |
| 05:30.10 | CunningPike | ANTILOCAS: And I prefer CentOS |
| 05:30.16 | ANTILOCAS | and red hat? |
| 05:30.24 | russellb | hads: but the code clearly intends to provide GMT with no timezone arg |
| 05:30.36 | CunningPike | ANTILOCAS: RHEL is RedHat Enterprise Linux |
| 05:30.50 | bintut | gtg now.. |
| 05:30.51 | bintut | thanks |
| 05:30.54 | CunningPike | CentOS is the 'free' version of RHEL |
| 05:30.55 | ANTILOCAS | ok thnks |
| 05:30.57 | harryvv | CunningPike thats what I use right now. |
| 05:31.05 | CunningPike | harryvv: CentOS? |
| 05:31.07 | hads | russellb: OK :) but the sayunixtime doc says that it defaults to the machine default timezone. |
| 05:31.08 | harryvv | yup |
| 05:31.23 | ANTILOCAS | RHEL was alwasy free for download as far as i remember |
| 05:31.25 | russellb | hads: ok, let me look at that one |
| 05:31.29 | ANTILOCAS | i installed the RHEL 7.1 |
| 05:31.30 | harryvv | but I love fedora |
| 05:31.33 | CunningPike | harryvv: Yes - our Asterisk servers are RedHat, but all the other stuff is CentOS |
| 05:31.39 | Juggie | theres no such thing as RHEL7.1 |
| 05:31.52 | ANTILOCAS | Red Hat i meant |
| 05:32.01 | ANTILOCAS | u mean it changed the name? |
| 05:32.08 | Juggie | rh7.1 is about 5 years old. |
| 05:32.13 | pfn | who the hell uses rh7 still? |
| 05:32.22 | pfn | rh9's been eol since like 2003 hasn't it? |
| 05:32.24 | hads | russellb: I think both of those apps/functions come back to ast_localtime, which calls ast_tzset |
| 05:33.08 | ANTILOCAS | im planning to install asterisk thats why i ask, then i will download RHEL right? |
| 05:33.27 | ANTILOCAS | or is not free anymore? |
| 05:33.44 | ANTILOCAS | whats the new page to download for free? |
| 05:33.58 | pfn | if you want rhel, go with centos |
| 05:34.05 | pfn | if you want plain ol' redhat, use fedora core |
| 05:34.06 | num000 | CunningPike here is the pastepin link to the extentions.conf http://channels.debian.net/paste/3468 |
| 05:34.14 | CunningPike | ANTILOCAS: I second that - use CentOS |
| 05:34.40 | num000 | CunningPike thank you very much in advance |
| 05:34.54 | CunningPike | num000: So, you are dialing '81' ? |
| 05:34.59 | Juggie | num000, did you solve your problem from today? |
| 05:35.00 | num000 | yes |
| 05:35.10 | Juggie | the missing libc function? |
| 05:35.14 | Juggie | what fixed it? |
| 05:35.16 | num000 | Juggie ohh yes i did |
| 05:35.22 | Juggie | the libc update i linked you to, or? |
| 05:35.33 | num000 | it wasn't the libc it was a broken package with the libncurses |
| 05:35.53 | Juggie | ah, so you just updated/reinstalled libncurses? |
| 05:35.58 | num000 | so i used a different package |
| 05:36.03 | Juggie | cool |
| 05:36.04 | num000 | Juggie yes, |
| 05:36.10 | russellb | hads: this time code hurts my eyes :) |
| 05:36.10 | Juggie | good stuff |
| 05:36.13 | Juggie | whats your problem now |
| 05:36.17 | num000 | Juggie it actually works very well |
| 05:36.25 | CunningPike | num000: OK, so now pastebin your CLI output |
| 05:36.30 | num000 | i was playing around a bit |
| 05:36.40 | hads | russellb: I know! I've been trying to figure this out for at least 8 hours. |
| 05:36.48 | num000 | CunningPike would be a verbosity of 3 enough? |
| 05:37.14 | russellb | hads: I'll tell you what ... the author of both STRFTIME() and SayUnixTime() is Corydon. You should try to catch him on IRC and find out the intended behavior |
| 05:37.29 | CunningPike | num000: Should be - let's go with that and see what it says |
| 05:37.39 | hads | russellb: OK, I'll do that. Thanks for taking the time :) |
| 05:37.46 | russellb | hads: from looking at this a little bit, it looks like you may be right |
| 05:37.53 | russellb | but i don't want to render a judgement on it quite yet |
| 05:38.11 | russellb | hads: sorry you've had so much trouble .. |
| 05:38.12 | hads | Yeah, I'm hoping it's not something dumb that I've done. |
| 05:38.56 | hads | It's no trouble - if something bugs me then I'll try and figure it out - sometimes I get a bit carried away and don't stop :/ |
| 05:39.06 | num000 | CunningPike this is the paste for the cli |
| 05:39.06 | num000 | daedalus-gw*CLI> |
| 05:39.07 | num000 | daedalus-gw*CLI> |
| 05:39.52 | CunningPike | I said pastebin, right? |
| 05:39.56 | CunningPike | :/ |
| 05:40.10 | russellb | lol |
| 05:40.10 | *** join/#asterisk num000 (n=numerobi@e177185080.adsl.alicedsl.de) |
| 05:40.15 | num000 | i'm really sorry |
| 05:40.17 | ManxPower | ~pastebin |
| 05:40.18 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 05:40.22 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:40.39 | num000 | http://channels.debian.net/paste/3469 |
| 05:40.49 | hads | russellb: Do you know what timezone Corydon is in? |
| 05:40.58 | num000 | CunningPike sorry, http://channels.debian.net/paste/3469 |
| 05:41.01 | file | CST |
| 05:41.10 | hads | file: Thanks! |
| 05:41.29 | ManxPower | GMT-6 or GMT-7 I don't recall |
| 05:41.39 | CunningPike | Croydon is a borough of London: GMT |
| 05:43.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:43.47 | num000 | CunningPike could it be that it is a conflict with the codecs? |
| 05:43.53 | CunningPike | num000: Is your UA registered properly? What does 'sip show peers' say |
| 05:44.00 | CunningPike | 404 means that it can't find the UA |
| 05:44.45 | num000 | 100/100 192.168.1.20 D N 255.255.255.255 5060 Unmonitored |
| 05:44.55 | num000 | CunningPike it is registered actually |
| 05:45.03 | num000 | calls out and in do work ok |
| 05:45.33 | ANTILOCAS | I have an Amd Athlon 64 3000+, which kind of centos should i donwload then? |
| 05:45.53 | ANTILOCAS | ia64? |
| 05:46.14 | ANTILOCAS | x86_64? |
| 05:46.30 | *** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir) |
| 05:47.38 | num000 | mhh |
| 05:48.04 | *** part/#asterisk pcm (n=pcm@72.146.59.132) |
| 05:48.27 | ANTILOCAS | cunninpike |
| 05:48.36 | num000 | Juggie are you still awake or again? |
| 05:48.40 | CunningPike | num000: It could be a codec issue - does your phone support gsm? |
| 05:48.54 | num000 | CunningPike i suppose, |
| 05:49.01 | num000 | CunningPike how could i test this? |
| 05:49.20 | harryvv | since its late this is for people enertainment :) I come from a jet engine background. |
| 05:49.23 | harryvv | http://video.google.com/videoplay?docid=-5417019303200331106 |
| 05:49.30 | num000 | CunningPike which codes is used when the echo answers |
| 05:49.57 | CunningPike | The codec of the file (gsm in this case) |
| 05:50.20 | CunningPike | Try disallow=all and then allow=gsm in your sip.conf and see if your phone breaks |
| 05:50.31 | num000 | CunningPike should i add something like allow=gsm ?? |
| 05:50.43 | num000 | ;) |
| 05:50.58 | ANTILOCAS | I have an Amd Athlon 64 3000+, which kind of Centos 4.0 should i donwload then? |
| 05:51.25 | mrfrenzy | I don't know centos, but I'd recommend the amd64-version of debian |
| 05:51.47 | CunningPike | num000: That's what I sid |
| 05:51.50 | ANTILOCAS | frenzy is it free? |
| 05:51.55 | mrfrenzy | yes |
| 05:52.13 | mrfrenzy | http://www.debian.org/distrib/ |
| 05:52.14 | ANTILOCAS | is it easy to install asterisk there? |
| 05:52.18 | mrfrenzy | apt-get install asterisk |
| 05:52.59 | ANTILOCAS | thnks |
| 05:53.14 | mrfrenzy | np |
| 05:54.05 | num000 | CunningPike thank you very much, i'm just trying it. |
| 05:54.10 | CunningPike | num000: OK |
| 05:55.59 | num000 | CunningPike no it says that there is no compatible codecs and ends with 488 not acceptable here |
| 05:56.16 | CunningPike | Aha - there's your problem then |
| 05:56.31 | num000 | how would i have to solve this? |
| 05:56.36 | CunningPike | num000: Either get your phone to do gsm, or play a file with a different codec |
| 05:57.07 | harryvv | num where are you |
| 05:57.07 | num000 | CunningPike so there must be a file laying somewhere which is encoded with gsm compression right? |
| 05:57.20 | num000 | harryvv how you mean? |
| 05:57.24 | harryvv | location |
| 05:57.29 | num000 | which country? ohh germany |
| 05:57.33 | CunningPike | num000: Not by default - you will have to record them |
| 05:57.58 | CunningPike | num000: Ooo - or check the list archives - someone has created all the sounds in ulaw |
| 05:58.08 | num000 | CunningPike can i place it somewhere and have him play it then |
| 05:58.44 | CunningPike | num000: Yes - put it in /var/lib/asterisk/sounds and then change the filename in extensions.conf |
| 05:58.56 | *** join/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 05:59.09 | num000 | CunningPike how do i reference to the file in extensions.conf? |
| 05:59.44 | CunningPike | num000: Change 'exten => 81,3,playback,demo-echotest' to 'exten => 81,3,playback,foo' |
| 05:59.52 | CunningPike | num000: Where foo is the name of your new file |
| 06:00.10 | num000 | CunningPike ok, i just see that this directory is empty here |
| 06:00.18 | ontae | Hi, may anyone help me with an RTP-stream problem when calling out? |
| 06:00.23 | num000 | maybe it can do the codec but the file is just missing |
| 06:00.44 | num000 | harryvv why did you ask for my location? |
| 06:00.50 | CunningPike | num000: /var/lib/asterisk/sounds is empty? |
| 06:00.58 | num000 | CunningPike yes |
| 06:01.28 | num000 | CunningPike it is actually lying somewhere else here on this distribution /usr/lib/asterisk/sounds but this direcotory is empty |
| 06:02.07 | CunningPike | num000: Oh, OK - well, wherever demo-echotest.gsm is, then |
| 06:04.16 | harryvv | num000 yes |
| 06:04.25 | harryvv | part of the world |
| 06:04.32 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 06:04.43 | num000 | harryvv where are you? |
| 06:04.51 | harryvv | dela bc |
| 06:05.02 | num000 | dela bc? where is it? |
| 06:05.26 | harryvv | delta |
| 06:09.21 | num000 | ahh the sound files are stored in a different package, but if i would use all this my router which runs asterisk would run out of space, maybe just this one file would be ok |
| 06:10.02 | CunningPike | num000: OK |
| 06:10.57 | *** join/#asterisk ionix (n=ionix@p3101-ipbfp05miyazaki.miyazaki.ocn.ne.jp) |
| 06:11.13 | ionix | hey, anyone can try 18003765501 and tell me if it works? I am from Japan and cannot dial those numbers |
| 06:13.08 | harryvv | It that a regional 1800 number? |
| 06:13.20 | harryvv | or a country or world wide? |
| 06:13.34 | harryvv | because its not aviable here. |
| 06:13.54 | JT | isn't it one digit too long? |
| 06:14.33 | CunningPike | JT: Rub your eyes |
| 06:14.36 | ionix | damn |
| 06:14.42 | ionix | it's US+CAN |
| 06:14.52 | JT | CunningPike: excuse me? |
| 06:15.08 | CunningPike | JT: It has the right number of digits for me........ |
| 06:15.11 | ionix | JT: no, 1 800 376 5501 |
| 06:15.17 | CunningPike | JT: Maybe it's my eyes |
| 06:16.09 | JT | okay, i thought the NANP was 10 chars for some reason |
| 06:16.13 | JT | maybe excluding the 1 |
| 06:16.49 | CunningPike | JT: Long distance is 11 - 1-NXX-NXX-XXX |
| 06:16.52 | CunningPike | Crap |
| 06:16.58 | CunningPike | You know what I meant lol |
| 06:17.41 | ontae | Hi, may anyone help me with an RTP-stream problem when calling out? |
| 06:17.47 | JT | i'm not from the US |
| 06:18.00 | JT | so it's not possible to call any 1-800s from outside the US? |
| 06:18.25 | ionix | well you can use FWD |
| 06:18.35 | JT | yes, but via the pstn |
| 06:18.43 | ionix | not unless it's international |
| 06:18.56 | CunningPike | JT: Not usually - it would have to be dialed as an international number |
| 06:18.57 | ionix | but international has 800 + 8digits |
| 06:19.07 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 06:19.15 | JT | so are some 1-800s international? |
| 06:19.19 | JT | so it wouldn't work? :) |
| 06:19.41 | ionix | CunningPike. Not even, from Japan, you can't reach a 1-800 number by making an international call |
| 06:20.16 | ionix | JT: It will if the company uses a global 800 number |
| 06:20.17 | ionix | like 800 1234 5678 |
| 06:20.23 | harryvv | Im going to crash |
| 06:20.32 | harryvv | night |
| 06:20.46 | JT | ionix: complicated now |
| 06:20.48 | JT | heh |
| 06:22.55 | lowlevel | mmm, anyone know what would cause the message 'phone off hook in weird state 3??' when answering a call on an analog line? It happened a few times and now its fine again... |
| 06:23.26 | lowlevel | night pike. |
| 06:23.38 | JT | i thought north american numbers in international format were meant to be +1 XXX XXX XXXX? |
| 06:23.58 | lowlevel | jt; they are, we commonly just leave out the 1. |
| 06:24.23 | lowlevel | er nevermind |
| 06:24.42 | JT | i understand the 1 is not needed within north america |
| 06:25.02 | JT | i don't understand how < ionix> like 800 1234 5678 would fit in |
| 06:25.28 | *** part/#asterisk vlrk (n=root@202.65.134.119) |
| 06:25.36 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 06:27.09 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 06:28.20 | lowlevel | jt: the 1 is needed when dialing, its just not written in, its assumed |
| 06:28.21 | num000 | does anyone know where a list of sip errorcodes is? like 404 etc? |
| 06:29.02 | mcnobody | Is it possible to allow all SIP calls from unknown callers to local SIP peers? |
| 06:29.04 | mcnobody | allowguest=yes, makes it almost. All calls with locally unknown user part in From:-header are accepted, but calls from other Asterisk with same user part of From:-header are tried to authenticate against local peer. |
| 06:30.26 | JT | lowlevel: ok, well what i was getting at was ionix's 800 number example is 1 character too long for the normal XXX XXX XXXX mask |
| 06:33.03 | lowlevel | jt: mask? what mask? ;) whats stopping you from putting in the 1? |
| 06:33.37 | lowlevel | I think Im missing the question. |
| 06:33.40 | lowlevel | rofl |
| 06:34.48 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 06:36.26 | lowlevel | jt; if you're trying to dial an 800 number from japan, you can't do that. |
| 06:36.38 | lowlevel | infact, some are invalid between canada and the us. |
| 06:36.49 | lowlevel | same applies to 888, 900, etc. |
| 06:36.52 | ionix | but the canada/usa is based on call id :) |
| 06:36.56 | ionix | not trunk |
| 06:37.02 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:37.15 | lowlevel | ionix: always? |
| 06:37.35 | ionix | yes |
| 06:37.37 | lowlevel | *shrug* doesn't really matter, he can't dial it direct |
| 06:37.53 | JT | lowlevel: the NANP, aren't numbers meant to be 10 characters after the +1? |
| 06:38.08 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 06:39.31 | lowlevel | jt: the 800 number will terminate or forward (forget the correct terminology) on a regular number, such as 416-722-2223 |
| 06:39.42 | lowlevel | you would require that number instead to dial from outside of +1 |
| 06:39.55 | *** join/#asterisk fafnir_ (i=hahaha@unaffiliated/fafnir) |
| 06:40.13 | JT | ok, right |
| 06:40.45 | JT | ionix's example of an 800 number with 11 characters threw me off |
| 06:40.52 | lowlevel | jt; the point is, 800, 888, 900, etc are 'special' and not accessable from outside of north america |
| 06:40.57 | lowlevel | ok |
| 06:41.09 | JT | i don't know if it's even valid in the us |
| 06:41.15 | lowlevel | yeah probably is |
| 06:41.20 | JT | hrm |
| 06:41.32 | JT | what are 888 and 900? |
| 06:41.45 | lowlevel | ohhhhh crap |
| 06:42.05 | lowlevel | what number was it? I see one he put in that is too long... |
| 06:42.19 | lowlevel | 888 is the same as 800, it was created when they ran out of room in 800 |
| 06:42.26 | ionix | 888/877/866 |
| 06:42.35 | lowlevel | yeah, all 3 |
| 06:43.18 | james_ | lowlevel: i called +1 866 xxxxxx today from australia |
| 06:43.24 | james_ | just call charges apply |
| 06:43.36 | lowlevel | james; interesting |
| 06:43.55 | james_ | hangon, i'll dial again, tell you what it says |
| 06:44.04 | JT | james_: do you have a sip or iax trunk in the US or similar? |
| 06:44.15 | ionix | it's called UIFN |
| 06:44.17 | james_ | nope, this is from australian pstn |
| 06:44.23 | JT | ah ok |
| 06:44.30 | ionix | UIFN uses ITU country code 800, so that no matter where the caller is, only the international access code (IAC) and the 8-digit UIFN need to be dialed. Currently, about 30 countries participate in the UIFN programme |
| 06:44.33 | ionix | so the country code is 800 |
| 06:44.53 | lowlevel | ahhhh |
| 06:44.56 | james_ | "access to the 800 number you have dialled is not free of charge from outside of the US" |
| 06:44.56 | lowlevel | ;) |
| 06:44.59 | JT | but he said he called +1, ionix? |
| 06:45.03 | james_ | i dial with a country code of +1 |
| 06:45.16 | james_ | +1 866 230 0800 |
| 06:45.28 | num000 | are you still talking about ionix toll free number? ;) |
| 06:45.29 | james_ | layered technologies in texas |
| 06:45.37 | docelmo | You cant dial PSTN 1+ Toll Free when your CIC is coming from outside the country |
| 06:45.39 | JT | yes num000 |
| 06:45.49 | JT | because everyone is contradicting each other :P |
| 06:45.55 | num000 | cool |
| 06:45.59 | docelmo | You can however do it from withing the states from a US provider |
| 06:46.20 | lowlevel | I'm thinking the rules arn't as hard and fast as I thought they were |
| 06:46.21 | lowlevel | ;) |
| 06:46.37 | JT | docelmo: i would've thought that'd be obvious to most |
| 06:46.53 | docelmo | You would be amazed at the amount of morons out there |
| 06:46.54 | james_ | lowlevel: $ makes everything possible :P |
| 06:46.57 | JT | the whole international calls costing money thing |
| 06:46.58 | JT | true |
| 06:47.12 | JT | james_: telstra pstn? |
| 06:47.25 | james_ | powertel |
| 06:47.32 | docelmo | Telstra sucks PRIMUS rules |
| 06:47.38 | orlock | nah |
| 06:47.40 | orlock | they both suck |
| 06:47.43 | orlock | go NXT! |
| 06:47.52 | lowlevel | heh |
| 06:47.54 | docelmo | I have colo's with Primus in Melbourne and Sydeny |
| 06:47.59 | james_ | pretty much every .au provider of everything sucks :P |
| 06:48.08 | orlock | docelmo: King St in Melb? |
| 06:48.15 | orlock | james_: nextep doesnt :) |
| 06:48.20 | docelmo | I would have to find the addy but that sounds right |
| 06:48.26 | docelmo | I ahve 60 E1's there |
| 06:48.31 | JT | nextep sucks money actually |
| 06:48.37 | orlock | james_: dslam's and modems and voip gear all r+d is done in melbourne for them :) |
| 06:48.39 | james_ | collins st i bet |
| 06:48.50 | james_ | orlock: sounds like they should give me a job |
| 06:48.56 | orlock | james_: seek.com.au |
| 06:49.00 | james_ | hahaha |
| 06:49.13 | james_ | yeah, i get an email from them each day |
| 06:49.18 | james_ | filtering voip + melbourne |
| 06:49.56 | orlock | james_: can you do firmware coding for linux, test people, etc |
| 06:50.05 | docelmo | Well with jobs beggers cant be choosy.. Find something till something better comes along |
| 06:50.22 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
| 06:50.43 | james_ | orlock: i *could*, not really what i'm lookign for though |
| 06:50.46 | james_ | i'm a developer |
| 06:50.46 | Qwell | docelmo: it took me 5 years until something better came along :p |
| 06:50.57 | orlock | james_: yeah, this is doing firmware development |
| 06:50.59 | docelmo | You call Digium better? ACK! |
| 06:51.07 | Qwell | docelmo: by far |
| 06:51.07 | james_ | haha |
| 06:51.16 | orlock | james_: they are also doing voip |
| 06:51.39 | docelmo | Qwell How's Hickville? |
| 06:51.42 | james_ | what sort of devices is the firmware dev for? |
| 06:52.33 | docelmo | I was thinking about coming there for a job.. but then I thought Alabama.. uhh naa.. |
| 06:52.55 | lowlevel | k, gotta stop messing with dialplan and go to bed |
| 06:53.00 | lowlevel | gawd |
| 06:53.01 | lowlevel | hahah |
| 06:53.03 | lowlevel | night guys ;) |
| 06:54.35 | orlock | james_: comms devices |
| 06:54.53 | orlock | james_: they mentioned network termination devices.. that office is NEC's center for dsl research |
| 06:55.03 | orlock | thats one of the reasons why nextep kicks ass so much |
| 06:55.22 | orlock | all of the voip gear.. dslam's, etc is pretty much made by them |
| 06:55.28 | Dico_ | hello everybody :) |
| 06:55.28 | orlock | and they run the ISP that suppotrs it |
| 06:55.40 | *** join/#asterisk vlt (n=dm@p54B33FE8.dip0.t-ipconnect.de) |
| 06:56.10 | orlock | JT: nextep suck money cos when things break, you can call up and they will get fixed |
| 06:56.19 | orlock | none of this telstra bullshit "sometime in the next few days" |
| 06:56.30 | Dico_ | has anyone already tryed to use queue then transfert the queue to meetme ? |
| 06:57.48 | orlock | oh@ |
| 06:57.58 | orlock | MoTeC looking for a sysadmin, heh |
| 06:58.05 | JT | orlock: heh |
| 06:59.04 | *** join/#asterisk daysmen3 (n=primus@host81-154-136-36.range81-154.btcentralplus.com) |
| 07:00.38 | *** part/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 07:00.41 | james_ | orlock: that sounds interesting |
| 07:00.52 | orlock | james_: what, sysadmin for motec? |
| 07:01.01 | orlock | its just plain old boring sysadmin stuff |
| 07:01.10 | orlock | just you get to work for MoTeC :) |
| 07:01.21 | james_ | oh nah, the nextep network |
| 07:01.29 | james_ | sounds very developmental |
| 07:01.31 | orlock | the nextep stuff would be more interesting, and have even more smart geeks around |
| 07:01.35 | orlock | james_: it is! |
| 07:01.36 | james_ | like a public network for testing |
| 07:01.36 | james_ | heh |
| 07:01.45 | james_ | that's something i'd like to support |
| 07:01.48 | james_ | but |
| 07:01.52 | james_ | i bet they dont have dsl |
| 07:01.55 | james_ | at my exchange |
| 07:01.56 | james_ | dsl2 |
| 07:01.57 | orlock | james_: had nextep at home for years now |
| 07:01.57 | james_ | sorry |
| 07:02.18 | orlock | they do dsl over the standard telstra network as well as their own higher speed stuff using their dslams |
| 07:02.26 | orlock | its not adsl2, but its 8m/640k |
| 07:02.33 | *** join/#asterisk _omer (i=_omer@202.166.161.23) |
| 07:02.36 | _omer | hi |
| 07:02.39 | orlock | i think they have just started adsl2 |
| 07:02.45 | _omer | how to remove MPG123 from my system ? |
| 07:03.07 | james_ | that's a pretty open ended question |
| 07:03.29 | james_ | orlock: yeah, can't beat my 19Mb/1Mb dsl2 though :P |
| 07:03.34 | _omer | <PROTECTED> |
| 07:04.07 | orlock | james_: actually... i reckon it could :) |
| 07:04.22 | orlock | james_: whats the max speed you can actually obtain on it? :) |
| 07:04.46 | james_ | i sync at 19000bps |
| 07:04.55 | orlock | james_: did i mention they also do voip, and _support_ it? |
| 07:04.59 | orlock | all sip based |
| 07:05.00 | james_ | fastest i've downloaded is 1.8Mb/s |
| 07:05.16 | james_ | that's not a concern to me :P |
| 07:05.27 | james_ | i have my own voip providers |
| 07:05.35 | james_ | but you know... a business with a clue is always good |
| 07:07.34 | JT | i think you sync at a little more than 19000bps |
| 07:07.37 | JT | pretty slow otherwise |
| 07:08.05 | orlock | james_: 6784/736 |
| 07:08.10 | orlock | from dmesg |
| 07:08.14 | orlock | i *heart* pci adsl cards |
| 07:08.18 | JT | hah |
| 07:08.31 | JT | haven't heard good things about those |
| 07:08.34 | JT | just that they're finicky |
| 07:08.45 | JT | and are pretty fussy about kernel |
| 07:08.46 | orlock | JT: these are the Traverse ones |
| 07:08.50 | orlock | not these ones |
| 07:08.54 | orlock | sourceforge project |
| 07:08.57 | orlock | gpl drivers |
| 07:09.00 | JT | 2.6 works fine? |
| 07:09.04 | orlock | i made my own .src.rpm to rebuild it for any kernel |
| 07:09.06 | orlock | yeah |
| 07:09.24 | orlock | 2.6 works fine, been testing it at home before deployment |
| 07:09.43 | orlock | its actually the exact same card as the Sangoma S518 (i think) adsl cards |
| 07:09.44 | JT | think i'd rather a hardware box though |
| 07:10.43 | orlock | yeah, its a full hardware card though at least |
| 07:11.02 | orlock | no cpu overhead, latest drivers also use the in-kernel atm layer |
| 07:11.17 | orlock | so you can use ppp with the atm plugin, etc |
| 07:11.42 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 07:13.42 | *** join/#asterisk marl (n=matt@82-40-232-252.cable.ubr01.dumb.blueyonder.co.uk) |
| 07:14.02 | james_ | haha |
| 07:14.07 | james_ | it's <3 not *heart* |
| 07:14.30 | james_ | i just had to sacrifice my cisco 837 router for a netcomm nb5 bridged to a linksys wrt |
| 07:14.34 | james_ | thanks to dsl2 |
| 07:14.45 | james_ | need an 877 or whatever model it is that supports dsl2 |
| 07:15.49 | ontae | May anyone help me with an RTP-stream problem when calling out? |
| 07:16.25 | *** join/#asterisk num000 (n=numerobi@e177179008.adsl.alicedsl.de) |
| 07:16.36 | JT | yes 877 |
| 07:16.42 | JT | only costs around $700 |
| 07:17.39 | james_ | yeah, slightly higher than 19000bps |
| 07:17.46 | james_ | i haven't had that for about 10 years |
| 07:18.02 | JT | also 1.8Mb/s isn't that great either :] |
| 07:18.06 | james_ | 19Mb |
| 07:18.08 | james_ | MB |
| 07:18.09 | james_ | shhhh |
| 07:18.25 | james_ | stoned*cough* |
| 07:18.41 | JT | unit dyslexicaritus |
| 07:18.47 | JT | heh |
| 07:19.58 | orlock | james_: now? lucky bastard! |
| 07:21.18 | _omer | Aug 17 12:46:10 WARNING[15236]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_conference.so: cannot restore segment prot after reloc: Permission denied |
| 07:21.21 | _omer | any help ? |
| 07:21.28 | james_ | orlock: i'm unemployed :P |
| 07:21.48 | james_ | no better time to waste the time i'm not at job interviews |
| 07:23.45 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 07:26.12 | orlock | james_: ahh, heh |
| 07:26.21 | orlock | james_: go for the jobs at nextep! :) |
| 07:28.43 | *** join/#asterisk X-Rob (n=rob@dsl-220-235-226-54.vic.westnet.com.au) |
| 07:33.42 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 07:33.43 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 07:34.09 | james_ | orlock: don't really sound like what i'm looking for.... i want a job developing around the sip protocol |
| 07:34.16 | james_ | higher level than firmware :P |
| 07:34.37 | james_ | orlock: you're in melbourne? |
| 07:34.44 | orlock | yeah |
| 07:34.48 | james_ | which suburb? |
| 07:34.52 | orlock | richmond |
| 07:34.55 | orlock | work in southbank |
| 07:35.02 | james_ | ahh nice |
| 07:35.04 | james_ | close to the city |
| 07:35.09 | james_ | i'm in caulfield |
| 07:35.15 | orlock | james_: well, i would say it would be worth it.. they are a quality voip provider as well, all sip based |
| 07:35.15 | james_ | well, malvern east |
| 07:35.26 | james_ | just near the tattersalls building on dandenong rd |
| 07:35.37 | orlock | ahh, think i know the one |
| 07:35.46 | james_ | everyone does :P |
| 07:36.27 | james_ | where abouts are their jobs advertised? |
| 07:36.42 | orlock | seek.com.au |
| 07:36.52 | james_ | i'll search, ta |
| 07:37.21 | james_ | do you work for them? or just really like them? haha |
| 07:37.37 | orlock | network admin for a major reseller |
| 07:37.56 | orlock | used a fair few different ISP's, they suck the least |
| 07:38.00 | vlt | Hello. Is there a way to SET() the outgoing CALLERID on SIP calls? |
| 07:38.08 | james_ | yes |
| 07:38.27 | james_ | probably the easiest thing in asterisk to search for and find the answer for |
| 07:38.49 | vlt | james_: Then I should search elsewhere? ;-) |
| 07:39.43 | james_ | www.google.com |
| 07:39.47 | james_ | "asterisk set caller id" |
| 07:39.57 | james_ | click "i'm feeling lucky" |
| 07:40.20 | ontae | vlt: Remember me? Yesterday, the problem with rtp-stream on outgoing call: Set externip, localnet and nat=no. When i set nat=yes, i get a "No one is available to answer at this time" |
| 07:45.20 | benjk | "No one is available to answer at this time" is a vanilla message of Asterisk and it means very little |
| 07:45.37 | benjk | many different causes are mapped to this message |
| 07:45.43 | benjk | many times it is even wrong |
| 07:45.53 | X-Rob | sip show peers! |
| 07:47.53 | JT | should a dual PIII 500MHz be fine with a T1 + channel bank? |
| 07:48.02 | JT | utilisation probably won't be near 24 ports |
| 07:48.09 | JT | but i am wondering if it would handle it |
| 07:50.39 | vlt | ontae: When does this error appear? When someone tries to call in or when you try to call out or ...? |
| 07:51.47 | *** join/#asterisk parag_ast (n=root@dxb-b111124.alshamil.net.ae) |
| 07:52.02 | parag_ast | Can anybody let me know why i gets this SIP response 481 "Call Leg/Transaction Does Not Exist" |
| 07:53.58 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 07:56.56 | benjk | JT, we have an IBM NetVista slim desktop system with a T1 card which we use for demos and tests, its got a PIII/500MHz, 384MB RAM, works with 23 PRI channels fully loaded just fine, but all G711 codecs |
| 07:57.20 | *** join/#asterisk af_ (n=af@ip-173-144.sn1.eutelia.it) |
| 07:58.59 | JT | i'm curious about what would happen if G711 was replaced with a cpu intensive codec |
| 07:59.02 | vlt | james_: Mmh ... I found SetCallerID() and it is executed when dialling out. I removed "fromuser=" from sip.conf but still the wrong the CallerID is shown on the other side ... What else could I check? |
| 08:00.11 | james_ | vlt: depends on the destination channel type |
| 08:00.16 | james_ | is it all completely sip? |
| 08:00.20 | ontae | vlt: When i try to call out and nat=yes; When i try to call out with nat=no i hear no voice at the called phone; Incoming calls function with no problem (voice on both end) |
| 08:00.26 | james_ | tried setting the callerid= in sip.conf? |
| 08:02.23 | vlt | james_: Channel type is completely SIP. My call goes to provider's asterisk and then back to one of my servers ... Maybe the callerID is mangled on the provider side. |
| 08:03.16 | benjk | JT, we've done some demos with mixed codec use and that was ok, but never trying to saturate the T1 with all low bandwidth codecs |
| 08:03.25 | JT | well a sensible provider isn't going to allow an illegal callerid setting |
| 08:03.33 | JT | benjk: hrm ok |
| 08:03.49 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 08:04.00 | benjk | more like 18-20 chans G711 and a handful GSM/ILBC/Speex |
| 08:04.10 | JT | i'm thinking even if that is sufficient, it will probably still be a squeeze to put my CCTV capture cards on the same box |
| 08:04.24 | benjk | so not sure at which point the thing would start having trouble |
| 08:04.48 | dorel__ | is there some special way to enable call recording on freepbx (amp)? |
| 08:04.51 | benjk | you probbly dont want to do that :) |
| 08:04.54 | JT | pulling upwards of a combined framerate of 100fps at PAL resolution might cause a bit of bus demand |
| 08:04.59 | JT | heh yeah |
| 08:05.00 | JT | :/ |
| 08:05.24 | JT | man |
| 08:05.34 | JT | i do not want too many servers at home |
| 08:05.38 | JT | it will be too hot in summer |
| 08:05.41 | JT | and noisy |
| 08:06.35 | benjk | also, JT, I have just noticed that the number of concurrent calls my systems can handle significantly increased with my changes to pbx.c using hash compares instead of strcmps and strcasecmps |
| 08:06.46 | benjk | so you may want to look at that |
| 08:06.50 | JT | hmm |
| 08:07.09 | JT | speak to a linux cctv guru i know to see what he says about that cpu requirement |
| 08:07.35 | JT | i suspect it will preclude using the same hardware as asterisk, even if asterisk is only use a half dozen lines |
| 08:07.35 | benjk | if you have 100 apps loaded in your *, then each time your dialplan calls an app, it will call strcasecmp 100 times |
| 08:07.43 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 08:08.01 | JT | especially if i use numberplate recognition |
| 08:08.02 | JT | hmm |
| 08:08.17 | vlt | ontae: I experienced different behaviors with different SIP providers. dus.net here works even without forwarding RTP ports, sipgate.de needs some special treatment ... |
| 08:08.28 | JT | i wonder if a PIII700 single cpu would be sufficient for asterisk |
| 08:10.30 | benjk | so I changed pbx.c to not use strcmps |
| 08:10.32 | bionoid | jt I'm running a P3/500/300mb ram without problems. Obviously very small pbx :) |
| 08:11.08 | benjk | and since this is an incredible waste of resources, it has dramatically increased the capacity |
| 08:11.12 | JT | bionoid: heh, yeah i'm wondering what would happen if it went up to 24 channels + low bandwidth codec |
| 08:11.25 | JT | benjk: submitted a bug fix? |
| 08:11.44 | bionoid | JT: Then you'll probably benefit from an upgrade.. ;p |
| 08:12.31 | JT | couldn't find any SBCs > 700MHz on ebay :( |
| 08:12.36 | benjk | Digium won't accept it, I add it to repo at opbx |
| 08:13.03 | JT | i have an SBC backplane for 2 SBCs that'd be great to save some space |
| 08:13.09 | JT | have a dual PIII500 sbc already |
| 08:13.28 | JT | benjk: did you disclaim it, or was it canned for other reasons? |
| 08:14.07 | benjk | my stuff is BSD licensed |
| 08:14.08 | benjk | http://trac.openpbx.org/cgi-bin/trac.cgi/browser/openpbx/branches/benjk |
| 08:14.22 | benjk | you can easily apply this to your pbx.c |
| 08:14.37 | JT | i see |
| 08:15.42 | benjk | its a lot of small changes |
| 08:15.53 | benjk | nothing big |
| 08:16.44 | benjk | like if (!strcmp(foobar, "FOO") { ... becomes if (hash == AST_KEYWORD_FOO) { ... |
| 08:16.56 | benjk | straightforward |
| 08:17.24 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 08:22.36 | *** join/#asterisk xnon (i=xnon@200.82.222.64) |
| 08:22.57 | xnon | anybody can tell me why tell me this Aug 17 03:15:06 NOTICE[18778]: rtp.c:331 process_rfc3389: Comfort noise supportincomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 212.36.71.106 |
| 08:22.57 | xnon | <PROTECTED> |
| 08:26.53 | JT | seems that the client is requesting comfort noise |
| 08:32.14 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 08:35.58 | *** join/#asterisk zedkatuf (n=zedkatuf@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk) |
| 08:40.49 | *** part/#asterisk parag_ast (n=root@dxb-b111124.alshamil.net.ae) |
| 08:54.41 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 08:54.59 | *** part/#asterisk MACscr (n=MACScr@adsl-75-23-104-12.dsl.peoril.sbcglobal.net) |
| 08:56.19 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:59.35 | *** join/#asterisk SHad|Work (n=kvirc@84.255.228.2) |
| 08:59.44 | SHad|Work | hi |
| 09:00.40 | SHad|Work | I know this isn't the topic for this channel but does anyone know of a free/opensource softphone that works on linux and windows and has the hold/transfer features as well as auto answer? |
| 09:01.36 | *** join/#asterisk bartpbx (n=bartpbx@p54B0486C.dip0.t-ipconnect.de) |
| 09:03.48 | bartpbx | hello |
| 09:04.01 | bartpbx | our asterisk crashed just again |
| 09:04.16 | E-bola | Where can i find info about having a dialplan thats depending ont he time of day? |
| 09:04.24 | intralanman | bartpbx: that sucks |
| 09:04.38 | bartpbx | anyone knows about a current problem in chan_iax2? |
| 09:04.41 | intralanman | E-bola: show application gotoiftime i think |
| 09:05.00 | E-bola | intralanman:_ thanks, trying |
| 09:05.12 | E-bola | i just wanna setup a recorded msg to be played when the office is closed |
| 09:05.13 | bartpbx | Tzangler said yersterday he seen this also an a frew servers |
| 09:05.16 | E-bola | thats the way to do it right? |
| 09:05.35 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:05.39 | backblue | morning * |
| 09:05.43 | intralanman | that's the way i do it |
| 09:07.23 | *** part/#asterisk intralanman (n=intralan@pool-72-82-74-171.nrflva.east.verizon.net) |
| 09:08.51 | ontae | vlt: sipgate.at is my "provider" |
| 09:09.28 | vlt | ontae: Have you set "canreinvite=no"? |
| 09:14.45 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:15.13 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 09:15.19 | *** join/#asterisk dyerf (i=dyer@cube.sut.ru) |
| 09:20.34 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:23.31 | Dico_ | bartpbx, how does it crash ? |
| 09:23.46 | bartpbx | with a core dump |
| 09:23.58 | Dico_ | ye but what was it doing ? |
| 09:23.59 | bartpbx | looks like pastbin is down |
| 09:24.07 | bartpbx | still searching |
| 09:24.19 | Dico_ | i see |
| 09:24.22 | bartpbx | i now activated more detailed logging |
| 09:25.42 | *** join/#asterisk [Airwolf] (n=airwolf@195.80.226.162) |
| 09:26.05 | bartpbx | looks like some variable not beeing availible |
| 09:26.19 | xnon | it is posible send via voicemail in at the same voice box to a 3 email address? |
| 09:27.12 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 09:27.19 | E-bola | Aug 17 11:26:01 DEBUG[17550]: app_dial.c:1619 dial_exec_full: Exiting with DIALSTATUS=CANCEL. |
| 09:27.19 | E-bola | <PROTECTED> |
| 09:27.25 | E-bola | is this bad? to exit none zero? |
| 09:27.41 | hads|home | E-bola: Normal |
| 09:27.58 | E-bola | but should i try to get it removed |
| 09:28.00 | E-bola | or doesnt it matter? |
| 09:28.08 | hads|home | It's normal. |
| 09:30.27 | xnon | friends i have 3 SIP Providers (adamvozip.es, fwd.net and peoplecall.com) can i set rulez in my asterisk for the incoming calls for any this sip providers enter to a operator ????????? |
| 09:30.39 | xnon | anybody have a example for this? |
| 09:31.51 | benjk | E-bola: exit with zero means the dialplan will continue, non-zero means it will stop |
| 09:32.03 | xnon | friends i have 3 SIP Providers (adamvozip.es, fwd.net and peoplecall.com) can i set rulez in my asterisk for the incoming calls for any this sip providers enter to a operator ????????? |
| 09:32.11 | xnon | anybody have a example for this? |
| 09:32.17 | benjk | so, if the call is hung up, then it has to exit non-zero or it will not stop |
| 09:32.20 | E-bola | benjk: ahh then it makes sence |
| 09:33.04 | xnon | my incoming calls for the any sip provider cant enter in my asterisk |
| 09:33.20 | benjk | this his how the Dial application lets the pbx know that the call was hung up so the pbx can stop |
| 09:33.52 | benjk | the pbx engine doens;t look at DIALSTATUS, that's only for us humans for easy reading |
| 09:38.46 | *** join/#asterisk razu_w (n=rasmus@tln-kontor.norby.ee) |
| 09:38.53 | bartpbx | here http://pastebin.com/770435 you can find a bt of the core dump |
| 09:40.55 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 09:42.23 | *** join/#asterisk nailbags|laptop (i=someone@c220-237-123-137.randw1.nsw.optusnet.com.au) |
| 09:43.26 | *** join/#asterisk |oranjia| (n=root@dsl-146-0-119.telkomadsl.co.za) |
| 09:47.09 | *** join/#asterisk BjornRobertsson (n=bjornr@213-213-148-71.xdsl.is) |
| 09:52.49 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:55.48 | bartpbx | I have a problem with one of our clients he is telling me that sometimes calls abort after a certain time |
| 09:56.15 | bartpbx | the onlything i could identify is a meesage saying. Didn't get a frame from channel: IAX2/100-24 |
| 09:56.38 | bartpbx | than the call is stoped and logged as "Answered" |
| 09:56.56 | bartpbx | the agend is running idefix |
| 09:58.03 | E-bola | Hey guys, have anybody seen a setup that lets user "turn on night mode" in asterisk? |
| 09:58.12 | E-bola | meaning they can somehow trigger night mode on their phone or from a homepage |
| 09:58.23 | E-bola | meaning the office is closed and all calls will be directed to an answering machine |
| 09:58.42 | luchshiy | ps -ax |grep asterisk |
| 09:58.46 | E-bola | i can use the gotoiftime but then the time the opffice is closed is static, woudl be much more userfriendly if the employees coudl just turn it on |
| 09:58.48 | luchshiy | îé |
| 09:58.58 | hads|home | E-bola: Use the asterisk database (astdb) and a gotoif. |
| 09:59.14 | E-bola | hads|home: and use a web front to set the mode in the db? |
| 09:59.18 | bartpbx | oder du mich |
| 09:59.32 | bartpbx | sorry.. wrong window |
| 09:59.44 | hads|home | E-bola: If you really want. YZou could just set up and extension for them to dial... |
| 09:59.59 | E-bola | ya that woudl be best |
| 10:00.09 | E-bola | do u have any examples of such a configuration? |
| 10:00.53 | hads|home | 'show function DB' from the CLI is your friend |
| 10:00.55 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 10:01.39 | E-bola | guess i need to readup on astdb |
| 10:01.44 | E-bola | i dont know if i need ot set it up somehow or what |
| 10:02.16 | hads|home | E-bola: http://www.asteriskguru.com/tutorials/dbget_function.html |
| 10:02.22 | E-bola | thanks |
| 10:02.42 | xnon | my incoming calls for the any sip provider cant enter in my asterisk |
| 10:02.47 | xnon | friends i have 3 SIP Providers (adamvozip.es, fwd.net and peoplecall.com) can i set rulez in my asterisk for the incoming calls for any this sip providers enter to a operator ????????? |
| 10:03.12 | xnon | anybody have a example for this? |
| 10:05.45 | *** join/#asterisk Kasimeng (n=chong_me@125.215.196.231) |
| 10:05.59 | BjornRobertsson | I just upgraded fromn 1.2.7 to 1.2.10 and after some time audio fails, outgoing sound is missing but incoming sound is ok. This is using bristuff-0.3.0s |
| 10:11.57 | vlt | Question: I have one "exten => _0." and one "exten => _0341.", but when I dial 0341555 always the "0." exten wins. Why? |
| 10:12.49 | xnon | friends i have 3 SIP Providers (adamvozip.es, fwd.net and peoplecall.com) can i set rulez in my asterisk for the incoming calls for any this sip providers enter to a operator ????????? |
| 10:15.01 | *** join/#asterisk grEvenX (n=even@pc100-15.ktv.no) |
| 10:15.15 | razu_w | hi ... anyone knows if there is any possibility to turn off OnHook Flash from template file. Phone is Granstream BT seies |
| 10:15.19 | *** join/#asterisk bofh42 (n=bofh42@p5482896A.dip0.t-ipconnect.de) |
| 10:16.58 | *** join/#asterisk florinm (n=florin_m@host-84-9-255-83.bulldogdsl.com) |
| 10:17.02 | florinm | hi there |
| 10:17.09 | *** join/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 10:17.16 | florinm | got a problem with the latest asterisk 1.2.10 |
| 10:17.44 | grEvenX | what.. |
| 10:17.48 | florinm | i have setup a q system with permanent members (zap/5, zap/6,zap/7) |
| 10:17.50 | RoyK | morning |
| 10:17.58 | florinm | and it doesn;t ring the members |
| 10:18.15 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 10:18.20 | florinm | is calling and hungup in same time |
| 10:18.31 | RoyK | anyone here that can help me out? If dialing sip->zap with the t flag, transfer works well, but if dialing zap->sip with the T flag, SIP client is not able to trigger transfer |
| 10:19.07 | florinm | before on 1.2.7 was working ok same setup :( |
| 10:19.25 | florinm | <PROTECTED> |
| 10:19.26 | florinm | <PROTECTED> |
| 10:19.26 | florinm | <PROTECTED> |
| 10:19.26 | florinm | <PROTECTED> |
| 10:19.26 | florinm | <PROTECTED> |
| 10:19.26 | florinm | <PROTECTED> |
| 10:19.27 | florinm | <PROTECTED> |
| 10:19.39 | florinm | any ideas? |
| 10:19.58 | florinm | the full logs shows: |
| 10:19.59 | florinm | Aug 17 11:10:35 VERBOSE[15397] logger.c: -- Called Zap/7 |
| 10:20.00 | florinm | Aug 17 11:10:35 DEBUG[15397] chan_zap.c: Hangup: channel: 5 index = 0, normal = 20, callwait = -1, thirdcall = -1 |
| 10:20.08 | florinm | so no reason for the hungup :( |
| 10:23.53 | ruskie | hmm can anyone recommend an easy to use web interface to asterisk? |
| 10:25.15 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_N@kantoor.jronline.nl) |
| 10:25.28 | QuAtRo[NL] | I have a little problem with a call-me script |
| 10:27.09 | E-bola | hehe dumb question |
| 10:27.18 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 10:27.24 | E-bola | anybody know if any of the default soundfiles has something like "enabled" "activated" or similar? |
| 10:27.31 | QuAtRo[NL] | When I start Asterisk and run the script it works fine. But when i run it for the second time, it doesn't work anymore |
| 10:27.41 | E-bola | i just need a voice that confirms |
| 10:27.46 | QuAtRo[NL] | Asterisk just says: Manager 'admin' logged on from 127.0.0.1 |
| 10:27.57 | QuAtRo[NL] | And after that the logoff |
| 10:28.08 | *** join/#asterisk [Airwolf] (n=airwolf@195.80.226.242) |
| 10:28.11 | QuAtRo[NL] | And the script does not what it suppose to do, call |
| 10:28.20 | E-bola | hmm |
| 10:28.23 | E-bola | guess i can use added and removed |
| 10:28.58 | QuAtRo[NL] | Does someone know what the problem can be? |
| 10:30.04 | QuAtRo[NL] | You can find the script here: http://www.asteriskextras.com/index.php?option=com_content&task=view&id=12&Itemid=2 |
| 10:32.42 | florinm | any idea for my problem? |
| 10:35.01 | *** join/#asterisk evisu (i=hIRC@bzq-88-154-45-231.red.bezeqint.net) |
| 10:36.18 | suma | is there is any PC boxed version of asterisk can take single pci card ? |
| 10:36.24 | suma | with small form factor ? |
| 10:40.33 | razu_w | QuAtRo[NL] : Does manager Action get all parameters it needs ? |
| 10:47.40 | ghenry | how can you tell if zap calls are in progress on cli? |
| 10:49.57 | *** join/#asterisk LoneShadow (n=duh@59.92.169.215) |
| 10:50.24 | evisu | Anyone know what dtmf settings are needed to get dtmf working with voipjet? |
| 10:50.29 | xnon | hey friends the files of voices in the other languaje must be in /var/lib/asterisk/sounds/ isnt it? |
| 10:51.00 | xnon | did u try rfc 2833? |
| 10:51.45 | evisu | yep, |
| 10:52.25 | *** join/#asterisk bofh42 (n=bofh42@p5482A52D.dip0.t-ipconnect.de) |
| 11:04.19 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:07.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:11.38 | benjk | suma: Soekris |
| 11:12.02 | benjk | also pcengines.ch |
| 11:14.14 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 11:14.34 | QuAtRo[NL] | razu_w: Yes it does. Why would it otherwise work the first time and not the second time? |
| 11:17.15 | razu_w | QuAtRo[NL] : I've had originate problems only when the channel is up or the script doesn't get parameters it needs |
| 11:17.35 | QuAtRo[NL] | Oke |
| 11:17.58 | QuAtRo[NL] | But in that case, Asterisk had to tell me there were problems |
| 11:18.43 | razu_w | but try to get the result of originate into some log file or smthing |
| 11:19.25 | QuAtRo[NL] | All the variables are being shown at the webppage I visit |
| 11:19.47 | QuAtRo[NL] | And the result is always the same |
| 11:19.53 | QuAtRo[NL] | Whether it works or not |
| 11:21.57 | ontae | vlt: yes, set "canreinvite=no" --> no voice/sound at the end i call |
| 11:22.42 | razu_w | QuAtRo[NL] : nono ... manager result I mean ... wait a sec ... I'll show you a modified script |
| 11:23.15 | ontae | vlt: What abaout the thing with the wrong ip, my asterisk replies? Have you seen that at "http://lists.digium.com/pipermail/asterisk-users/2006-August/162999.html" |
| 11:23.39 | QuAtRo[NL] | That would be nice |
| 11:27.27 | vlt | ontae: /join #asterisk.de |
| 11:28.23 | *** part/#asterisk bartpbx (n=bartpbx@p54B0486C.dip0.t-ipconnect.de) |
| 11:28.33 | vlt | Question: I have one "exten => _0." and one "exten => _0341.", but when I dial 0341555 always the "0." exten wins. Why? |
| 11:28.47 | razu_w | QuAtRo[NL] : this should write manager responses into log.txt file in the same directory where script is ... (let me know if it has some error) http://razu.pri.ee/callme.php.blah |
| 11:29.10 | benjk | because its shorter and both ends with . |
| 11:29.30 | bionoid | vlt: because . means _any number_ of any digit, so it'll match |
| 11:30.01 | QuAtRo[NL] | razu_w: Nothings there |
| 11:30.30 | benjk | what you can is this |
| 11:30.38 | vlt | benjk, bionoid: How can I seperate _0341. from _0[^341]. calls? |
| 11:30.39 | benjk | [match0] |
| 11:31.15 | razu_w | QuAtRo[NL] : try this one: http://razu.pri.ee/callme.blah |
| 11:31.22 | QuAtRo[NL] | ;) |
| 11:31.28 | bionoid | vlt: use X instead of . (X denotes a single digit), so exten => _0341X |
| 11:31.34 | benjk | exten => _0.,1,Goto(***${EXTEN},1) |
| 11:31.53 | bionoid | Or that. |
| 11:32.08 | benjk | exten => _***0341.,1,NoOp(this matches 0341...) |
| 11:32.25 | benjk | exten => i,1,NoOp(this matches anything else) |
| 11:32.52 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 11:34.08 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-a9ef85c18d23297a) |
| 11:34.49 | QuAtRo[NL] | razu_w: it works :D |
| 11:35.00 | razu_w | QuAtRo[NL] : great :) |
| 11:35.27 | razu_w | QuAtRo[NL] : so now you can debug it better if you need to :) |
| 11:41.28 | ontae | vlt: o.k. try it there, thanks |
| 11:47.04 | *** join/#asterisk Makenshi (n=chaz@2001:630:1c0:0:212:219:188:81) |
| 11:50.23 | *** join/#asterisk rdsousa (n=kvirc@213.205.87.88) |
| 11:51.16 | rdsousa | anyone knows a ip dect solution that works in asterisk |
| 11:51.20 | rdsousa | ? |
| 11:52.48 | benjk | apparently nobody does |
| 11:53.32 | ionix | what is dect? |
| 11:53.55 | rdsousa | dect is a wireless communication method |
| 11:53.57 | benjk | European digital cordless telephone system |
| 11:54.01 | rdsousa | for telephony |
| 11:58.57 | *** join/#asterisk eliXier (i=GTI16V@gti.twice-irc.de) |
| 11:59.25 | chexum | I saw a .tw sip/dect gateway announced, but don't know if it's available... |
| 12:00.18 | benjk | .tw? |
| 12:01.14 | rdsousa | i want a solution not just for 4 or 5 phones |
| 12:01.32 | rdsousa | but for 20-40 phones |
| 12:01.47 | rdsousa | with various dect base stations |
| 12:02.05 | rdsousa | and dect phones are a lot cheaper that wi-fi phones |
| 12:02.28 | rdsousa | i can get a good siemens phone for about 50 $ |
| 12:03.15 | benjk | they are a lot better |
| 12:03.19 | *** join/#asterisk merryberry (n=chatzill@193.189.66.86) |
| 12:03.20 | benjk | never mind the price |
| 12:04.03 | chexum | (Taiwanese, but what isn't :) |
| 12:04.13 | benjk | ah |
| 12:04.21 | benjk | export model I guess |
| 12:04.40 | benjk | Taiwan uses the Japanese PHS system as far as I recall |
| 12:05.10 | rdsousa | what model? |
| 12:05.20 | benjk | DECT |
| 12:05.20 | chexum | rdsousa: well, I'd think you can have multiple ones for the increasing number of phones.. |
| 12:05.49 | benjk | WiFi isnt exactly suitable for telephony |
| 12:05.56 | benjk | WiFi is half duplex |
| 12:06.16 | benjk | only one side can send at a time |
| 12:06.19 | rdsousa | yes i had a very bad experience with wifi |
| 12:06.52 | benjk | you can verify this by copying a very large file from your notebook computer over WiFi to the LAN |
| 12:06.54 | merryberry | Hi, can someone point me in the right direction for docs on asterisks 3G video gateway features, thanks. |
| 12:07.00 | rdsousa | and then i switched for a dect solution and it worked perfectly |
| 12:07.16 | benjk | the progress indicator will go in steps, stop for a while, make another large jump, stop again etc etc |
| 12:07.35 | benjk | this is because it has to wait its turn |
| 12:07.35 | rdsousa | that's why i want to use dect |
| 12:07.57 | rdsousa | dect works fine and the phones are a lot cheaper |
| 12:07.57 | benjk | for telephony you need very fast turn around times |
| 12:08.19 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:08.21 | benjk | small packets but lots of them and process them asap |
| 12:08.52 | rdsousa | now what's missing is an ip dect base station that works with asterisk |
| 12:08.56 | benjk | versus few large packets queuing up and getting processed in batches |
| 12:09.11 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 12:11.10 | vlt | It works. _0341X. and _0X. are different extensions. I just don't know why there's a difference between "Zero and anything after" and "Zero, a digit and anything after" ... |
| 12:11.37 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:11.38 | vlt | ... when the dialled number is always 0341555 |
| 12:12.34 | rdsousa | i think it might be necessary for asterisk work with dect some kind of software/driver that controls the dect base stations |
| 12:12.37 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:12.44 | benjk | because the dot makes it an immediate match and without the X its treated like an exact match |
| 12:13.23 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:13.44 | benjk | not really, if you get a DECT base that speaks SIP, that'll be just fine |
| 12:21.57 | rdsousa | yes |
| 12:22.26 | rdsousa | but i also want that the dect base stations be able to do roaming |
| 12:23.29 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:27.36 | *** join/#asterisk koolie (n=admin@85-189-109-186.signaturenetworks.managedbroadband.co.uk) |
| 12:27.40 | koolie | hi |
| 12:28.24 | koolie | what motherboard would be the most effective to use with asterix when in a corporate environment ? |
| 12:31.50 | [TK]D-Fender | koolie: depends on your needs. Describe your projected setup |
| 12:32.51 | jbalcomb | I have a call queue that has memebers and accepts call but doesn't show up in my 'show queues'. Why and how do I get it to show up? |
| 12:34.09 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 12:35.06 | *** part/#asterisk Makenshi (n=chaz@2001:630:1c0:0:212:219:188:81) |
| 12:36.09 | chexum | rdsousa: from the looks of it, roaming may need special "multi-cell" capable handsets too? otherwise it would interest me too :) btw, with a DECT SIP gw, making the calls roam should be much easier, since they could easily pass on call to another IP |
| 12:41.28 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 12:42.18 | koolie | well, its a small business, with internal extensions, external analog line, with VOIP too |
| 12:42.26 | koolie | currently we are having packet loss |
| 12:43.44 | koolie | also when ringing takes a few seconds to disconnect the call... (termination of call) |
| 12:44.05 | koolie | also, we've had some ghost phone calls, probably just residue packets going around the boards ? |
| 12:45.33 | mut | no, its actaul ghosts calling |
| 12:45.40 | mut | -- |
| 12:47.14 | ionix | Anyone has a good SIP/IAX2 provider? |
| 12:47.45 | [TK]D-Fender | koolie: Packet loss is a networking issue, taking time for disconnect to be detected is either a telco side setting to be changed if available, or just a fact of life. |
| 12:48.29 | [TK]D-Fender | ionix: all ITSP's suck, just some less than others a varying points in time. |
| 12:49.01 | ionix | like I wouldn't mind paying more than the 1.39 cents a call |
| 12:49.07 | ruskie | hmm can anyone recommend either a good tutorial or a decent web interface to configuring asterisk(I'd like to use it as a SIP gateway between internal clients and external providers) and IAX with the same thing){having a symetrical firewall} |
| 12:49.29 | ionix | but my calls rely on DMTF so I need good quality or ITSP that are directly connected to TDM and use outband DTMF |
| 12:49.45 | ionix | i.e: I don't care about paying 10cents a minute if my calls are successful. |
| 12:50.17 | [TK]D-Fender | ionix: Why not just get a PRI? |
| 12:50.39 | ionix | I am in Japan, server is in Canada |
| 12:50.51 | ionix | and PRI is too expensive for now, I plan to launch a service in September |
| 12:50.56 | [TK]D-Fender | ionix: So you need JP DID's or Canadian? |
| 12:51.15 | ionix | I don't really need a DID just to terminate calls internationaly |
| 12:51.18 | *** join/#asterisk silentfury (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 12:51.25 | ionix | mostely US and Can |
| 12:51.28 | silentfury | morning |
| 12:51.32 | ionix | morning signuts |
| 12:51.34 | ionix | silentfury |
| 12:51.49 | silentfury | i've got a quick q relating to voip.. have a home user that we use an ip phone for work |
| 12:52.05 | silentfury | is there any decent QoS router that anyone would recommend? |
| 12:52.12 | [TK]D-Fender | ionix: Ok, well they may not be the best rates out there, but they offer good quality service : www.unlimitel.ca |
| 12:52.51 | vlt | Hello. How can I suppress my CallerID in ougoing SIP calls? |
| 12:53.05 | ionix | [TK]D-Fender: would it be possible to make a couple of free test calls? |
| 12:53.18 | [TK]D-Fender | ionix: one of my customers was using them amongst 3 other providers hoping to get LCR up and running using Broadvoice and other in atddition to Unlimitel. They ditched everyone but Unlimitel by the time they were through. |
| 12:53.48 | [TK]D-Fender | ionix: Call them up. I've just heard of several happy cutomers and have called there myself once or twice. |
| 12:54.56 | ionix | ok thx |
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| 13:06.19 | mut | check out the woot today |
| 13:06.20 | mut | http://www.woot.com/ |
| 13:06.23 | mut | usb voip phon |
| 13:07.34 | ionix | that's lame |
| 13:08.15 | ionix | it's nothing more than a mic and speaker for skype etc |
| 13:08.33 | *** join/#asterisk BudaH (n=budah@201.21.236.5) |
| 13:08.36 | mut | yea |
| 13:08.50 | bionoid | get this for your cellphone instead: http://www.thinkgeek.com/gadgets/electronic/7830/ |
| 13:09.24 | tzanger | hmm ... iax2 rejecting calls with "no authority found" usually means that the user/pass does not match, or the context does not have a match for hte dialed number... |
| 13:09.27 | tzanger | but both of these seem fine |
| 13:09.39 | MrChimpy | any recommendations for a SIP or IAX softphone which is free source? I need to customise the interface... |
| 13:10.17 | bionoid | MrChimpy: http://www.voip-info.org/wiki-Open+Source+VOIP+Software has some good info |
| 13:10.26 | MrChimpy | thanks bio |
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| 13:21.03 | kappelmi | hello all, anyone have used Nokia E60 with asterisk? |
| 13:23.40 | UlbabraB | kaldemar: i'm using an E70, works perfectly with g711 codec, some scratches with g729 |
| 13:23.53 | mut | man |
| 13:24.05 | mut | those qos sipura SPA-2100's suck |
| 13:24.11 | mut | the qos dun do crap |
| 13:24.19 | mut | except make calls worse |
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| 13:45.23 | ionix | can anybody try 1.800.599.3114 |
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| 13:48.22 | tparcina | asterfax, does anybody use it? |
| 13:48.50 | tparcina | is asterfax only for sending fax or it can also receive fax? |
| 13:50.08 | coppice | it can apparently receive, which is not obvious from their web page |
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| 13:52.43 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:52.45 | RoyK | tparcina: asterfax only uses app_rxfax and app_txfax |
| 13:53.27 | coppice | no it doesn't. it has other options, like eicon cards |
| 13:53.48 | RoyK | it still uses spandsp, doesn't it? |
| 13:54.44 | coppice | it can use spandsp, but it can use iaxmodem (OK, that's spandsp hiding), eicon hardware modems and other things now |
| 13:55.29 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 13:55.50 | bXi | i'm having a weird issue here |
| 13:55.57 | bXi | phone A can call phone B and talk |
| 13:56.02 | tparcina | ok, thank you all |
| 13:56.11 | tparcina | i'll try to install it tomorow |
| 13:56.12 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:56.16 | tparcina | now i have to go |
| 13:56.19 | bXi | the otherway it doesnt work |
| 13:56.25 | bXi | it hangs up immediatly |
| 13:56.27 | tparcina | thank you and see you tomorow |
| 13:56.27 | AsteriskAlbania | I am trying the call back, when somebody calls from mobile it brings 00355xxxxxxx I want to translate it to 0xxxxxxx |
| 13:57.03 | AsteriskAlbania | somebody help with dialplan ? |
| 13:57.06 | AsteriskAlbania | I am trying the call back, when somebody calls from mobile it brings 00355xxxxxxx I want to translate it to 0xxxxxxx |
| 13:57.11 | *** part/#asterisk tparcina (n=tparcina@lns01-1088.dsl.iskon.hr) |
| 13:57.14 | jbalcomb | bXi: does it say why in the logs? |
| 13:57.22 | AsteriskAlbania | is it possible ? |
| 13:58.10 | bXi | i'm looking at the CLI |
| 13:58.18 | bXi | it says that phone B answered |
| 13:58.27 | jbalcomb | AsteriskAlbania: it is possible |
| 13:58.37 | bXi | and then spawn extension (sip, number, 1) exited nonzero |
| 13:58.59 | bXi | and i have the verbosity level quite high |
| 13:59.03 | bXi | 255555555 or so |
| 13:59.04 | bXi | :p |
| 14:00.04 | jbalcomb | bXi: perhaps you could pastebin.ca the relevelant part of the screen? |
| 14:01.07 | bXi | jbalcomb: theres nothing different from a correct call on CLI :/ |
| 14:01.11 | AsteriskAlbania | jbalcomb: please tell me how |
| 14:01.31 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.88) |
| 14:01.37 | caio1982 | coppice: hello there, i'm playing with your T.38 code in asterisk and found a backported patch for asterisk 1.2.10 and i'm wondering whether it worth to try or not. do you think it's okay to use it in 1.2.x instead of trunk? |
| 14:01.39 | AsteriskAlbania | jbalcomb: exten => _X.,1,mcc2(${EXTEN}|nopickup) |
| 14:01.39 | ionix | can anybody try 1.800.599.3114 |
| 14:02.16 | coppice | caio1982: I have no idea what state it might be in |
| 14:03.27 | caio1982 | coppice: the patch provided by some third-party or the current code in trunk? |
| 14:04.21 | coppice | both really. we are finding that what is in trunk seems to have serious limitations in the real world. people don't follow the specs |
| 14:05.01 | *** join/#asterisk TeePOG (n=TeePOG@196.211.231.163) |
| 14:05.06 | coppice | in the current openpbx SVN code things have been reworked. T.38 passthrough and T.38 termination are both working pretty well with that |
| 14:05.08 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 14:05.44 | caio1982 | coppice: nice to hear that, i'll take a look at their repository |
| 14:05.48 | RoyK | coppice: using spandsp 0.0.3? |
| 14:05.57 | coppice | yes |
| 14:06.06 | RoyK | how about bridging? |
| 14:06.20 | RoyK | sip/zap |
| 14:06.25 | cybertrickle_ | I am using a callfile to originate a call . THe dialplan uses IAX to make the call using another server. It calls me, but it doesnt continue in the dial plan. Anyone know why it would do that? heres my callfile+extensions http://rafb.net/paste/results/ktzWV633.html |
| 14:06.27 | coppice | that is less well tested. we concentrated on termination |
| 14:06.43 | RoyK | I see |
| 14:07.52 | coppice | T.38 gateway is basically working too, but not in openpbx SVN. I am testing a new version with ECM support, then it will be time for a big system test of everything |
| 14:08.01 | AsteriskAlbania | I need to RIP 00355XXXXXXX from a incomming call to 0XXXXXXX anyone can help ? |
| 14:08.05 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 14:08.23 | RoyK | coppice: what is ECM? |
| 14:08.31 | coppice | error correction mode |
| 14:09.01 | coppice | this will be a pretty complete implementation of fax |
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| 14:11.31 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 14:13.28 | AsteriskAlbania | I need to RIP 00355XXXXXXX from a incomming call to 0XXXXXXX anyone can help ? |
| 14:14.11 | RoyK | coppice: that's nice :) |
| 14:14.21 | eurocrash | debugging... i get 503's on a new setup... trying the 'hello-world' playback test, tried level 10 debug, but not much more than sip-debug... how can i see more to determine 503 errors? |
| 14:14.37 | RoyK | coppice: so then we'll just wait till asterisk's pure GPL and port it over? |
| 14:14.38 | RoyK | rotfl |
| 14:15.49 | caio1982 | that's a quite interesting question, despite the funny part of it |
| 14:17.42 | dasenjo | Hi, ¿what could be a reason to get a 480 "emporarily Unavailable" response from a sip peer when the registry is OK and DND is not set? |
| 14:18.19 | coppice | I guess the main reason you'd get "emporarily Unavailable" is because someone can't spell :-) |
| 14:18.39 | caio1982 | haha |
| 14:19.47 | MrChimpy | arses. I'm getting "No authority found" when trying to connect via IAX2 from IAXComm on windows to asterisk box |
| 14:20.16 | MrChimpy | from tcpdump I can see correct username and passwords and the UDP getting there |
| 14:20.28 | MrChimpy | i'm trying to dial from iaxcomm through dialplan on asterisk |
| 14:21.24 | MrChimpy | Aug 17 15:21:14 NOTICE[11341]: chan_iax2.c:7203 socket_read: Host 10.1.240.105 failed to authenticate as tonyhsoft |
| 14:21.29 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 14:21.36 | MrChimpy | CAUSE : No authority found |
| 14:22.01 | dasenjo | coppice, yeah, maybe .. maybe my PC does not like capital t :p |
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| 14:35.11 | cybertrickle_ | I am using a callfile to originate a call . THe dialplan uses IAX to make the call using another server. It calls me, but it doesnt continue in the dial plan. Anyone know why it would do that? heres my callfile+extensions http://rafb.net/paste/results/ktzWV633.html |
| 14:35.51 | mut | woo |
| 14:36.00 | mut | manager interface has a ping pong |
| 14:36.05 | mut | BINGO! |
| 14:36.40 | *** join/#asterisk parag_ast (n=root@dxb-b16451.alshamil.net.ae) |
| 14:36.44 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
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| 14:36.56 | MrChimpy | wtf does "no authority found"? |
| 14:36.58 | MrChimpy | mean? |
| 14:37.10 | MrChimpy | my iax hardware phone is working with same config. |
| 14:38.11 | MrChimpy | dammit! |
| 14:38.21 | TrixVox | Are you sending the DID to the phone when it's expecting the IAX username? or vice-versa? |
| 14:38.31 | parag_ast | My voip provider is authenticating threw ip address.....so when i dial any international call it returns "Answered" even though the call is not picked up by the callled party and it showes in my cdr ......Kindly let me know the solution for this... |
| 14:38.31 | parag_ast | \ |
| 14:39.10 | MrChimpy | <PROTECTED> |
| 14:39.10 | MrChimpy | <PROTECTED> |
| 14:39.10 | MrChimpy | <PROTECTED> |
| 14:39.10 | MrChimpy | <PROTECTED> |
| 14:39.10 | MrChimpy | <PROTECTED> |
| 14:39.11 | MrChimpy | <PROTECTED> |
| 14:39.14 | MrChimpy | <PROTECTED> |
| 14:39.15 | MrChimpy | <PROTECTED> |
| 14:39.32 | MrChimpy | looks ok to me... |
| 14:40.04 | florinm | anyone has experience with el400 2FX/2FXO device? |
| 14:40.13 | florinm | 2FXS |
| 14:40.14 | TrixVox | what is your dial statement |
| 14:43.02 | mmealling | gah.... every tutorial I can find on setting up a SIP trunk assumes trixbox.... Just give me the sip.conf and extensions.conf files.... ghees.... |
| 14:43.31 | TrixVox | try searching for sip.conf instead of sip trunk |
| 14:43.45 | MrChimpy | hmm. cause code 50 is dropped frames.... |
| 14:46.16 | mmealling | TrixVox, thanks..... that helped! |
| 14:46.33 | mmealling | sorry.... just trying to get Asterisk to work before I start hiding things behind GUIs |
| 14:47.43 | parag_ast | TrixVox, My voip provider is authenticating threw ip address.....so when i dial any international call it returns "Answered" even though the call is not picked up by the callled party and it showes in my cdr ......Kindly let me know the solution for this... |
| 14:47.50 | *** join/#asterisk pbx1 (i=pbx1@netblock-66-245-193-85.dslextreme.com) |
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| 14:53.34 | florinm | anyone experience with VIP-450 ? |
| 14:53.41 | florinm | i try to conect it to asterisk |
| 14:53.45 | florinm | but i can't :( |
| 14:54.04 | florinm | and no dam info about it on net :( |
| 14:54.07 | mmealling | if the host that I'm running asterisk on is also my firewall running NAT, I shouldn't have to worry with NAT stuff since asterisk by default answers requests on all interfaces, right? |
| 14:55.00 | mmealling | assuming I don't tell it a bindaddr.... |
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| 15:06.30 | BudaH | dial 2011 |
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| 15:07.47 | *** join/#asterisk hohum (n=dcorbe@12.195.58.236) |
| 15:07.55 | puzzled | hi |
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| 15:15.18 | *** join/#asterisk WildPikachu (n=WildPika@unaffiliated/wildpikachu) |
| 15:15.25 | WildPikachu | wow lots of iax termination providers |
| 15:16.31 | ionix | yeh |
| 15:18.35 | WildPikachu | any recommendations, I would like to route a few of my clients international calls (to usa & uk) over VOIP .... I tried voiptalk but appears there is a 1-2s latency, also tried nufone with a 2s latency to usa ... my latency to the pabx is 250ms |
| 15:18.45 | TrixVox | Lots of bad iax providers too! |
| 15:18.52 | vader-- | has anyone have a problem with the asterisk voicemail where messages are getting deleted but only the wav file is being delete and not the txt file? |
| 15:19.06 | *** join/#asterisk vlt (n=dm@p54B33FE8.dip0.t-ipconnect.de) |
| 15:19.10 | WildPikachu | TrixVox, yea |
| 15:19.38 | WildPikachu | price doesn't REALLY bother me, I just need low latency and no 2s lag |
| 15:20.02 | *** join/#asterisk mopar_one (n=Jaymz@207.91.46.139) |
| 15:20.29 | vlt | Hello. How is the right expression for the an exten matching the following regex: "/0555[012].*/" |
| 15:20.38 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:21.04 | TrixVox | where are your clients? in the US? |
| 15:21.47 | vlt | This means 05550 and zero or more digits following, 05551 and zero or more ... |
| 15:22.13 | WildPikachu | brb |
| 15:23.06 | vlt | Can I use brackets for groups in extens, e. g. "_0555(0|[12].)"? |
| 15:24.13 | *** join/#asterisk Ebola (n=Ebola@user-54458db0.lns1-c13.telh.dsl.pol.co.uk) |
| 15:25.10 | vlt | Mmh, not as I just tried ... |
| 15:27.27 | vlt | So do i need two different extens for 0555-0 and 0555-10...29? |
| 15:27.39 | nicox | Hello, did anybody know if its possible to get T38 working with asterisk? |
| 15:28.02 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:28.26 | Ebola | Isn't that like, a fighter jet? |
| 15:28.49 | coppice | yeah. he means T.38 :-) |
| 15:29.16 | nicox | there is a t38-passthrough implemented in asterisk, what does that mean, can i passthrough a fax to pstn or only to another sip-user? |
| 15:29.45 | coppice | T38s help the space shuttle land safely. pity someone didn't more effort into helping it take off :-) |
| 15:30.34 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:30.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:31.22 | nicox | any answer? |
| 15:31.27 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 15:31.47 | nicox | did anybody testes T.38 Passthrough? |
| 15:31.53 | coppice | the T.38 code in * right now only does passthrough. no termination, or PSTN gateway |
| 15:31.55 | nicox | sorry, tested, |
| 15:32.11 | CunningPike | Morning, gents |
| 15:32.15 | nicox | thanks, thats what i have to know |
| 15:32.18 | nicox | morning |
| 15:32.34 | nicox | did anybody tested T.38 implementation from Spandsp? |
| 15:32.43 | coppice | I did |
| 15:32.52 | nicox | did it work? |
| 15:33.01 | coppice | I wrote it |
| 15:33.52 | nicox | you always mean spandsp? |
| 15:34.31 | nicox | cause there are 2 implementations, spandsp and from bugs.digium.com number 5090 |
| 15:34.47 | coppice | nicox: no there aren't |
| 15:35.50 | nicox | you make me cry |
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| 15:37.50 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
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| 15:39.08 | nicox | coppice: you writed the code? |
| 15:39.18 | coppice | yes |
| 15:39.57 | eKo1 | writed? |
| 15:40.04 | intralanman | you writ that? |
| 15:41.50 | nicox | sorry, my english is not so good, cause my mother language is not english. when is it planned to implement T.38 Gateway function to PSTN (PRI)? |
| 15:42.18 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:42.20 | coppice | i'm testing it now |
| 15:42.23 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:42.44 | nicox | can i get the code anywhere to test? |
| 15:43.27 | caio1982 | coppice: do you plan to port it to * later or are you writing it for openpbx only? |
| 15:43.35 | caio1982 | coppice: i mean, the termination code |
| 15:43.54 | coppice | not really. there are snapshots of spandsp with it in, but I haven't released any code to make it work in a PBX yet |
| 15:44.37 | coppice | caio1982: the termination code is working pretty well in openpbx now. that's the limit of what I am prepared to do for it |
| 15:46.21 | *** join/#asterisk switch (n=switch@61.206.115.5.user.ad.il24.net) |
| 15:46.39 | eKo1 | nicox: there are people whose native language is english and still make mistakes ;) |
| 15:47.16 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 15:47.54 | nicox | can i get a pre-alpha version of openpbx anywhere to test the code? |
| 15:49.15 | Corydon-w | nicox: openpbx is pretty well dead at this point |
| 15:49.27 | *** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com) |
| 15:49.27 | file | Corydon-w: I wouldn't say that |
| 15:49.44 | brad_mssw | coppice: T.38 termination is in OpenPBX already? is it integrated to the point where it will drop to the fax extension and rxfax will work? |
| 15:49.45 | intralanman | and the debates continue |
| 15:49.54 | nicox | corydon-w: and why anybody is programming for that? |
| 15:50.15 | Corydon-w | file: the conundrum is that if I say it's dead, somebody might care enough to revive it... but if nobody says anything, it'll stay dead |
| 15:50.29 | WildPikachu | TrixVox, my pabx is in Kelowna, British Columbia .... connectivity from my clients to there is fast |
| 15:50.45 | mmealling | cool.....got telasip routing calls to xlite, now to get calls routed from xlite to the outside world. |
| 15:51.00 | file | I don't consider it dead, the momentum isn't certainly what it was - but stuff is still happening |
| 15:51.35 | Corydon-w | file: and at the current pace, they might have a pre-alpha milestone ready in about 5 years |
| 15:53.00 | *** join/#asterisk JPinela (n=JPinela@static-b4-248-148.telepac.pt) |
| 15:53.06 | JPinela | hi |
| 15:53.40 | nicox | so, is there any chance for T.38? |
| 15:53.57 | WildPikachu | anyone feel free to msg me if you think you can provide me good quality termination :) |
| 15:54.45 | eKo1 | define 'good quality' |
| 15:54.52 | JPinela | I need a little help, on getting started, pelase......... |
| 15:54.58 | JPinela | any1? |
| 15:55.07 | file | JPinela: if you ask a question, someone might answer |
| 15:55.20 | mog | you can do t.38 with spandsp and asterisk |
| 15:55.20 | WildPikachu | well .. over 1s lag is a bit annoying and my clients think i'm dumb :) |
| 15:55.27 | eKo1 | JPinela: step 1, turn off your computer |
| 15:55.35 | mog | you can do pass through with whats in asterisk |
| 15:55.56 | JPinela | file.... ok..... |
| 15:56.06 | TrixVox | WildPikachu, what's your latency to connect01.voicepulse.com? |
| 15:56.15 | JPinela | eko1... ok.... u first.... so I can c how |
| 15:57.19 | eKo1 | JPinela: Done. Your turn. |
| 15:57.34 | JPinela | eKo1: ok.... ... also done |
| 15:58.02 | WildPikachu | TrixVox, 105ms |
| 15:59.00 | eKo1 | JPinela: very good. Step 2, repeat step 1. |
| 15:59.01 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:59.03 | coppice | mog: you could. it won't get you very far as it stands, but you could :-) |
| 15:59.28 | TrixVox | If that's acceptable to you, I'd definitely recommend them |
| 15:59.53 | TrixVox | them being voicepulse connect for asterisk -- connect.voicepulse.com |
| 16:00.05 | *** join/#asterisk erauqssidlroweht (n=walkerbo@co.nezperce.id.us) |
| 16:00.11 | JPinela | what does the extensions.conf do anyway??? I almost erased the hole file, and I can still make calls between Ipphones.... but a little extension I added, doesn't do anything... |
| 16:00.16 | JPinela | eKo1 on it |
| 16:00.21 | Qwell | erauqssidlroweht: That nick hurts my eyes... |
| 16:00.26 | erauqssidlroweht | Hello all. Anyone know a lot about Cisco Ip Phones? |
| 16:00.37 | erauqssidlroweht | Hee hee |
| 16:01.13 | Nugget | why don't you just ask your question? You don't care how much we know, just that we know the answer to whatever question you're avoiding actually asking. |
| 16:01.46 | erauqssidlroweht | Good answer. Thanks |
| 16:01.47 | erauqssidlroweht | :-) |
| 16:02.02 | erauqssidlroweht | I have a 7970 that keeps 'rebooting' after I did a factory restore. |
| 16:02.31 | JPinela | JPinela anyone with some usefull info? |
| 16:02.33 | Nugget | that's usually because your OS79XX.TXT and the phone's config file disagree about which version to use. |
| 16:02.44 | Qwell | get the firmware, setup dhcpd, tell it where your tftpd is |
| 16:02.58 | Nugget | make sure SIPDefault and OS79XX agree |
| 16:03.13 | intralanman | JPinela: RTFM |
| 16:03.15 | erauqssidlroweht | I have Windows for DHCP |
| 16:03.18 | Qwell | I don't think 7970 uses OS79XX |
| 16:03.21 | erauqssidlroweht | I've reserved it an IP address |
| 16:03.24 | WildPikachu | TrixVox, that gives me 450ms to the worst of my clients .... i wonder what the latency is from them to US48 and europe |
| 16:03.27 | JPinela | intralanman ... what is that? |
| 16:03.39 | Nugget | ah, ok, I've never touched a 7970. |
| 16:03.49 | intralanman | Read The Friendly Manual |
| 16:03.57 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
| 16:04.00 | JPinela | intralanman and where is that? |
| 16:04.01 | erauqssidlroweht | I have the newest firmware |
| 16:04.23 | intralanman | generally the same place you got the software |
| 16:04.32 | erauqssidlroweht | And I just intalled PumpKIN TFTP |
| 16:04.43 | JPinela | intralanman i've been through dozens of pages, but I can't get a conclusive answer...... |
| 16:05.02 | JPinela | intralanman it just helps installing |
| 16:05.24 | coppice | brad_mssw: the T.38 termination in openpbx seems to be working pretty well. someone else has been doing some changes so a fax extension will automatically invoke rxfax. I think that is working now |
| 16:05.38 | JPinela | intralanman just answer me this: why is it, with almost no extensions.conf, I can still make calls, and answer them? |
| 16:05.54 | intralanman | http://www.asterisk.org/support |
| 16:06.14 | brad_mssw | coppice: that's cool |
| 16:06.18 | intralanman | some pretty decent docs there |
| 16:06.18 | Nugget | JPinela: clearly because "almost" is "enough". Stop asking stupid questions. |
| 16:06.19 | JPinela | intralanman that's the site where I found this channel.... |
| 16:06.41 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 16:06.46 | brad_mssw | coppice: now if I only had V150 support :) |
| 16:07.02 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 16:07.07 | JPinela | Nugget the "almost" doens't concern those calls....... and pardon me for not being born with the whole knowledge of the universe! |
| 16:07.10 | coppice | do people really want V.150? I thought it was a protocol looking for a problem |
| 16:07.17 | Nugget | what did you expect to hear. "OK, You caught us! extensions.conf doesn't really do anything, it's all a big trick!" |
| 16:07.24 | intralanman | lmao |
| 16:07.43 | erauqssidlroweht | I'm looking through the DHCP options there is no "Bootp" or "TFTP" in the list to use. Where do I add the TFTP address? |
| 16:07.49 | JPinela | Nugget I simply would like an answer to the question. or.... you can simply say...... " I don't know^" |
| 16:07.50 | brad_mssw | coppice: well, we deal with a lot of modem stuff here, credit card processing software and all, it's amazing how much stuff requires dialup still |
| 16:07.52 | Qwell | erauqssidlroweht: in a non-stupid dhcpd |
| 16:07.54 | *** part/#asterisk pyrom (n=pyro@86.84-48-44.nextgentel.com) |
| 16:07.59 | Nugget | your question is invalid. |
| 16:08.10 | JPinela | Nugget oh really........ that's cute...... |
| 16:08.11 | Nugget | and you don't want an answer, you just want to bitch. |
| 16:08.20 | Nugget | if you wanted an answer you'd have asked a valid question. |
| 16:08.27 | *** join/#asterisk RickNZ (n=rick@ip-202-37-229-70.internet.co.nz) |
| 16:08.28 | brad_mssw | coppice: we're all voip here, it seems to work _ok_ using ulaw over a 100Mbps connection back to a land-line though |
| 16:08.39 | erauqssidlroweht | Is there a way to assign TFTP with Windows Server 2003? |
| 16:08.47 | JPinela | Nugget ......... I asked a valid question. and U aren't a stupid computer or program to answer "invalid question" |
| 16:08.47 | intralanman | JPinela: we gave you the answer.... read up on it |
| 16:08.49 | coppice | brad_mssw: I bought a couple of FAX modems this week for further testing against spandsp. People treat enquiries about modems like you're enquiring about flint knives |
| 16:08.50 | brad_mssw | coppice: some higher-speed connections crap-out though |
| 16:08.55 | Nugget | no you did not. |
| 16:09.17 | JPinela | intralanman like I said, the RTFM u pointed out, simply helps installing...... |
| 16:09.19 | coppice | brad_mssw: if any of that stuff works, its by luck and not design |
| 16:09.51 | brad_mssw | coppice: yeah, i know :/ |
| 16:10.05 | JPinela | intralanman and the sites, i've been to, suggest, that the following code, would do something. but obviously, something is missing... |
| 16:10.06 | brad_mssw | coppice: hence v150 would be nice :) |
| 16:10.11 | Dr-Linux|work | guys, mgp123 is taking 50% of my CPU, is it fine? or what should i do? |
| 16:10.12 | intralanman | you mean all those documents only help with installation? |
| 16:10.14 | intralanman | BS |
| 16:10.27 | JPinela | exten => 130,1,Answer() |
| 16:10.27 | JPinela | exten => 130,2,Playback(pbx-invalid) |
| 16:10.27 | JPinela | exten => 130,3,Hangup() |
| 16:10.46 | JPinela | intralanman ..... no BS |
| 16:10.53 | coppice | I've like to get a spec for the V.22 fast connect that a lot of these POS boxes use. its proprietary, though. seems you need to reverse engineer it |
| 16:11.25 | WildPikachu | anyone got examples of bad iax termination companies, or a review site might be a better idea? |
| 16:11.32 | WildPikachu | i mean ... got a review site i can visit? |
| 16:11.53 | Juggie | isnt v.22 a standard modem protocol? |
| 16:12.33 | coppice | V.22 fast connect is a proprietary extension of V.22, heavily used in POS terminals. It, er, fast connects :-) |
| 16:12.44 | intralanman | JPinela: YES BS.... one of the links on that page will get you to the asterisk handbook which has an entire list of application as well as what they do and how they're used |
| 16:13.00 | Juggie | coppice, http://en.wikipedia.org/wiki/V.22 |
| 16:13.01 | intralanman | s/application/applications/ |
| 16:13.10 | cybertrickle_ | Anyone ever had problems making remote calls with a callout file, like the channel is IAX/sfax/number or SIP/number@server, it hangs up on me every time. But when I do it through the dialplan, it works. |
| 16:13.12 | Juggie | i think you mean V.22Bis |
| 16:13.17 | Dr-Linux|work | intralanman, mgp123 is taking 50% of my CPU, is it fine? or what should i do? |
| 16:13.19 | intralanman | tx jbot |
| 16:13.32 | intralanman | kill it |
| 16:13.34 | intralanman | jk |
| 16:13.41 | JPinela | .. |
| 16:13.43 | coppice | Juggie: Duh! |
| 16:13.54 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 16:13.55 | Dr-Linux|work | intralanman, are you talking to me? kill it? |
| 16:13.56 | coppice | Juggie: I think I mean V.22 fast connect |
| 16:14.17 | intralanman | Dr-Linux|work: depends ..... what CPU and how many concurrent calls |
| 16:14.22 | Juggie | coppice, i'm only saying that because i've only ever heard of v22 and v22 bis. |
| 16:14.35 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 16:14.46 | coppice | Juggie: then you haven't worked with POS terminals. I said its proprietary |
| 16:15.07 | Dr-Linux|work | intralanman, concurrent calls are only 2 or 4, but mpg123 is taking 48% CPU |
| 16:15.18 | sevard | does anyone know to get the email server hostname for using SMS with a certian carrier? |
| 16:15.31 | sevard | I know the carrier, well, I think I do, from whitepages.com, but I can't find the server address |
| 16:15.38 | syzygyBSD | Dr-Linux|work: do you have mpg123 or mpg321? |
| 16:15.52 | Dr-Linux|work | mpg123 |
| 16:15.55 | erauqssidlroweht | To answer my own question. Server 2003 option 066 has the TFTP boot peramiter. |
| 16:16.01 | benjk | dont use mpg123 |
| 16:16.02 | Dr-Linux|work | but i insalled mpg321 as well |
| 16:16.05 | benjk | use madplay |
| 16:16.13 | syzygyBSD | Dr-Linux|work: they conflict |
| 16:16.17 | sevard | they will. |
| 16:16.28 | benjk | neither of those are any good |
| 16:16.30 | JPinela | intralanman didn't see that link...... going to investigate thanks. but one more question: if I completely empty my extensions.conf file, will the phones still be able to communicate? Yes, or No? |
| 16:16.42 | syzygyBSD | benjk: why not? |
| 16:16.50 | Dr-Linux|work | syzygyBSD, well, the other one is not running, also i'm not facing any problem, but i'm worried about it |
| 16:16.50 | intralanman | JPinela: only one way to find out ;) |
| 16:16.56 | benjk | mpg321 won't work for asterisk |
| 16:17.14 | syzygyBSD | benjk: i could make it work if I wanted to |
| 16:17.15 | benjk | mpg123 is no longer maintained and has various security issues |
| 16:17.17 | Dr-Linux|work | benjk, madplay? |
| 16:17.18 | JPinela | intralanman no. 2 ways. doing, or asking someone who knows allerady |
| 16:17.28 | *** join/#asterisk Kylun (i=StarHawk@adsl-068-157-090-228.sip.bct.bellsouth.net) |
| 16:17.30 | JPinela | intralanman I would first like your opinion |
| 16:17.30 | benjk | madplay is the way to go |
| 16:17.49 | syzygyBSD | You just have to update the input/output to be the correct number of channels and frequencies, easy to do with sox |
| 16:18.01 | Dr-Linux|work | :S |
| 16:18.07 | benjk | you can use sox as a player too |
| 16:18.28 | benjk | and if you built sox with libmad it can play mp3 too |
| 16:18.32 | syzygyBSD | doesn't cat work too |
| 16:18.34 | Dr-Linux|work | benjk, i'm using sox as well, into my Asterisk for mixing the calls |
| 16:18.36 | JPinela | intralanman and why didn't that code I posted earlier, work? any idea? |
| 16:18.37 | intralanman | my opinion is it depends..... largely on whether you're using ARA (not likely) and whether you reload after changes (i'm thinking that's also not likely).... so my guess would be..... it wouldn't do anything differently |
| 16:18.58 | syzygyBSD | basically, what we are saying is you can do anything! |
| 16:19.06 | Dr-Linux|work | :S |
| 16:19.15 | Dr-Linux|work | what should i use |
| 16:19.15 | erauqssidlroweht | term71. default.loads requested |
| 16:19.18 | JPinela | intralanman ARA, I have no idea what that is. after changes: yes, I reload, or reboot. |
| 16:19.46 | benjk | you should be using sox with madlib, or another player using madlib |
| 16:19.47 | CunningPike | Dr-Linux|work: Neither - use native MOH |
| 16:19.52 | benjk | or that |
| 16:19.53 | Dr-Linux|work | i believe there is an MOH player .. or something that comes with asteirsk new versions .. maybe anthm build it? |
| 16:20.00 | CunningPike | ~moh |
| 16:20.02 | jbot | extra, extra, read all about it, moh is Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
| 16:20.02 | vader-- | has anyone have a problem with the asterisk voicemail where messages are getting deleted but only the wav file is being delete and not the txt file? |
| 16:20.23 | intralanman | and, JPinela, yes.... i have several ideas as to why that wouldn't work..... and they all lead me back to RTFM so we don't have to teach you asterisk here |
| 16:20.27 | Dr-Linux|work | CunningPike, hey :) |
| 16:20.31 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 16:20.33 | JPinela | ....... |
| 16:20.49 | CunningPike | vader--: That has come up before....... but I can't remember what it was. Try searching the list archives - it was on there a while back |
| 16:21.05 | CunningPike | Dr-Linux|work: Hi there |
| 16:21.15 | JPinela | intralanman if u know, why can´t u simply point it out in general topics? |
| 16:21.24 | Dr-Linux|work | CunningPike, when different experts give me different suggestions .. i really get confused :S |
| 16:22.05 | Dr-Linux|work | CunningPike, i know benjk is an expereinced guy .. he is suggestion madplay .. and MOH native .. |
| 16:22.15 | Dr-Linux|work | not sure where should i go :S |
| 16:22.22 | coppice | brad_mssw: I think when T.38 is polished, the code base could be turned into V.150 without a mountain of work |
| 16:22.26 | CunningPike | JPinela: extensions.conf contains your dialplan - depending on what you need, 'exten => _X.,1,Dial(SIP/${EXTEN})' will allow all your phones to call each other |
| 16:22.43 | JPinela | CunningPike huh........ thks |
| 16:22.54 | CunningPike | JPinela: However, if you need more stuff, like voicemail access, you need more stuff in there |
| 16:22.55 | benjk | whatever you do, don't use mpg123 |
| 16:23.44 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 16:24.33 | CunningPike | JPinela: I can get a PC to boot with just a handful of files on its boot partition. So, why do we need all the other stuff? |
| 16:25.19 | intralanman | i think everyone should know.... CunningPike is a really nice person |
| 16:25.26 | intralanman | :) |
| 16:25.31 | JPinela | CunningPike thks. the more complex stuff, I'll worry about it latter. my problem is, I can't get a simple thing to work. after that it will be easier |
| 16:25.45 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 16:25.53 | CunningPike | intralanman: What brought that on? I'm an asshole, really |
| 16:26.02 | intralanman | ru? |
| 16:26.22 | intralanman | the time and care you took with JPinela..... i just ASSumed |
| 16:26.36 | CunningPike | ;) |
| 16:26.52 | CunningPike | (It's the quickest way to deal with them) |
| 16:27.08 | sevard | does anyone know? i'm stuck as hell :\ |
| 16:27.12 | intralanman | yeah, sometimes the alternitives are more fun though |
| 16:27.17 | CunningPike | lol |
| 16:27.26 | intralanman | s/alternitives/alternatives/ |
| 16:27.28 | CunningPike | sevard: Wassup? |
| 16:27.34 | intralanman | jbot too slow again |
| 16:27.35 | intralanman | lol |
| 16:27.49 | vader-- | CunningPike what should i search for in regards to the voicemail? |
| 16:27.50 | CunningPike | Dr-Linux|work: |
| 16:27.52 | CunningPike | ~moh |
| 16:27.54 | jbot | moh is probably Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
| 16:27.54 | sevard | CunningPike: I'm trying to find out which email server for which carrier to send SMS to |
| 16:28.24 | intralanman | ~dial |
| 16:28.26 | intralanman | no? |
| 16:28.31 | intralanman | hmmm |
| 16:28.31 | CunningPike | vader--: Try 'voicemail text file not deleted' or something |
| 16:28.34 | vader-- | CunningPike whats the url to the list? |
| 16:28.46 | CunningPike | vader--: Use google |
| 16:29.00 | CunningPike | vader--: 'Groups' in google |
| 16:29.22 | CunningPike | sevard: What carrier |
| 16:30.05 | CunningPike | sevard: It's often (but not always) number@msg.carriersdomain |
| 16:30.12 | sevard | CunningPike: I can't find any database holding this information |
| 16:30.30 | CunningPike | sevard: Why would there be one? |
| 16:30.37 | sevard | CunningPike: well whitepages.com I think has the wrong carrier, it lists it as American Cellular Corporation |
| 16:30.44 | sevard | but I think it's actually unicel or something |
| 16:30.55 | sevard | CunningPike: is there a better way to find out? |
| 16:32.01 | CunningPike | sevard: You could try googling (again!) :) |
| 16:32.33 | brad_mssw | coppice: that's good to know, definitely appreciate the work you've put into T.38 and all :) |
| 16:32.35 | intralanman | sevard: you could meet a lot of people and eventually you'd have one from every carrier |
| 16:32.39 | intralanman | lol |
| 16:33.45 | sevard | ;/ |
| 16:34.52 | RoyK | ~ecfo |
| 16:34.53 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 16:35.34 | RoyK | ~mg2 |
| 16:35.40 | WildPikachu | anyone here use connect2voice? |
| 16:36.25 | coppice | Echo Canceler Freak Out happens when the echo canceller suddenly realises its a crappy design based on a half baked 20 year old apps note. |
| 16:36.28 | vader-- | if someone is saving a voicemail what files should be in the Old directory? |
| 16:36.32 | vader-- | i have 4 |
| 16:36.35 | JPinela | CunningPike thks. but...... getting a pc to boot it's simple.... it's just booting.... my surprise is the communication he establishes, without the extensions.conf.... |
| 16:36.40 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 16:36.57 | vader-- | msg0000.gsm, msg0000.txt, msg0000.wav, msg0000.WAV |
| 16:37.52 | coppice | ~ecfo is also what happens when the echo canceller suddenly realises its a crappy design based on a half baked 20 year old apps note. |
| 16:37.53 | jbot | okay, coppice |
| 16:38.04 | coppice | ~ecfo |
| 16:38.06 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly ... |
| 16:38.59 | coppice | what happened. jbot can usually handle a longer definition than that |
| 16:42.32 | Nivex | ~more |
| 16:42.34 | jbot | i heard more is Displays the contents of the named files, one screenful at a time. Syntax: more (file1) (file2) ...(fileN). Where file1 through fileN are the files to display. Example: more papers/history-final displays the file papers/history-final. |
| 16:42.38 | Nivex | bah |
| 16:42.43 | JPinela | (j debia |
| 16:42.55 | benjk | coppice, its run out of stack :) |
| 16:43.13 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 16:43.14 | coppice | ~documentation |
| 16:43.16 | jbot | from memory, documentation is not the issue for why the new stuff isn't completed |
| 16:43.29 | coppice | ~doc |
| 16:43.30 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 16:43.58 | coppice | where's something with an enormous list of URLs |
| 16:44.36 | intralanman | i'll have to remember that one |
| 16:44.58 | Qwell | ~slashdot |
| 16:45.31 | intralanman | jbot: why aren't thos links;) |
| 16:45.34 | Qwell | Why would somebody want to replace anything with VMS? |
| 16:45.35 | coppice | well that looks longer than ecfo |
| 16:45.42 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net) |
| 16:46.12 | coppice | its reliable? it clusters well? its as slow as a dog? |
| 16:46.27 | WildPikachu | i wonder if XO.com has good service |
| 16:46.41 | intralanman | WildPikachu: pretty good, yeah |
| 16:46.51 | benjk | is it VMs or VMS ? |
| 16:47.00 | WildPikachu | any xo resellers? heheh |
| 16:47.01 | Qwell | benjk: VMs :p |
| 16:47.33 | benjk | I guess coppice read that as OpenVMS |
| 16:47.33 | coppice | benjk: but where's the potential for bad humour in that? |
| 16:47.43 | benjk | not that the "Open" in OpenVMS means anything |
| 16:47.59 | coppice | Open to abuse, of course |
| 16:48.03 | AndyCap | voice mail system? :) |
| 16:48.25 | coppice | its from the era of open systems. opening the floodgates for MS to win, i think |
| 16:48.40 | Qwell | MS to win? Nice unintentional pun |
| 16:49.08 | benjk | they should have open sourced VMS at the time, the world would be a better place today |
| 16:51.51 | benjk | in the end they were trying to protect their turf with the outcome that they got picked up by a texas outlet known for cloning half heartedly cobbled together home computing designs by IBM intended to compete with the C-64 |
| 16:54.18 | coppice | after cloning the IBM PC, they cloned the cheapo taiwanese machines, by giving their expensive servers a case so flimsy you couldn't pick them up without bending them. :-) |
| 16:57.33 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 17:02.00 | *** join/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 17:02.53 | *** join/#asterisk Givemelove (n=non@208.57.229.162) |
| 17:03.20 | StewLG | I have both physical SIP phones and SIP VOIP providers in my sip.conf file. When my internet goes down, my physical SIP phones don't register any more, effectively disabling my phone system whenever the internet goes down. Is this normal? Is there something I can do about it? |
| 17:04.42 | eKo1 | StewLG: No it isn't normal. Are the SIP phones and * on the same lan? |
| 17:04.51 | StewLG | eko1: Yes. |
| 17:04.59 | StewLG | 10.0.0.x subnet. |
| 17:05.31 | StewLG | If I take out the external SIP providers from sip.conf, while the internet is down, the physical sip phones register. |
| 17:06.09 | StewLG | Put them back in, and they never register. The debug output implies Asterisk is spending all its time worrying about the SIPs it is supposed to register. |
| 17:06.30 | StewLG | (But I'm not expert enough to be sure about that.) |
| 17:06.32 | eKo1 | StewLG: What version of * are you using? |
| 17:06.42 | StewLG | 1.2. something. |
| 17:06.48 | StewLG | I can chekc. |
| 17:07.47 | Vorondil | hi all, quick question. i'm using queues to ring different groups of phones in our office depending on what an incoming caller chooses in an auto-attendant menu. when someone calls and rings, say, the software department, and one of the software guys has his phone off the hook, but hangs up before his group stops ringing, he's effectively not in the group until the next call. is there any way to jump back in mid-ring and pick up a call? |
| 17:07.51 | eKo1 | That is a strange problem. How are you registering with your providers? |
| 17:08.02 | eKo1 | Vorondil: that is not a quick question :( |
| 17:08.05 | StewLG | "Asterisk 1.2.7.1 built by buildd @ rothera on a i686 running Linux on 2006-05-26 01:42:12 UTC" |
| 17:08.43 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
| 17:08.44 | StewLG | I'll dig up the whole SIP file and pastebin it.. hang on a moment.. |
| 17:09.05 | RoyK | coppice: ping |
| 17:09.06 | Vorondil | eKo1: hehe, well, i intended it to be, but that apparently went awry |
| 17:10.02 | eKo1 | no kidding |
| 17:10.17 | eKo1 | well, I've never dealt with queues so I can't really help. |
| 17:10.55 | StewLG | eko1: Here's the sip file: http://pastebin.ca/136391 |
| 17:11.21 | [TK]D-Fender | Vorondil: Actually it IS a quick question, with an equally quick answer : NO. |
| 17:12.13 | [TK]D-Fender | Vorondil: once the dial is in progress there is no way to inject any other extra checks without a big rewrite of app_dial. Translation : forget it. |
| 17:13.08 | eKo1 | <PROTECTED> |
| 17:13.17 | CunningPike | StewLG: Are you using IP addresses or hostnames anywhere? |
| 17:13.38 | StewLG | eko: I believe you are probably right. |
| 17:13.44 | CunningPike | vader--: You will have a msg00n.format file for each format you use, plus the .txt file for each message |
| 17:13.46 | StewLG | cunnigpike: I don't understand the question? |
| 17:13.57 | Vorondil | [TK]D-Fender: ah, i see. i figured that it's either impossible or extraordinarily hard to do |
| 17:14.03 | Vorondil | [TK]D-Fender: thanks though. ^_^ |
| 17:14.06 | StewLG | I use FQDN in the sip.conf file, if that's what you are asking. |
| 17:14.12 | StewLG | Not IP addresses. |
| 17:14.17 | CunningPike | StewLG: Sorry - I was wondering if you are using hostnames in your sip.conf |
| 17:14.36 | StewLG | Cunning: So, yes, I am. Is that a problem? |
| 17:14.46 | StewLG | Strange that using DNS should be discouraged.. |
| 17:14.52 | *** join/#asterisk trnygaar (i=hFizYPNK@antapex.odalen.com) |
| 17:14.58 | CunningPike | StewLG: Asterisk has a problem whereby if it can't resolve a FQDN (because the DNS is unavailable) it stops responding |
| 17:15.04 | StewLG | Aha! |
| 17:15.10 | StewLG | Is this considered a bug? |
| 17:15.14 | trnygaar | Any easy way to show codecs in use in cli? Not sure which codec they negotiate to use |
| 17:15.31 | CunningPike | StewLG: It is a limitation for sure - but you can either use a local DNS cacheing proxy or use IP addresses instead |
| 17:15.47 | Qwell | trnygaar: on of the "show channels" either "show channels" or "sip show channels" shows the codec.. I can never remember which though |
| 17:15.59 | StewLG | Cunning: I guess I need to add one more thing to IPCop then... |
| 17:16.16 | trnygaar | Qwell: thx |
| 17:16.48 | CunningPike | JPinela: With SIP, most of the intelligence is on the phone - it is possible to have a network of SIP phones that can call each other without any pbx, provided the phones know how to contact each other |
| 17:17.47 | CunningPike | Is anyone else getting tired of the endless emails from RedHat Network? |
| 17:18.03 | [TK]D-Fender | trnygaar: "show channels" |
| 17:18.16 | *** part/#asterisk viler (i=1000@200.114.70.228) |
| 17:18.19 | CunningPike | StewLG: IPCop? |
| 17:19.08 | [TK]D-Fender | Qwell: So Now with you working for Digium, what spefic aspects should we be seeing your name coming up associated with? |
| 17:19.15 | Qwell | [TK]D-Fender: code |
| 17:20.04 | file | no no, bugs |
| 17:20.18 | eKo1 | coding bugs? |
| 17:20.28 | Qwell | code..bugs..same thing |
| 17:20.49 | russellb | he's going to be working hard on the bug tracker! |
| 17:21.00 | russellb | Qwell: don't be afraid to commit stuff |
| 17:21.02 | russellb | have at it |
| 17:21.06 | file | COMMIT! |
| 17:21.13 | Qwell | I just committed something the other night :p |
| 17:21.15 | russellb | you'll be informed if you do something stupid :) |
| 17:21.19 | russellb | i know, but MORE |
| 17:21.22 | Qwell | heh |
| 17:21.24 | file | by trillions of people around the world |
| 17:21.35 | Qwell | working on something else atm :) In time... |
| 17:21.41 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 17:21.50 | Katty | allo. |
| 17:21.56 | Qwell | Katty: :D |
| 17:22.18 | Katty | i have issues =< |
| 17:22.24 | Qwell | we knew that |
| 17:22.31 | Katty | teehee |
| 17:22.34 | Katty | so true |
| 17:22.36 | Katty | but! |
| 17:22.47 | Katty | http://pastebin.ca/136403 <- i think i just need to update my source list. whatcha's think? |
| 17:23.51 | eKo1 | that isn't an * question... |
| 17:23.52 | Katty | it installed and everything...but i'm getting errrrrrorsss. |
| 17:24.05 | Katty | eKo1: which is why i didn't ask you. |
| 17:24.09 | Katty | eKo1: run along and play. |
| 17:24.42 | eKo1 | maybe you'd better ask in #debina |
| 17:24.46 | eKo1 | err, #debian |
| 17:25.04 | Katty | hmm... no (= |
| 17:25.36 | eKo1 | and the question wasn't asked to anyone in particular so... |
| 17:25.47 | Katty | details, details. |
| 17:25.51 | CunningPike | In my experience, #debian makes this channel look like a Royal tea party |
| 17:25.59 | Katty | Qwell, file, would you have a look at my pastebin? |
| 17:26.10 | Katty | twisted[asteria]: or you, if you're around. |
| 17:27.54 | CunningPike | Katty: It looks like maybe one of the mirrors is down and your config doesn't contain any more? You could try updating your mirror list |
| 17:28.19 | Katty | CunningPike: precisely. exactly what i typed after my pastebin. |
| 17:28.23 | Katty | CunningPike: only one problem. |
| 17:28.31 | Katty | CunningPike: i don't actually remember how to do that anymore ;) |
| 17:28.40 | CunningPike | Aha! |
| 17:28.59 | Katty | i'm doomed! =< |
| 17:29.19 | justinu|laptop | katty lives! mew |
| 17:30.05 | Katty | justinu|laptop: she does! |
| 17:30.18 | Katty | justinu|laptop: how crazy is that (= |
| 17:30.50 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:31.17 | CunningPike | Katty: /etc/apt/sources.list contains the mirrors that you are using now |
| 17:31.17 | malverian | Has anyone experienced issues with getting blank caller id from sprint mobile phones? |
| 17:31.17 | Katty | CunningPike: ah ha! |
| 17:31.39 | CunningPike | Katty: And http://www.debian.org/mirror/list has a complete list of all mirrors |
| 17:31.53 | Katty | CunningPike: ! |
| 17:31.59 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:32.00 | Katty | CunningPike: you get the awesome award for the day. |
| 17:32.07 | TripleFFFF | can i add host=1.2.3.4/24 ? |
| 17:32.11 | TripleFFFF | in sip pseers ? |
| 17:32.15 | CunningPike | Katty: Just in case you want to get your sources from Chile or something |
| 17:32.30 | TripleFFFF | i mean i need to be able to add a cllass.. |
| 17:32.36 | Katty | CunningPike: i'm diggin Chile. |
| 17:32.43 | CunningPike | Katty: ;) |
| 17:32.44 | TripleFFFF | unless its host=dynamic.. and accept=1.2.3.4/24 |
| 17:32.50 | justinu|laptop | mmmm chili |
| 17:33.11 | CunningPike | How come no-one ever eats chilli sin carne? |
| 17:33.29 | Katty | because i don't eat meat. |
| 17:33.35 | Katty | kinda puts a stopper on the carne bit |
| 17:33.47 | justinu|laptop | there's some ok vegetarian chili recipes |
| 17:33.52 | TripleFFFF | guys ? |
| 17:33.53 | CunningPike | Katty: Chilli sin carne would be perfect for you then |
| 17:34.03 | jbroome | con == with sin == w/o |
| 17:34.04 | CunningPike | TripleFFFF: Sorry - got carried away there |
| 17:34.13 | TripleFFFF | can i add host=1.2.3.4/24 ? |
| 17:34.18 | TripleFFFF | or i use permit=192.168.40.0/255.255.255.0 |
| 17:34.25 | TripleFFFF | and host=dynamic |
| 17:34.27 | Katty | CunningPike: =< |
| 17:34.30 | TripleFFFF | its from invboudn from a class c |
| 17:34.34 | CunningPike | TripleFFFF: AFAIK, I don't think so for host - permit works |
| 17:34.53 | TripleFFFF | well then how can i tell sip.conf to accept all class c for one peer ? |
| 17:35.17 | CunningPike | TripleFFFF: I'm not sure you can - why would you want to be able to? |
| 17:35.25 | *** join/#asterisk svenna (n=svenna@p548D233E.dip0.t-ipconnect.de) |
| 17:36.32 | TripleFFFF | ? |
| 17:36.44 | TripleFFFF | caus guys sending inbound to us.. have like 200 servers |
| 17:36.47 | TripleFFFF | on a class c |
| 17:36.50 | TripleFFFF | i cant make 200 entries |
| 17:37.22 | CunningPike | TripleFFFF: And permit won't work for you? |
| 17:37.49 | Katty | jbroome: oh. |
| 17:38.02 | Katty | jbroome: suddenly chilli sin carne sounds more appealing. |
| 17:38.03 | CunningPike | TripleFFFF: I guess it doesn't give you ip authentication if you are allowing access from other networks |
| 17:38.17 | jbroome | :) |
| 17:38.27 | Katty | CunningPike: all these sources are ftp based, and my source list has a couple http based. |
| 17:38.36 | Katty | CunningPike: is that normal? |
| 17:38.42 | CunningPike | Katty: Shouldn't matter - they both work |
| 17:38.47 | Katty | m'kay |
| 17:38.56 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 17:38.58 | TripleFFFF | but if i use permit.. i use host=dynamic right ? |
| 17:39.17 | CunningPike | TripleFFFF: Yes, I would think so |
| 17:39.19 | Katty | CunningPike: and it won't matter if i'm running sarge or woody or whatever, right? |
| 17:39.23 | TripleFFFF | ok llet me try |
| 17:39.28 | Katty | CunningPike: as long as i keep my sarge thing in the source lsit |
| 17:39.53 | CunningPike | Katty: It does matters - but yes, you need to specify your build in the source list |
| 17:39.57 | CunningPike | As you say |
| 17:40.01 | *** join/#asterisk babyju (n=babyju@151.202.195.132) |
| 17:40.40 | Katty | well. |
| 17:40.43 | Katty | i appear to be in a pickle then. |
| 17:40.51 | CunningPike | Katty: Oh? |
| 17:41.19 | [TK]D-Fender | Katty: Mew. |
| 17:41.34 | Katty | [TK]D-Fender: mew. |
| 17:41.39 | Katty | CunningPike: i'm pastebinning |
| 17:41.44 | CunningPike | ok |
| 17:41.58 | [TK]D-Fender | Katty: Long time no see.... what drags you back to this pit of ours? |
| 17:42.00 | Katty | it'd help if vista didn't keep lockering up and annoying me |
| 17:42.07 | Katty | [TK]D-Fender: work finally slowed down a bit. |
| 17:42.14 | Katty | [TK]D-Fender: but i got about a grand of royalties off it (= |
| 17:42.26 | [TK]D-Fender | Katty: $ = Good |
| 17:42.48 | Katty | CunningPike: http://pastebin.ca/136427 |
| 17:42.55 | Katty | CunningPike: originally, i put woody on. |
| 17:43.04 | Katty | CunningPike: then decided that was riddicirus, and changed everything to sarge |
| 17:44.10 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 17:44.30 | CunningPike | Katty: Lines 10 and 11 you just added? |
| 17:44.38 | Katty | newp, i've not touched it yet. |
| 17:45.31 | Katty | not exactly sure what goes where. |
| 17:45.41 | TripleFFFF | thanks it worked |
| 17:45.48 | [TK]D-Fender | Katty>CunningPike: originally, i put woody on. <- that just sounded .... wrong ;) |
| 17:46.02 | CunningPike | Katty: You've run apt-get before, right? With sarge packages |
| 17:46.08 | Katty | [TK]D-Fender: oh you hush silly. |
| 17:46.14 | CunningPike | [TK]D-Fender: Cleanse your mind ;) |
| 17:46.14 | Katty | CunningPike: many many times. |
| 17:46.29 | Katty | CunningPike: i just pick an http and an ftp from the usa mirrors |
| 17:46.37 | Katty | CunningPike: and put them in the file, right? |
| 17:46.40 | [TK]D-Fender | CunningPike: Already well washed, rinsed, and conditioned :) |
| 17:46.52 | backblue | extensions in the database, i have to use "," or "|" in goto function? |
| 17:47.04 | CunningPike | Katty: So, if you make a new entry like 'deb <mirror of choice> sarge main contrib non-free' you should be good to mew |
| 17:47.53 | Katty | CunningPike: well i just replaced the ftp one with a different mirror |
| 17:47.56 | Katty | CunningPike: i'll see what happens |
| 17:51.20 | CunningPike | Katty: k |
| 17:52.08 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:52.08 | Katty | hmm, didn't like the http one i picked |
| 17:53.09 | Katty | i don't get it. |
| 17:53.23 | CunningPike | Katty: What did it say? |
| 17:53.46 | Qwell | backblue: | |
| 17:53.59 | Qwell | backblue: same goes for all apps |
| 17:54.07 | Katty | http://pastebin.ca/136435 <- CunningPike |
| 17:54.16 | Katty | CunningPike: that's the second http mirror i tried. |
| 17:54.47 | Katty | surely not both of them are mia |
| 17:57.20 | CunningPike | Katty: Unlikely - let me take a closer look |
| 17:57.59 | backblue | Qwell: all the apps in the database, i have to change "," to "|" ? |
| 17:58.16 | Qwell | backblue: I think | is the only one that works in realtime |
| 17:58.29 | Katty | CunningPike: i bet i typoed the source.list |
| 17:58.56 | CunningPike | Katty: What does it say now? |
| 17:59.26 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:59.36 | Katty | CunningPike: http://pastebin.ca/136445 |
| 17:59.41 | backblue | Qwell: ok, tks. |
| 18:01.30 | mmealling | hmm.....finally got inbound and outbound working..... |
| 18:01.46 | mmealling | but I can't seem to set caller id (using telasip) |
| 18:01.51 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:01.52 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:03.31 | CunningPike | Katty: Try this as your first line: 'deb http://debian.crosslink.net/debian/ sarge main contrib non-free' |
| 18:03.41 | CunningPike | Katty: Comment out the rest for now |
| 18:04.09 | *** join/#asterisk Vec (n=Vector@dsl-146-77-133.telkomadsl.co.za) |
| 18:04.48 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 18:04.48 | *** part/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 18:05.33 | Zodiacal | anyone know how i could have my dialplan dial an applicationmap command for me? |
| 18:05.49 | Katty | CunningPike: i commented out my two http lines and put that one in...left the ftp alone since it's fine |
| 18:05.56 | Katty | CunningPike: tryin it again |
| 18:06.00 | CunningPike | Katty: OK |
| 18:06.09 | CunningPike | Katty: paws crossed....... |
| 18:06.13 | Katty | CunningPike: Yay! |
| 18:06.17 | myiagy | -- Executing BackGround("SIP/6100-081fef60", "urasounds/ura_menu") in new stack |
| 18:06.21 | CunningPike | Katty: Excellent |
| 18:06.21 | myiagy | but i don't hear anything |
| 18:06.22 | Katty | guess that first line in my source.list was just.... |
| 18:06.30 | CunningPike | Katty: Kitty litter |
| 18:06.34 | Katty | exactly. |
| 18:06.37 | myiagy | rtp debug i only see Got packages, and only one Sent package at the beginning |
| 18:06.44 | myiagy | any ideas why this happens? |
| 18:07.46 | Katty | CunningPike: much appreciated (= |
| 18:07.55 | CunningPike | Katty: Any time |
| 18:07.57 | myiagy | i don't even know what to search for at google or voip-info.. tried "background no sound", no luck though.. |
| 18:08.03 | Katty | now if vista would just stop crashing every 3 hours. |
| 18:08.09 | CunningPike | myiagy: Codec mismatch? |
| 18:08.20 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 18:08.22 | CunningPike | Katty: Can't help with that, I'm afraid |
| 18:08.38 | Katty | CunningPike: yeah. vista's just a giant memory leak. |
| 18:08.40 | myiagy | CunningPike i tried ulaw and gsm |
| 18:08.48 | myiagy | but a codec mismatch would print an error wouldn't it? |
| 18:08.54 | Katty | CunningPike: IE7 crashes about once an hour...and I loose my audio a couple times a day usually. |
| 18:08.57 | quid246 | If putting * into a production environment... is it better to go with the last stable release or bleeding-edge SVN? |
| 18:09.13 | Katty | CunningPike: excel 2007 beta2, has this issue with excel charts liking to disappear when you cut/paste them. |
| 18:09.18 | CunningPike | myiagy: I'd have thought so....... |
| 18:09.27 | Katty | CunningPike: and the widget bar thingy randomly disappears sometimes too |
| 18:09.28 | CunningPike | Katty: Why do you do this to yourself? |
| 18:09.35 | Katty | CunningPike: we sell microsoft stuff. |
| 18:09.41 | Katty | CunningPike: so the company wants me to learn vista. |
| 18:09.42 | CunningPike | quid246: Latest stable - that's why it's called stable |
| 18:09.48 | CunningPike | Katty: Ah |
| 18:09.54 | Katty | CunningPike: it's shiny. |
| 18:09.59 | myiagy | CunningPike thats my problem, it doesn't print any errors.. but i hear no audio |
| 18:10.00 | CunningPike | Katty: :D |
| 18:10.01 | Katty | CunningPike: resource hoggy :< |
| 18:10.16 | quid246 | Cunning: haha, yeah... I thought so... |
| 18:10.23 | Katty | i'm not so fond of the User Layout |
| 18:10.24 | myiagy | the file is ok too |
| 18:10.34 | Katty | or these insane security features |
| 18:10.49 | Katty | anytime you try to run an mmc snapin, or anything from 'run' vista freaks out |
| 18:10.56 | CunningPike | myiagy: What format is the file, and is it 8KHz mono? |
| 18:11.06 | myiagy | it's wav |
| 18:11.08 | myiagy | yes |
| 18:11.18 | myiagy | it plays normally on another asterisk server |
| 18:11.21 | CunningPike | Katty: I haven't seen it - I barely know Windows XP |
| 18:11.29 | Katty | CunningPike: want a screenshot? |
| 18:11.31 | myiagy | i'll try to force only one codec |
| 18:11.37 | CunningPike | Katty: No thanks :D |
| 18:11.39 | malverian | Has anyone had issues with Sprint and Altel numbers not showing up properly in caller ID when going through a 1-800 number that you have ringing to your PBX? |
| 18:11.40 | Assid | hell.. i wouldnt mind seeing a screenshot |
| 18:12.13 | CunningPike | Katty: From the sounds of it, it'll just be a big blue square :) |
| 18:12.13 | Assid | Katty: you in the beta? or... ? |
| 18:12.20 | CunningPike | myiagy: Good plan - I would suspect codec issues |
| 18:12.22 | Katty | Assid: beta 2 |
| 18:13.21 | myiagy | CunningPike forced it to use ulaw, still no audio :/ |
| 18:13.49 | CunningPike | myiagy: But is it a ulaw file? |
| 18:14.04 | myiagy | well, maybe i should force gsm |
| 18:14.08 | myiagy | and convert the file to gsm too |
| 18:14.10 | CunningPike | myiagy: How does the non-working server differ from the working, config wise? |
| 18:14.32 | CunningPike | myiagy: How did you prepare the wav? On Windows? I have never gotten a native Windows wav to work |
| 18:14.35 | myiagy | CunningPike well, should be the same, i made a .tar.gz with the server that works config file |
| 18:14.46 | myiagy | and put it on the other one |
| 18:14.49 | CunningPike | myiagy: Always had to stroke it with sox to get it working |
| 18:15.01 | myiagy | yes, on windows.. |
| 18:15.13 | CunningPike | myiagy: Hmm - I would stroke it with sox then |
| 18:15.14 | Katty | CunningPike: Assid: http://www.copi-rite.com/horrors.jpg |
| 18:15.29 | myiagy | i'll try that.. but what's bothering me is that it works on the other server.. |
| 18:15.48 | Qwell | Katty: eww |
| 18:15.49 | CunningPike | myiagy: Hmmm |
| 18:16.01 | Assid | not bad |
| 18:16.02 | CunningPike | Katty: Yikes - hideous |
| 18:16.06 | Assid | mac copy |
| 18:16.14 | Qwell | Katty: sting? |
| 18:16.14 | CunningPike | Assid: Not even close |
| 18:16.21 | Katty | the start menu hates me =< |
| 18:16.28 | Katty | Qwell: yesh. |
| 18:16.45 | Katty | Qwell: i've got 13 other gigs of music too :P |
| 18:16.55 | Qwell | link? |
| 18:17.02 | Katty | =< |
| 18:17.49 | myiagy | CunningPike i'm thinking i forgot libasound2 |
| 18:18.11 | CunningPike | myiagy: You need that? |
| 18:18.16 | myiagy | i guess |
| 18:18.23 | backblue | Qwell: if i enable rtcachefriends=yes, i should not have nat problems, right? just like reading from the filesystem, correct? |
| 18:19.18 | myiagy | or not, still doesn't work |
| 18:19.52 | Assid | what do you use for irc Katty |
| 18:20.20 | myiagy | sox ura_menu.wav -r 8000 -c 1 ura_menu.gsm |
| 18:20.26 | myiagy | rm ura_menu.wav |
| 18:20.27 | Qwell | Assid: VERSION Gaim IRC |
| 18:20.29 | myiagy | still not playing |
| 18:20.31 | myiagy | damn :/ |
| 18:21.11 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 18:23.42 | CunningPike | myiagy: Interesting |
| 18:23.42 | CunningPike | myiagy: Permissions? |
| 18:23.46 | myiagy | CunningPike for the audio file? |
| 18:23.47 | myiagy | 644 |
| 18:23.52 | myiagy | owner asterisk.asterisk |
| 18:24.02 | CunningPike | myiagy: Hmmm..... |
| 18:24.33 | myiagy | and i'm guessing if it was permission would print an error too |
| 18:24.52 | CunningPike | myiagy: It doesn't (ask me how I know) |
| 18:24.55 | CunningPike | :) |
| 18:25.14 | myiagy | anyways, i even ran asterisk as root and still won't play |
| 18:25.19 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 18:25.32 | CunningPike | myiagy: I'm out of ideas - sorry |
| 18:25.41 | Un1x | hey is there a way to ban a specific number or even have asterisk hang up the call after certain ammount of minutes? |
| 18:25.43 | myiagy | it's ok, thanks anyways |
| 18:25.47 | CunningPike | myiagy: Can you play other files? |
| 18:25.52 | myiagy | no |
| 18:25.53 | CunningPike | myiagy: demo-congrats etc |
| 18:25.55 | myiagy | not even asterisk default |
| 18:26.07 | CunningPike | myiagy: Sucks to be you - sorry I can't help more |
| 18:26.25 | myiagy | i'll let you know if i figure it out, thanks :) |
| 18:26.44 | CunningPike | myiagy: Ya - I'd be interested to know |
| 18:28.45 | *** join/#asterisk num000 (n=numerobi@e177183003.adsl.alicedsl.de) |
| 18:29.14 | Un1x | hey is there a way to ban a specific number or even have asterisk hang up the call after certain ammount of minutes? |
| 18:29.15 | Un1x | ? |
| 18:29.34 | CunningPike | Un1x: Yes - search for 'ex-girlfriend' |
| 18:29.41 | Un1x | :s |
| 18:29.54 | Un1x | not exgirlfreind lmao, i wanna block a number so my damn brother cant call out via it :) |
| 18:29.59 | Un1x | err not via |
| 18:30.01 | Un1x | it! |
| 18:30.09 | coppice | or "death to telemarketers" |
| 18:30.26 | Qwell | Un1x: yeah, there are "exgirlfriend" scripts/examples |
| 18:30.31 | Qwell | Un1x: does exactly what you want |
| 18:30.37 | CunningPike | Un1x: Sure: exten => 1234567,1,Congestion |
| 18:30.53 | Un1x | Qwell are you serious or you joking with me |
| 18:31.03 | num000 | hi CunningPike how you doing? |
| 18:31.03 | Qwell | You want to block certain numbers from calling you? |
| 18:31.07 | Un1x | no |
| 18:31.10 | Un1x | from me calling them lol |
| 18:31.13 | CunningPike | num000: Good thanks - you? |
| 18:31.32 | num000 | CunningPike well, i did sleep very well, i had a long night as you know ;) |
| 18:31.44 | CunningPike | Un1x: What I gave you is the * equivalent of > /dev/null |
| 18:31.51 | CunningPike | num000: ;) |
| 18:32.16 | num000 | CunningPike you may remember the discussion about the 800 number from ionix? it remaind for more than 2 hours that discussion this morning ;) |
| 18:32.26 | CunningPike | num000: lol |
| 18:32.33 | Un1x | lol |
| 18:32.38 | num000 | ;) |
| 18:32.51 | num000 | 10 or 11 digits ;) |
| 18:32.53 | num000 | cool |
| 18:33.36 | benjk | CunningPike, the true /dev/null equiv is to set PRI_CAUSE to 1 ("unallocated number") then execute Hangup() |
| 18:33.48 | benjk | only works on ISDN though |
| 18:35.27 | Ebola | <Un1x> hey is there a way to ban a specific number or even have asterisk hang up the call after certain ammount of minutes? <--- That reminds me, I want to know how I can ban outgoing to international calls to countries I haven't specified :P |
| 18:35.41 | Ebola | remove the first to |
| 18:35.42 | Ebola | hehe |
| 18:36.24 | Un1x | ebola |
| 18:36.28 | Un1x | yes how do i do that :) |
| 18:37.58 | CunningPike | Ebola: You have to make an extension for each country you want to call, and then send the rest to Congestion |
| 18:38.32 | Ebola | Ah cool, that's how what I thought |
| 18:38.33 | Ebola | Thanks |
| 18:38.53 | CunningPike | Ebola: It's a bit of a pain for a long list, but it's the only real way |
| 18:39.06 | Ebola | heh |
| 18:39.51 | Ebola | Yeah, considering I'll want to block all but 30 countries |
| 18:40.18 | CunningPike | Ebola: 30 lines then - put them in a separate file and #include it |
| 18:40.34 | Ebola | k |
| 18:41.44 | Un1x | cant i just do sometihn like this exten => _6193420265,1,Congestion |
| 18:41.48 | ionix | heheh dear good 800 numbers :) |
| 18:41.52 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 18:42.28 | vooduhal | Stupid question. Which variable holds the the calling device? ie. SIP/1234? |
| 18:42.50 | CunningPike | Un1x: Just what I told you - leave out the _ though |
| 18:43.13 | Un1x | ok |
| 18:43.28 | wunderkin | would anyone have any idea why a peer would not get loaded .. it works ok on one box.. but using the same config on a vps, broadvoice does not show in sip show peers, only if i move it to the bottom.. also on the vps when it tries to register with broadvoice, it says no such host sip.broadvoice.com even though it is listed in /etc/hosts.. and when i do a sip reload i get WARNING[21812]: acl.c:244 ast_get_ip_or_srv: Unable to lookup 'ôÿ"@ î |
| 18:43.30 | *** join/#asterisk awannabe (n=gti@ip24-251-149-32.ph.ph.cox.net) |
| 18:43.40 | CunningPike | vooduhal: None - you have to parse it out of another one - let me look up what we did....... |
| 18:43.52 | vooduhal | K. |
| 18:44.07 | *** join/#asterisk angom_w (n=angom@red-corp-200.79.148.126.telnor.net) |
| 18:44.16 | awannabe | hi, is their any articles out there on how to get zaptel drivers loaded on freebsd from source? |
| 18:45.26 | CunningPike | vooduhal: exten => s,6,SetVar(AgentChannel=${CHANNEL}) |
| 18:45.27 | CunningPike | exten => s,7,Cut(AgentChannel=AgentChannel,-,1) |
| 18:45.33 | *** join/#asterisk tamp4x (n=tampon@vonmail.vonworldwide.com) |
| 18:45.37 | Un1x | heheh that actualy worked |
| 18:45.44 | Un1x | it executed congestion :) |
| 18:45.48 | CunningPike | vooduhal: We had to cut it from ${CHANNEL} |
| 18:45.57 | CunningPike | Un1x: Of course it worked :) |
| 18:45.57 | Un1x | Cunningpike what if i wanted to limit the ammount of time to talk to 5 minutes? |
| 18:46.01 | Un1x | thanks alot man :) |
| 18:46.28 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:46.53 | CunningPike | Un1x: That's a good question....... you might have to experiment with a MeetMe or something....... unless there's a Dial() trick that I don't know |
| 18:47.00 | Un1x | -- Starting simple switch on 'Zap/1-1' |
| 18:47.00 | Un1x | -- Executing Congestion("Zap/1-1", "") in new stack |
| 18:47.03 | vooduhal | CunningPike, k. I'll parse that. :) |
| 18:47.14 | Un1x | ok thanks for all the help cunningpike :) |
| 18:47.18 | [TK]D-Fender | CunningPike: You really need to read up on 1.2 spec...... |
| 18:47.19 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 18:47.27 | CunningPike | [TK]D-Fender: Probably |
| 18:47.37 | CunningPike | [TK]D-Fender: You refering to my use of CUT? |
| 18:47.47 | [TK]D-Fender | CunningPike:and SetVar.... |
| 18:48.07 | CunningPike | [TK]D-Fender: Ya - pre 1.2 code that I haven't migrated yet - mea culpa |
| 18:48.40 | intralanman | Un1x: look at dial's L() option |
| 18:48.41 | CunningPike | [TK]D-Fender: If it's any consolation, post-1.2 stuff uses 1.2 syntax |
| 18:49.04 | CunningPike | Un1x: There ya go - I figured there'd be a Dial() trick somewhere |
| 18:49.15 | CunningPike | Dial() is a very underrated application |
| 18:49.21 | intralanman | very |
| 18:49.26 | Un1x | where can i read about that, or know the syntax? |
| 18:49.31 | CunningPike | ~wiki |
| 18:49.36 | intralanman | show application dial |
| 18:49.52 | intralanman | or L(x[y[x]]) |
| 18:49.57 | intralanman | x is the cutoff in ms |
| 18:50.06 | intralanman | the first x |
| 18:50.10 | CunningPike | ~dial |
| 18:50.14 | intralanman | lol |
| 18:50.17 | CunningPike | :D |
| 18:50.26 | intralanman | i tried that earlier |
| 18:50.52 | intralanman | L(x[y[z]]) |
| 18:50.57 | intralanman | there, that's better |
| 18:51.06 | CunningPike | [TK]D-Fender: Code from a queue I created post 1.2: |
| 18:51.09 | CunningPike | exten => s,n,Set(AgentChannel=${CHANNEL}) |
| 18:51.09 | CunningPike | exten => s,n,Set(AgentChannel=${CUT(AgentChannel,-,-2)}) |
| 18:51.18 | CunningPike | Like that better? :) |
| 18:51.35 | [TK]D-Fender | CunningPike: Much :) |
| 18:51.37 | Qwell | L(x[:y[:z]]) |
| 18:51.56 | intralanman | damn, yeah, those too |
| 18:52.01 | CunningPike | vooduhal: Did you see that? Same effect, but with 1.2 syntax - use that instead |
| 18:52.21 | [TK]D-Fender | Qwell: O(m[:g[:z}}) ! |
| 18:52.28 | Qwell | CunningPike: Why set that var first? |
| 18:52.33 | vooduhal | CunningPike, I was just reading the CUT page at voip-info too. :) |
| 18:52.40 | vooduhal | That will work perfectly. :) |
| 18:52.42 | Qwell | Set(AgentChannel=${CUT(CHANNEL,-,-2)}) |
| 18:53.00 | [TK]D-Fender | Qwell: Very true, just beat me to it :) |
| 18:53.04 | CunningPike | vooduhal: Only because I need that variable elsewhere |
| 18:53.20 | *** join/#asterisk arkonadev (n=arkonaj@65.203.186.131) |
| 18:53.28 | CunningPike | vooduhal: Actually - I have no idea lol |
| 18:53.37 | arkonadev | hey what events can i subscribe to in the manager API to see if a phone call gets picked up? |
| 18:55.41 | IOscanner | Anyone know of a good 24 port FXS channel bank for Asterisk. |
| 18:56.03 | IOscanner | I need to add 48 analog lines for faxes for our location. |
| 18:57.03 | IOscanner | Also, Does anyone know where I can get a SIP account for Australia? I need to be able to make and recieve calls. |
| 18:57.06 | benjk | ADTRAN |
| 18:57.35 | IOscanner | What about the Rhino units. I have seen those too. |
| 18:57.54 | benjk | sure they should work too |
| 18:57.55 | IOscanner | The both use T-1 card correct to connect to the server? |
| 18:58.00 | benjk | yes |
| 18:58.10 | IOscanner | Can you put two on a Dual T-1 card? |
| 18:58.21 | benjk | there are E1 channel banks too, but they are typically much more expensive |
| 18:58.22 | IOscanner | or can a dual card handle that many lines? |
| 18:58.55 | benjk | for 48 channels you need two T1 circuits, hence a dual port T1 card or two single port T1 cards |
| 19:00.10 | IOscanner | So I would plug the fax machines into the ADTRAN then connect both ADTRANS to the T-1 ports on the server and configure with ztcfg. |
| 19:00.35 | IOscanner | Then setup group and use as normal extensions? |
| 19:01.41 | benjk | yep |
| 19:02.07 | awannabe | or a Cisco AS5300 box :) |
| 19:02.24 | benjk | pricey |
| 19:02.28 | Un1x | can soemeone point me to that Dail () thing on VOip-inof.org |
| 19:02.28 | awannabe | yeah, sure is |
| 19:02.28 | Un1x | ? |
| 19:02.31 | Un1x | *info |
| 19:02.53 | benjk | why don't you just use the search box on Voip-Info.org |
| 19:03.09 | Nugget | maybe because he can't spell "Dial" |
| 19:03.20 | intralanman | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 19:03.27 | intralanman | lol |
| 19:03.35 | Dovid | any way that i can change the invite header from asterisk for specific calls ? |
| 19:04.04 | Un1x | i did found nothign |
| 19:07.56 | Dovid | anyone know anything about changing the SIP header ? |
| 19:08.57 | justinu|laptop | there's lots of headers in SIP |
| 19:10.14 | Un1x | intralanman: exten => 6193420264,1,Dail L(x[:y][:z]): |
| 19:10.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:10.32 | Un1x | would that be proper format i know what xyz stand for just wanting to use that as example... |
| 19:11.09 | Dovid | justinu|laptop: i need to change something in it. i dont fully understand what i have to do. can i show u in a pb ? |
| 19:11.30 | IOscanner | Anyone know a provider that has a SIP or IAX account for Australia? |
| 19:11.58 | justinu|laptop | go ahead |
| 19:12.07 | Dovid | justinu|laptop: this is what they sent me http://pastebin.ca/136532 |
| 19:12.16 | Dovid | its to change the CID for the outbound call |
| 19:12.33 | Dovid | IOscanner: to make calls there or for a DI ? |
| 19:12.37 | Dovid | DID* |
| 19:13.07 | justinu|laptop | i'm not sure how to add arbitrary headers to a sip packet, maybe someone else knows? |
| 19:13.12 | IOscanner | Correct both I need to to call and recieve calls |
| 19:13.23 | justinu|laptop | I remember how to query arbitrary sip headers |
| 19:13.28 | Dovid | IOscanner: is it do able ? |
| 19:13.43 | Qwell | justinu|laptop: addsipheader(), unless it's changed |
| 19:13.57 | Dovid | IOscanner: u dont need to use one provider. u can use one for inbound and another for out. see voxbone.com and didx.org |
| 19:14.00 | Qwell | I think there is a func now |
| 19:14.11 | justinu|laptop | Dovid: there ya go, look on the wiki for the func qwell posted |
| 19:14.12 | Dovid | Qwell: did u see my pb ? |
| 19:14.23 | Dovid | okies |
| 19:15.02 | Dovid | nothing under addsipheader |
| 19:15.11 | cybertrickle | anyone gotten fax detection to work ? |
| 19:15.21 | Dovid | Qwell: do u know if its under 1.2.10 ? |
| 19:15.24 | *** join/#asterisk treetoap (n=pbaker@nnat-gw.adeptra.com) |
| 19:15.28 | Qwell | SIPAddHeader |
| 19:16.41 | Un1x | exten => 6193420264,1,Dail L(x[:y][:z]): if ive read correctly is that how it's done and im supposed to change x to how much ever MS.. i want it to be... and so on? |
| 19:16.49 | treetoap | hello all, I'm attempting to figure out how to write something that will simulate ackcall with message options. IE an enduser will call into the system and wait on hold, when the that user is placed on hold it would kick off an escalation process to various people. When the "tech" person picks up I want them to be directed with options to press 1 to accept the call. Could someone point me in the right direction on how to do that |
| 19:19.01 | *** join/#asterisk c4t3l (n=c4t3l@72.54.108.105) |
| 19:19.55 | intralanman | Un1x: yup |
| 19:22.09 | Un1x | intralanman wich one is right? |
| 19:22.12 | Un1x | the last one i said ? |
| 19:22.30 | intralanman | yeah, that last one |
| 19:22.48 | intralanman | L(5000) will cut it at like 5 seconds |
| 19:23.02 | Un1x | what about the AUDIO alerts i dont quite understand how to turn them on, is it done via the .c file |
| 19:23.23 | justinu|laptop | chanvars |
| 19:23.29 | trelane_ | where does one download iaxyprov from these days? |
| 19:24.21 | mog | svn co http://svn.digium.com/svn/iaxyprov/trunk iaxyprov-trunk |
| 19:25.27 | Un1x | mog: where is app_dial.c |
| 19:25.59 | mog | svn co http://svn.digium.com/svn/asterisk/trunk/apps/app_dial.c |
| 19:26.26 | mog | err svn cat http://svn.digium.com/svn/asterisk/trunk/apps/app_dial.c > app_dial.c |
| 19:29.26 | Un1x | mog: so i have to compilwe that in order to use the Dail L() function? |
| 19:30.10 | mog | to use app_dial it must be loaded in your asterisk system it was distributed to you with it |
| 19:30.21 | mog | i dont see that you could do much with asterisk without app_dial loaded |
| 19:31.07 | Un1x | i did locate app_dail.c |
| 19:31.10 | Un1x | couldn't find it |
| 19:31.45 | mog | it should have been there |
| 19:32.17 | Un1x | heh nevermind it's loaded as the module itself... |
| 19:32.18 | Un1x | app_dial.so Dialing Application |
| 19:32.20 | *** join/#asterisk doolph (n=doolph@200.46.148.58) |
| 19:32.22 | doolph | hi |
| 19:32.26 | mog | yes |
| 19:32.29 | Un1x | :) |
| 19:32.34 | doolph | how do I record at g729 codec |
| 19:32.40 | *** join/#asterisk TrickFinlay2 (n=Trickste@71-10-242-220.dhcp.oxfr.ma.charter.com) |
| 19:33.15 | Corydon-w | Record(foo.g729) |
| 19:33.17 | Lyfe | mog: well, you could make a voicemail server... :P |
| 19:33.42 | Lyfe | and um.. an ivr that goes nowhere.. and.. |
| 19:34.03 | mog | yeah |
| 19:34.07 | mog | you could have stuff go in |
| 19:34.28 | Un1x | mog: where do i set the ON/OFF for this option |
| 19:34.28 | Un1x | # LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the caller. |
| 19:34.28 | Un1x | # LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee. |
| 19:34.30 | mog | there are probably people out there using it that way |
| 19:34.30 | Lyfe | right, i know, stop being a smartass, go back to work. :P |
| 19:34.34 | mog | ? |
| 19:34.34 | doolph | nice |
| 19:37.46 | Un1x | it's saying MS |
| 19:37.47 | Un1x | L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c) |
| 19:37.55 | Un1x | doesn't that mean miliseconds... |
| 19:38.37 | intralanman | i think you can Set() those in the dialplan |
| 19:39.01 | intralanman | and yes.... those are milliseconds |
| 19:39.09 | intralanman | L(5000) will cut it at like 5 seconds |
| 19:39.15 | Un1x | yea :/ |
| 19:39.21 | Un1x | how do i change it from MS to seconds? |
| 19:39.24 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 19:39.27 | intralanman | * 1000 |
| 19:39.38 | Un1x | pardon? |
| 19:39.48 | intralanman | what are you asking exactly? |
| 19:39.50 | justinu|laptop | lmao |
| 19:40.04 | Un1x | how can i change the default Milliseconds, to Seconds... |
| 19:40.40 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 19:40.52 | intralanman | what do you want to do exactly? |
| 19:41.04 | intralanman | you want to cut the call at 5 minutes? |
| 19:41.28 | Un1x | yes.. |
| 19:41.32 | justinu|laptop | sounds like he's really confused about it wanted the parameters in millis vs seconds |
| 19:41.36 | justinu|laptop | s/wanted/wanting |
| 19:41.37 | intralanman | 5 * 60 * 1000 |
| 19:42.16 | treetoap | is there a way to play an IVR to a user when someone joins a queue? |
| 19:42.18 | intralanman | 6000 x however many minutes you wanna drop the call at |
| 19:42.19 | Un1x | can't i just change it so then it calculates in Seconds, instead of Miliseconds.. |
| 19:42.29 | intralanman | sure |
| 19:42.53 | intralanman | you may need to rewrite some of the source code.... but that's why it's open-source ;) |
| 19:43.33 | treetoap | I think there is a way to do it with the dial plan, just need to figure it out |
| 19:43.49 | intralanman | Un1x: what are you using this for? if it's going to be static in a dialplan.... you only have to do it once |
| 19:44.00 | Un1x | yea it's static |
| 19:46.20 | Un1x | 30000 |
| 19:46.27 | Un1x | miliseconds is 5 minutes :p |
| 19:47.45 | Un1x | ,1,Dail L(30000[:6000][:3000]): there we go that will terminate call at 5 minutes give first warning at 1 minute and another warning at 30 seconds ;) |
| 19:54.50 | [TK]D-Fender | treetoap: : can you elaborate about this IVR..... What would it do? You can already use 1-touch DTMF to EXIT a queue to go somewhere else. Does this not serve whatever it is you're thinking of doing? |
| 19:55.06 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 19:57.07 | *** join/#asterisk backblue (n=moo@87-196-98-179.net.novis.pt) |
| 19:58.25 | treetoap | ok then, well how about call screening - can someone point me in the direction on how to do that? |
| 20:03.16 | toerkeium | guys, if my SIP provider doesn't give me username and password because they allow by IP address, how my register string will looks like? |
| 20:04.56 | intralanman | toerkeium: you do have a static ip address, right? |
| 20:05.46 | toerkeium | intralanman, yes I have (my * box) and my SIP provider |
| 20:07.31 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:08.51 | benjk | you won't need to register if they use your IP address to authenticate you |
| 20:08.57 | charles___ | toerkeium: just set a SIP type=peer |
| 20:09.41 | toerkeium | I have set the type=peer for outbound calls, I don't need to do anything else then? |
| 20:09.42 | [TK]D-Fender | treetoap: There are WIKI pages showing that, go check them out. |
| 20:11.33 | Un1x | intralanman: help please.. |
| 20:11.34 | Un1x | Aug 17 16:12:00 WARNING[30340]: pbx.c:1700 pbx_extension_helper: No application 'Dail L' for extension (default, 9058039446, 1) |
| 20:11.34 | Un1x | <PROTECTED> |
| 20:11.34 | Un1x | <PROTECTED> |
| 20:13.21 | *** join/#asterisk loconut (n=blt@webtrotter.com) |
| 20:13.23 | loconut | hello |
| 20:14.00 | loconut | We have a bunch of Sip phones off-site that are experiencing horrible jitter problems. Will a) the jitterbuffer patch b) some sort of asterisk-sip proxy placed on-site help? |
| 20:14.08 | intralanman | Un1x: you're killin me dude |
| 20:14.10 | intralanman | lol |
| 20:14.26 | intralanman | dial.... not dail |
| 20:14.28 | loconut | ideally we'd have an asterisk box on-site that the sip-phones go through and the calls are then routed over IAX on a TCP connection |
| 20:14.29 | intralanman | let's start there |
| 20:14.47 | loconut | even UDP would be ok, if it were routed over iax |
| 20:15.36 | intralanman | you'll want something like dial(sip/peer/number|L(30000)) |
| 20:15.44 | intralanman | something like taht |
| 20:15.46 | intralanman | that |
| 20:16.37 | justinu|laptop | intralanman: i admire your patience |
| 20:16.59 | Un1x | intralanman: i'm sorry but my extensions.conf has it correctly.. according to me im not sure if it's right but this is what i have |
| 20:16.59 | Un1x | exten => 9058039446,1,Dail L(30000[:6000][:3000]): |
| 20:17.35 | benjk | Dail L? |
| 20:17.47 | Un1x | benjk it's to limit calls |
| 20:17.53 | twisted[asteria] | huminah |
| 20:17.54 | benjk | did you type this (typo?) or paste it |
| 20:17.57 | *** join/#asterisk num000 (n=numerobi@e177183003.adsl.alicedsl.de) |
| 20:18.00 | loconut | why not just make it Kal-El |
| 20:18.07 | Un1x | benjk just paste it |
| 20:18.12 | benjk | then its wrong |
| 20:18.16 | Un1x | whats wrong should the L be lower case or soemthing.. |
| 20:18.20 | Un1x | whats' wrong exactly benjk? |
| 20:18.27 | benjk | first of all tis Dial, not Dail |
| 20:18.28 | Qwell | remove the [] |
| 20:18.36 | twisted[asteria] | Un1x, it's DIAL, not DAIL |
| 20:18.53 | twisted[asteria] | heh. |
| 20:18.55 | Un1x | Qwell remove [] but thats how it shos on voip-info |
| 20:19.02 | justinu|laptop | it's also Dial(SIP/thatguy,30,L(30000:60000:3000)); |
| 20:19.04 | twisted[asteria] | yeah, and the []'s are not valid. |
| 20:19.10 | Qwell | okay, well I'm telling you to remove them |
| 20:19.15 | twisted[asteria] | those simply mean optional args. |
| 20:19.21 | benjk | and once you corrected that typo we can look at the args |
| 20:19.25 | num000 | CunningPike i still have ReliablyTransmitting (NAT) \ SIP/2.0 404 Not Found error if I do call the echo-test. How can I find out what it is loooking for which it does not find. |
| 20:19.33 | Un1x | ok |
| 20:19.50 | tamp4x | im running debian, and gettign this error, Aug 17 16:18:46 WARNING[9066]: chan_zap.c:915 zt_open: Unable to open '/dev/zap/ channel': No such device or address |
| 20:19.58 | tamp4x | any ideas why? |
| 20:20.00 | Un1x | ok removed them |
| 20:20.01 | Un1x | (30000:6000:3000): |
| 20:20.02 | toerkeium | guys, is there any important risk on allowing anon sip calls? |
| 20:20.09 | twisted[asteria] | bahahaha |
| 20:20.29 | twisted[asteria] | do you let just anyone walk into your house? |
| 20:20.37 | twisted[asteria] | and use your phone? |
| 20:20.45 | toerkeium | they can use my trunks? |
| 20:20.50 | toerkeium | or just call my users? |
| 20:20.58 | twisted[asteria] | depends on what context you have in general |
| 20:21.36 | toerkeium | because if I enable anon sip calls, my incoming calls get in, but if I set it off, I can't receive incoming calls.. |
| 20:22.07 | Un1x | justinu|laptop: so it's like this exten => 9058039446,1,Dial(SIP/thatguy,30,L(30000:60000:3000)); |
| 20:22.26 | nestar | tamp4x: unload chan_zap.so in modules.conf |
| 20:22.27 | twisted[asteria] | toerkeium, then you need user/peer entries for each of your incoming hosts |
| 20:22.42 | intralanman | isn't that what i said like a 1/2 hour ago?!?!? |
| 20:22.45 | justinu|laptop | uni1x: that should work better for you, but obviously SIP/thatguy isn't correct for your configuration |
| 20:22.56 | toerkeium | twisted[asteria]: how would be that? could you give me an example please? |
| 20:23.00 | benjk | thatguy has to be a valid sip peer or a sip uri, if you checked that, it should be ok |
| 20:23.00 | justinu|laptop | and 30 may not be the right timeout value |
| 20:23.04 | twisted[asteria] | toerkeium, look at sip.conf.example |
| 20:23.25 | toerkeium | twisted[asteria]: where exactly? I get confused of what I need to setup |
| 20:23.55 | twisted[asteria] | in the section where users/peers are defined |
| 20:24.03 | twisted[asteria] | that's the best i can offer, gotta get back to work |
| 20:24.09 | toerkeium | thank you |
| 20:24.42 | Un1x | i see thanks |
| 20:25.16 | Un1x | justinu|laptop: you sure about the end brackets after 3000 u got 3 brackets there shouldn't it be 1? |
| 20:27.15 | justinu|laptop | those are called parenthesis, and they must balance |
| 20:28.15 | Un1x | justinu|laptop: now i get this |
| 20:28.16 | Un1x | http://pastebin.ca/136652 |
| 20:28.30 | tamp4x | Anyone know how to correct this error?: Aug 17 16:18:46 WARNING[9066]: chan_zap.c:915 zt_open: Unable to open '/dev/zap/channel': No such device or address |
| 20:28.54 | tamp4x | in debian |
| 20:28.54 | *** join/#asterisk deb_user (n=Hypnotis@70-59-108-105.albq.qwest.net) |
| 20:28.55 | justinu|laptop | Un1x: looks like you finally got the syntax right |
| 20:29.06 | deb_user | i'm getting a lot of strange sounding distortion on my zaptel channels |
| 20:29.17 | deb_user | using a tdm400 22b |
| 20:29.29 | Un1x | justinu|laptop the problem is the call wont go through see the error... in that pastebin |
| 20:29.38 | deb_user | anybody know of any way to reduce this noise via the config files? |
| 20:29.54 | deb_user | or at least give me some tips on troubleshooting it maybe? |
| 20:30.48 | viler | does anyone know how to make h323 to sip calls ? the system shows to me "cleared, reason 1" thanks |
| 20:31.04 | *** join/#asterisk Ebola (n=Ebola@user-54458db0.lns1-c13.telh.dsl.pol.co.uk) |
| 20:31.34 | Un1x | justinu|laptop: please... http://pastebin.ca/136652 |
| 20:32.07 | *** join/#asterisk RoyK (n=roy@ti211310a080-3288.bb.online.no) |
| 20:33.40 | CunningPike | tamp4x: Yes - load the correct zaptel module for your hardware |
| 20:34.43 | CunningPike | Un1x: What is your Dial command |
| 20:35.21 | deb_user | i'm getting a lot of strange sounding distortion on my zaptel channels |
| 20:35.24 | deb_user | using a tdm400 22b |
| 20:35.26 | deb_user | anybody know of any way to reduce this noise via the config files? |
| 20:35.40 | CunningPike | deb_user: Maybe play around with the rxgain and txgain settings |
| 20:35.51 | Un1x | exten => 9058039446,1,Dial(SIP/splitinfinity,30,L(30000:60000:3000)); |
| 20:35.55 | Un1x | CunningPike: there |
| 20:36.12 | toerkeium | when setting up a peer for incoming calls, what should I put between the [] ? IP address? my sip provider don't let me register with the register string |
| 20:36.19 | intralanman | Un1x: your warn time is more than the calltime |
| 20:36.22 | deb_user | pike: sometimes i wonder if its not interference from other lines...or moving through the internet |
| 20:36.34 | deb_user | pike: know of anyway to diagnose this type of thing? |
| 20:36.38 | CunningPike | Un1x: So why is a Zap channel involved? |
| 20:36.41 | *** join/#asterisk topping (n=topping@207.47.6.207.static.nextweb.net) |
| 20:36.52 | intralanman | Un1x: and you want it to warn you every 3 seconds? |
| 20:37.00 | Un1x | no |
| 20:37.05 | Un1x | every 30 seconds |
| 20:37.12 | Un1x | when the last 1 minute is remaining |
| 20:37.16 | intralanman | then bump that last number one more 0 |
| 20:37.29 | intralanman | and probably the first number too |
| 20:37.31 | CunningPike | deb_user: "Moving through the internet" will not affect Zap channels - try ztmonitor |
| 20:37.43 | intralanman | right now you're limiting the call to 30 seconds |
| 20:37.44 | Un1x | ok |
| 20:37.56 | intralanman | and playing the first warning at 60 seconds LOL |
| 20:38.00 | Un1x | so now it's 30,L(300000:600000:30000)); |
| 20:38.00 | CunningPike | deb_user: Or fxotune, with your telcos test tone number |
| 20:38.17 | deb_user | fxotune? |
| 20:38.17 | deb_user | never heard of that one... |
| 20:38.19 | Un1x | intralanman i want it to play the firstwarning, at 60 seconds left.. |
| 20:38.21 | intralanman | Un1x: that should be closer |
| 20:38.22 | Un1x | and then 30 seconds left |
| 20:38.29 | intralanman | yeah, that's cool |
| 20:38.29 | *** join/#asterisk sebatk (n=sebatk@r200-40-61-230.ae-static.anteldata.net.uy) |
| 20:38.45 | intralanman | but if you limit the call to 30 seconds.... you won't ever have 60 seconds left |
| 20:38.48 | intralanman | ;) |
| 20:39.05 | intralanman | so now the times should be right |
| 20:39.39 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 20:39.45 | Un1x | intralanman: http://pastebin.ca/136662 |
| 20:40.10 | Un1x | exten => 9058039446,1,Dial(SIP/splitinfinity,30,L(300000:600000:30000)); |
| 20:40.39 | Un1x | but it doesn't even call lmao! |
| 20:40.40 | Un1x | :/ |
| 20:42.23 | intralanman | did you ever have this working before you tried to limit it? |
| 20:42.46 | Un1x | as int he calls yes |
| 20:42.49 | *** join/#asterisk Lyfe (n=lyfe@69.8.146.78) |
| 20:42.54 | Un1x | it's only this number the limit crap is stopping it |
| 20:43.31 | Un1x | yep works everywhere else just not that number |
| 20:45.10 | intralanman | this is a SIP channel you're trying to dial out? or a Zap? |
| 20:46.11 | deb_user | umm... |
| 20:46.15 | deb_user | how do you use ztmonitor? |
| 20:46.21 | deb_user | does it have to be installed first? |
| 20:47.06 | CunningPike | deb_user: http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment |
| 20:47.31 | deb_user | thanks |
| 20:47.33 | *** join/#asterisk Eonz (n=Icarus@irc.americatelnet.com.pe) |
| 20:47.35 | CunningPike | intralanman: His CLI pastebin shows a Zap channel, so I'm not sure what's going on |
| 20:47.59 | intralanman | CunningPike: that's what i'm getting to.... doesn't make sense to me |
| 20:48.05 | vooduhal | Are there any variables that get set on a call attempt from Queue? |
| 20:48.06 | Un1x | intralanman i havemy analogue phone plugged into a TDM 400P 22b card ;) |
| 20:48.11 | Un1x | but i dont use th FXO ports only FXS |
| 20:48.16 | vooduhal | Anyway to determine if it was a queue calling or just a normal call? |
| 20:48.17 | Un1x | and all my calls are via SIP |
| 20:48.52 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 20:48.58 | WildPikachu | how would i specify my username (which is an email address) in my dial ... Dial(IAX2/my.email@address@PROVIDER/${EXTEN:1}) that doesn't work :( |
| 20:49.06 | CunningPike | vooduhal: If your queue is in its own context, you should be able to use ${CONTEXT} |
| 20:49.45 | CunningPike | Un1x: Paste your CLI now that you have the correct Dial() syntax |
| 20:50.46 | intralanman | WildPikachu: you're sure it's the entire address? |
| 20:50.51 | Un1x | Cunningpike i just did |
| 20:50.51 | Un1x | exten => 9058039446,1,Dial(SIP/splitinfinity,30,L(300000:600000:30000)); |
| 20:50.56 | intralanman | not just the name part? |
| 20:50.56 | Un1x | thats my syntax |
| 20:50.57 | CunningPike | WildPikachu: Try assigning your user name to a variable and then substituting that - still might not work, because 2 @ is not a valid address |
| 20:50.58 | WildPikachu | i was wrong :) |
| 20:51.01 | Un1x | cli = http://pastebin.ca/136662 |
| 20:51.16 | sebatk | I'm having an error that I cant find any solution: NMI dazzed and confused anyone can help me???? |
| 20:51.16 | WildPikachu | sorry |
| 20:51.17 | CunningPike | Un1x: Forget the ; at the end - what is this? javascript? |
| 20:51.24 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 20:51.24 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 20:51.35 | CunningPike | sebatk: How many of them? |
| 20:51.42 | intralanman | the ; doesn't hurt anything |
| 20:51.43 | intralanman | lol |
| 20:51.51 | Un1x | lmao |
| 20:51.53 | benjk | the semicolon at the end will be ignored as its an empty comment |
| 20:51.58 | Un1x | they pasted it to me like that |
| 20:52.51 | Un1x | ok Cunningpike with that semicolon gone it's this |
| 20:52.52 | Un1x | http://pastebin.ca/136677 |
| 20:53.02 | intralanman | i might even be the one that pasted it to him.... too much php/perl/and javascript lol |
| 20:53.22 | justinu|laptop | it was me, i write code, so semicolons just seem natural |
| 20:54.21 | Un1x | ok but why is it saying it's busy when the phone aint busy!!!! |
| 20:54.21 | Un1x | :( |
| 20:54.40 | sebatk | <PROTECTED> |
| 20:54.43 | CunningPike | Un1x: Did you pastebin your CLI output? |
| 20:54.53 | Un1x | http://pastebin.ca/136677 |
| 20:54.54 | CunningPike | sebatk: How many of them? |
| 20:54.54 | intralanman | ummm.... Un1x do you have a number you're calling on there? |
| 20:55.03 | Un1x | exten => 9058039446,1,Dial(SIP/splitinfinity,30,L(300000:600000:30000)); |
| 20:55.07 | Un1x | err |
| 20:55.12 | Un1x | without the colon on my server tho :) |
| 20:55.20 | sebatk | what do you mean with how many of them?? |
| 20:55.23 | *** join/#asterisk kupesoft (n=dave@CPE000c418c08cf-CM0013718cb08a.cpe.net.cable.rogers.com) |
| 20:55.44 | CunningPike | Un1x: Does 'sip show peers' show a peer called splitinfinity? |
| 20:55.46 | intralanman | Un1x: splitinfinity is a peer, right? |
| 20:55.50 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:55.51 | Un1x | YES! |
| 20:55.57 | CunningPike | sebatk: How many errors ? |
| 20:56.07 | Un1x | Name/username Host Dyn Nat ACL Port Status |
| 20:56.08 | Un1x | didww-in 212.150.36.116 5060 Unmonitored |
| 20:56.08 | Un1x | splitinfinity/10085 38.96.4.15 5060 Unmonitored |
| 20:56.08 | Un1x | 2 sip peers [2 online , 0 offline] |
| 20:56.21 | intralanman | Un1x: what number are you trying to call? |
| 20:56.22 | *** join/#asterisk c4t3l (n=c4t3l@72.54.108.105) |
| 20:56.32 | Un1x | 9058039446 |
| 20:56.38 | Un1x | the one i pasted, with the Syntax |
| 20:56.43 | Un1x | exten => 9058039446,1,Dial(SIP/splitinfinity,30,L(300000:600000:30000)) |
| 20:57.02 | intralanman | you need a "/9058039446" after the peer name |
| 20:57.16 | Un1x | so like this |
| 20:57.22 | Un1x | exten => 9058039446,1,Dial(SIP/splitinfinity/9058039446,30,L(300000:600000:30000)) |
| 20:57.23 | Un1x | ? |
| 20:57.30 | CunningPike | Is it a full moon? |
| 20:57.31 | intralanman | that's definitely closer |
| 20:57.37 | intralanman | CunningPike: i think it is |
| 20:58.15 | Un1x | see i have this as my syntax from befiore |
| 20:58.16 | Un1x | exten => _X.,1,Dial(${splitinfinity}/${EXTEN}) |
| 20:58.27 | Un1x | and it is working ok to make calls :) |
| 20:58.49 | Un1x | but that syntax up there aint working |
| 20:59.24 | justinu|laptop | we need D-Fender to sort this guy out |
| 20:59.27 | justinu|laptop | he speaks n00b well |
| 20:59.35 | intralanman | heheh |
| 20:59.54 | Un1x | ahh, com'on as i said i am new to asterisk :) |
| 20:59.56 | CunningPike | Un1x: So, try 'exten => 9058039446,1,Dial(${splitinfinity}/9058039446,30,L(300000:600000:30000)) |
| 20:59.56 | justinu|laptop | first of all, ${splitinfinity} is a variable |
| 21:00.11 | [TK]D-Fender | justinu : It COULD be a CONSTANT....... |
| 21:00.23 | justinu|laptop | so it gets substituted with whatever its set to when it gets evaluated |
| 21:00.25 | [TK]D-Fender | Un1x : Show us where you define it |
| 21:01.01 | justinu|laptop | CunningPike: good idea, and replace 905... with ${EXTEN} |
| 21:01.34 | sebatk | I get the error many times |
| 21:01.41 | sebatk | and it hangs the server |
| 21:01.54 | Un1x | heh that worked cunningpike... |
| 21:02.10 | CunningPike | Un1x: So, try 'exten => 9058039446,1,Dial(${splitinfinity}/${EXTEN},30,L(300000:600000:30000)), as justinu|laptop suggested |
| 21:02.40 | CunningPike | Un1x: If you had a working entry, it really shouldn't have taken you 2 hours to add the L option to it.......... |
| 21:02.40 | Un1x | it's working without the $exten.. |
| 21:03.19 | Un1x | i didn't realize it would work till just now sorry.. but now i gotta figure out the time in miliseconds coz this thing starts tellingme every 30 seconds how much time i have left.. |
| 21:04.03 | intralanman | minutes x 60000 |
| 21:04.18 | CunningPike | Un1x: Provided you can type, it will work with the ${EXTEN}, too. And the numbers need to be 300000:60000:30000 |
| 21:04.33 | CunningPike | You have too many zeros |
| 21:04.46 | sebatk | CunningPike: can you help me?? |
| 21:04.48 | Un1x | the one you just said Cunningpikwe `need to be 300000:60000:30000` |
| 21:05.12 | Un1x | is it going to tell me starting at 1 minute... |
| 21:05.22 | CunningPike | Un1x: YES!!! |
| 21:05.25 | Un1x | kk |
| 21:05.42 | CunningPike | Un1x: You will find that things work a whole lot better when you enter them correctly :D |
| 21:06.26 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 21:06.29 | CunningPike | sebatk: You have some hardware issue - the reason I asked how many, is that we always get one at boot and it doesn't seem to cause any problems. Many causing a crash is bad |
| 21:06.52 | CunningPike | sebatk: What card is it? |
| 21:06.55 | sebatk | is not at boot |
| 21:07.22 | *** join/#asterisk redondos (n=redondos@190.48.27.147) |
| 21:07.29 | sebatk | TE411P |
| 21:07.31 | sebatk | 2 |
| 21:07.34 | CunningPike | sebatk: What does 'cat /proc/interrupts' say? |
| 21:07.40 | CunningPike | sebatk: What server/OS? |
| 21:07.45 | redondos | Is it OK to register as a SIP client in sip.conf *and* in extensions.conf? |
| 21:07.48 | sebatk | CentOS |
| 21:08.00 | sebatk | the proc interrupts seems fine |
| 21:08.08 | redondos | The wiki says to do it in sip.conf, but I have a "register =>" line in extensions.conf and it just works fine. |
| 21:08.15 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-4.getnet.net) |
| 21:08.25 | sebatk | server : HP Proliant DL 380 G3 |
| 21:08.32 | sebatk | Asterisk 2.1.4 |
| 21:08.34 | redondos | Of course the SIP user is configured in sip.conf, but no register line there. |
| 21:08.35 | num000 | does anyone know why asterisk claims even though the file exists? the failure it gives is ast_streamfile: Unable to open demo-echotest (format alaw): No such file or directory |
| 21:08.42 | Assid | 2.1.4???? |
| 21:08.52 | sebatk | 1.2.4 |
| 21:09.02 | sebatk | sorry, bad keying |
| 21:09.08 | CunningPike | redondos: They do 2 different things - sip.conf controls access to your server - who can register, and what as. Extensions.conf then defines what happens when someone dials that SIP registration |
| 21:09.23 | CunningPike | num000: Permissions? :D |
| 21:09.30 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 21:09.32 | *** join/#asterisk Skyelar (n=planet@222-153-145-60.jetstream.xtra.co.nz) |
| 21:09.37 | num000 | CunningPike I set them to 777 should be ok ? ;) |
| 21:09.55 | redondos | CunningPike: So why would I want to have a register line in sip.conf for example? |
| 21:10.04 | CunningPike | sebatk: We have a TE410P in a DL360 on RHEL, so we're almost the same. Did you say you had 2 cards? |
| 21:10.06 | sebatk | CunningPike: any idea?? could be solved in any new driver of the digium boards?? |
| 21:10.19 | sebatk | yes 2 cards |
| 21:10.31 | quid246 | Hmm... I just built 1.2.10 stable on a Dual-Opteron box... with no calls, it's eating up almost 100% of CPU time... anybody seen this recently? |
| 21:10.38 | CunningPike | redondos: If you want to register asterisk as a UA for a SIP trunk provider |
| 21:10.57 | sebatk | I have disbale hyperthrading someone told me this could be the problem |
| 21:11.08 | CunningPike | sebatk: What does 'cat /proc/interrupts' say? |
| 21:11.20 | sebatk | I tell you in a minute |
| 21:11.38 | CunningPike | sebatk: Tell me now, dammit ;) |
| 21:11.43 | num000 | what is asterisk here doing: Unable to find a path from gsm to alaw |
| 21:11.53 | [TK]D-Fender | redondos : You don't do "register" in extensions.conf. thats not where it belongs and will have no net effect |
| 21:11.54 | num000 | what does gsm to alaw mean? |
| 21:11.58 | redondos | CunningPike: So if I wanted to configure extension 1234 for incoming and outgoing SIP connections I would have to have a "register" line in both extensions.conf and sip.conf? (along with a [1234] definition in sip.conf) |
| 21:12.13 | redondos | [TK]D-Fender: Ok, so that line in extensions.conf is superfluous? It's there, and it works. |
| 21:12.28 | [TK]D-Fender | redondos : No, its there and its IRRELEVENT. |
| 21:12.31 | CunningPike | redondos: No - register is to register with another provider's server |
| 21:12.42 | sebatk | <PROTECTED> |
| 21:12.43 | sebatk | <PROTECTED> |
| 21:12.43 | sebatk | <PROTECTED> |
| 21:12.43 | sebatk | <PROTECTED> |
| 21:12.43 | sebatk | <PROTECTED> |
| 21:12.43 | sebatk | <PROTECTED> |
| 21:12.45 | sebatk | <PROTECTED> |
| 21:12.45 | redondos | I see. |
| 21:12.47 | sebatk | 177: 14523026 4229 IO-APIC-level eth0 |
| 21:12.47 | RoyK | ~pb |
| 21:12.49 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 21:12.49 | sebatk | 185: 121 73076 IO-APIC-level eth1 |
| 21:12.51 | sebatk | 193: 22478 318914 IO-APIC-level cciss0 |
| 21:12.53 | sebatk | 201: 2756682 4544165 IO-APIC-level wct4xxp |
| 21:12.54 | syzygyBSD | and the spammer award goes to... |
| 21:12.55 | sebatk | 209: 4544225 2756601 IO-APIC-level wct4xxp |
| 21:12.56 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 21:12.57 | sebatk | NMI: 1 0 |
| 21:12.59 | sebatk | LOC: 7331265 7330704 |
| 21:13.01 | sebatk | ERR: 0 |
| 21:13.04 | sebatk | MIS: 0 |
| 21:13.05 | quid246 | Dang... got disconntected right after my Q. |
| 21:13.05 | eKo1 | ack! |
| 21:13.06 | [TK]D-Fender | sebatk : do NOT do that again. Please use a pastebin |
| 21:13.07 | [TK]D-Fender | ~pb |
| 21:13.09 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 21:13.13 | sebatk | there is my proc interrupts |
| 21:13.52 | CunningPike | sebatk: OK - please pastebin that, along with lspci -vb |
| 21:15.05 | sebatk | <PROTECTED> |
| 21:15.05 | sebatk | <PROTECTED> |
| 21:15.05 | sebatk | <PROTECTED> |
| 21:15.06 | sebatk | <PROTECTED> |
| 21:15.06 | sebatk | <PROTECTED> |
| 21:15.06 | sebatk | <PROTECTED> |
| 21:15.08 | sebatk | <PROTECTED> |
| 21:15.10 | sebatk | 177: 14523026 4229 IO-APIC-level eth0 |
| 21:15.12 | sebatk | 185: 121 73076 IO-APIC-level eth1 |
| 21:15.14 | sebatk | 193: 22478 318914 IO-APIC-level cciss0 |
| 21:15.15 | quid246 | ~pb |
| 21:15.16 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 21:15.16 | sebatk | 201: 2756682 4544165 IO-APIC-level wct4xxp |
| 21:15.18 | sebatk | 209: 4544225 2756601 IO-APIC-level wct4xxp |
| 21:15.20 | sebatk | NMI: 1 0 |
| 21:15.22 | sebatk | LOC: 7331265 7330704 |
| 21:15.24 | sebatk | ERR: 0 |
| 21:15.26 | sebatk | MIS: 0 |
| 21:15.26 | redondos | I am getting a "Bad auth" error when registering now. I put the register line in sip.conf where it belongs. |
| 21:15.28 | sebatk | wait |
| 21:15.42 | redondos | That does just mean invalid password or something? It works OK for making outgoing calls. |
| 21:15.44 | CunningPike | sebatk: What part of pastebin do you not understand? |
| 21:15.45 | quid246 | sebatk: Use pastebin |
| 21:16.04 | CunningPike | redondos: Who are you registering with? |
| 21:16.10 | quid246 | you want help... help us! |
| 21:16.37 | redondos | CunningPike: Voxee |
| 21:17.04 | CunningPike | redondos: Likely a faulty set of credentials |
| 21:17.55 | CunningPike | sebatk: We are using nofb acpi=off noht nousb, and have disabled hyperthreading and USB in the BIOS |
| 21:18.06 | redondos | CunningPike: Is it possible that it works for outgoing calls but not for registering as a sip peer? |
| 21:18.13 | Skyelar | Anyone around that would like to volunteer to be a "Backport to 1.2.10 of Manager eventq producer-consumer system" guinea pig? (requested quite a bit as a possible solution for bug #6626) |
| 21:19.23 | *** join/#asterisk RoyK (n=roy@ti211310a080-3288.bb.online.no) |
| 21:19.32 | sebatk | CunningPike: I have a txt with the info |
| 21:19.41 | sebatk | how can i give it to you? |
| 21:19.56 | [TK]D-Fender | sebatk : PASTEBIN! |
| 21:19.57 | [TK]D-Fender | ~pb |
| 21:19.59 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 21:19.59 | CunningPike | sebatk: 1, 2, 3........ PASTEBINNNNNNNNNNNNNNNN |
| 21:21.32 | CunningPike | Definitely a full moon |
| 21:22.16 | justinu|laptop | is that why all the wackos are here? |
| 21:22.32 | CunningPike | Must be |
| 21:22.40 | vader-- | tkd have you ever seen this problem |
| 21:22.49 | vader-- | i have a problem with the asterisk voicemail where messages are getting deleted but only the wav file is being delete and not the txt file? |
| 21:22.53 | redondos | What's the CLI command for dialing? |
| 21:22.57 | vader-- | and it doesn't happen all the time |
| 21:23.16 | [TK]D-Fender | vader-- : not to my awareness. Is it consistant or random? |
| 21:23.18 | CunningPike | vader--: List didn't show up anything, then |
| 21:23.23 | sebatk | I'm sorry but my language is from sweden so its not easy for me understand every thing |
| 21:23.24 | vader-- | cunning na |
| 21:23.32 | vader-- | it's been random |
| 21:23.37 | vader-- | like when i test it it's fine |
| 21:23.41 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 21:23.45 | vader-- | but people complain about not being able to get rid of old messages |
| 21:23.53 | file | yo yo JunK-Y, you're hurting me |
| 21:23.55 | CunningPike | sebatk: OK, but when half a dozen people tell you to use pastebin and send the links........... |
| 21:24.04 | JunK-Y | yo yo file! |
| 21:24.09 | sebatk | yes Im on it |
| 21:24.18 | Skyelar | vader--: what vewrsion of Asterisk? There were a bunch of fixes for this in 1.2.10 |
| 21:24.20 | Un1x | file is there a way to record a call :p? |
| 21:24.40 | file | Un1x: of course, MixMonitor or Monitor |
| 21:24.41 | [TK]D-Fender | vader-- : What is accessing VM? Are they being e-mailed out and then deleted? Is there something that could be locking that file? |
| 21:25.04 | sebatk | <PROTECTED> |
| 21:25.28 | vader-- | 1.2..1 |
| 21:25.31 | vader-- | 1.2.7.1 |
| 21:26.06 | vader-- | what is happening is people are getting the voice saying you have 1 old voicemail |
| 21:26.09 | vader-- | so they go to check it |
| 21:26.17 | vader-- | when it tries to play there is no wave file |
| 21:26.17 | Skyelar | vader--: ... and there's nothing there but the .txt file |
| 21:26.23 | vader-- | so it boots them out of the voicemail |
| 21:26.39 | Skyelar | vader--: upgrading to 1.2.10 should fix it |
| 21:26.41 | vader-- | i go into the Old directory for voicemail and all that is in there is msg0000.txt |
| 21:27.04 | vader-- | anything i should fear in upgrading? |
| 21:27.09 | vader-- | im in production now :( |
| 21:29.20 | sebatk | CunningPike: any idea?? |
| 21:29.27 | Skyelar | vader--: try it on a dev system first - alternatively, I packported the changes from 1.2.10 to 1.2.7.1, so can give you the patch I guess |
| 21:30.44 | CunningPike | sebatk: Both your cards are sharing IRQ5 with both your NICs. Bad scene, man |
| 21:30.56 | CunningPike | sebatk: You need to get each card on its own IRQ |
| 21:31.32 | sebatk | dou you have a link to info on how to do it |
| 21:31.39 | sebatk | ?? |
| 21:31.51 | CunningPike | sebatk: Not really - we had to disable USB to free up IRQs |
| 21:32.24 | CunningPike | sebatk: If you search for zaptel and DL380, you should turn up some stuff - people have done it.... |
| 21:32.25 | sebatk | so you say if ifree some irqs it should get one for ech one?? |
| 21:32.50 | CunningPike | sebatk: You will need each card on its own IRQ - not shared with anything else |
| 21:33.04 | CunningPike | sebatk: Or switch to Sangoma cards :D |
| 21:33.18 | *** join/#asterisk num000 (n=numerobi@e177183003.adsl.alicedsl.de) |
| 21:33.48 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 21:33.51 | sebatk | CunningPike: ok i understando that , but you say if i free some irq the card should get automatically a new irq without further change?? |
| 21:33.59 | sebatk | and about the Sangoma cards |
| 21:34.05 | sebatk | i don't know them |
| 21:34.05 | num000 | CunningPike I need to find out how to point asterisk to soundfiles, can you point me to a documentation? |
| 21:34.09 | sebatk | if you send me info |
| 21:34.15 | variable_office | how do you make asterisk dial two things at the same time? isnt it DIAL/res && DIAL/res ? |
| 21:34.18 | sebatk | I have like 50 digium cards |
| 21:34.26 | sebatk | and always buying new ones |
| 21:34.29 | CunningPike | sebatk: You may have to dick around in the BIOS settings to get the cards on their own IRQs |
| 21:34.59 | sebatk | ok thanks |
| 21:35.00 | *** part/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:35.05 | sebatk | I will try |
| 21:35.05 | CunningPike | num000: It's usually a relative path from /var/lib/sounds/asterisk |
| 21:35.38 | num000 | CunningPike no, here they are lying in /usr/lib/asterisk/sounds/ |
| 21:35.45 | num000 | and asterisk can not find them |
| 21:35.47 | CunningPike | num000: So Background(foo/bar) would play /var/lib/sounds/asterisk/foo/bar.whatever |
| 21:36.39 | num000 | can i force asterisk to use a particular directory for looking up sound files? |
| 21:36.42 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-134.dynamic.qsc.de) |
| 21:36.52 | [TK]D-Fender | num000 : yes if you provide an absolute path |
| 21:36.54 | intralanman | num000: you can indeed |
| 21:37.23 | num000 | so where do I do this? asterisk.conf? something like astsound => sounddirectory/ ? |
| 21:37.28 | intralanman | num000: are all of your sound files there or just a gourp of them? |
| 21:37.36 | [TK]D-Fender | num000 : And * does not put sound files in usr/lib, but rather var/lib. If you moved something you'd need to change a few other things to fix the default apths |
| 21:37.37 | intralanman | s/gourp/group/ |
| 21:37.39 | num000 | no all of them |
| 21:37.50 | intralanman | jbot's just slow |
| 21:38.18 | num000 | intralanman no all of them are lying there all in gsm format |
| 21:38.19 | CunningPike | num000: How did they end up there? |
| 21:38.29 | intralanman | if all of them are there..... you can change it in asterisk.conf |
| 21:38.34 | num000 | CunningPike it must have something todo with the distribution openwrt |
| 21:38.36 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 21:38.37 | CunningPike | num000: Did you change the Makefile? |
| 21:38.40 | CunningPike | num000: Ah, OK |
| 21:38.50 | CunningPike | num000: Then asterisk.conf it is then :D |
| 21:38.51 | num000 | intralanman where do i specify this? |
| 21:39.08 | num000 | CunningPike let me try? could it be astsoundir => ??? |
| 21:39.43 | CunningPike | num000: There is an extremely good chance |
| 21:39.46 | CunningPike | :) |
| 21:39.52 | num000 | i'm going to try astsoundir ;) |
| 21:39.53 | intralanman | heheh |
| 21:40.08 | num000 | asterisk has very good configuration files ;) |
| 21:40.38 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:41.40 | num000 | no unfortunately not, it still gives me this error: channel.c:1703 ast_set_write_format: Unable to find a path from gsm to alaw |
| 21:43.23 | num000 | ahhh, it is trying to open them in alaw format: Unable to open demo-echotest (format alaw) |
| 21:43.30 | num000 | where did i tell him this |
| 21:44.38 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 21:46.28 | syzygyBSD | grep alaw /etc/asterisk/* |
| 21:47.11 | num000 | syzygyBSD the sip.conf contains some allow=alaw in the phone sections |
| 21:47.38 | syzygyBSD | I dont' know what you are trying to do, just where alaw was |
| 21:49.55 | num000 | why is asterisk trying to play the demo-echotest as a alaw encoded sound file and not as gsm? |
| 21:49.58 | *** part/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 21:50.37 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 21:54.12 | sebatk | CunningPike: why is that the cat /proc/int.... shows thath are in differnets irqs and the lspci -vb shows the same?? dou you think this is the problem?? |
| 21:54.46 | *** join/#asterisk zeppelin_ (n=zeppelin@201.11.211.183) |
| 21:55.12 | CunningPike | sebatk: ACPI - they are 'virtual' interrupts, whereas Digium require separate physical interrupts |
| 21:55.21 | sevard | Does anyone have anything for linux that takes CSV files and spits out XLS files? Pref CLI :| |
| 21:55.42 | CunningPike | sevard: OpenOffice :) |
| 21:55.46 | [hC] | Theres a perl module for dealing with xls<->csv |
| 21:56.01 | eKo1 | why do you need csv to xls? |
| 21:56.27 | num000 | CunningPike why is asterisk trying to play the demo-echotest as a alaw encoded sound file and not as gsm? |
| 21:57.08 | hads | num000: You are on an alaw channel? |
| 21:57.22 | sevard | eKo1: billing, my telco sends me a friggen file that's generated by cobol, i wrote a script to dice it into a neat csv, now i found out that the billing platform is rejecting csvs due to a bug but it does like xls files |
| 21:57.32 | num000 | hads how do i find this out? |
| 21:57.32 | sevard | so i'm just dealing with accounting right now |
| 21:57.33 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6E199.versanet.de) |
| 21:58.06 | hads | num000: show channel Foo/Bar |
| 21:58.22 | eKo1 | sevard: wouldn't it make sense to fix the billing platform? |
| 21:59.06 | sevard | eKo1: I'm not incharge of that. |
| 21:59.35 | num000 | hads yes it is a alaw channel, where can i change this? |
| 22:01.47 | *** join/#asterisk topping (n=topping@207.47.6.207.static.nextweb.net) |
| 22:03.06 | num000 | hads can you point me to a documentation which explains how to force to usage of a particular format, like gsm? |
| 22:03.36 | *** part/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 22:08.03 | hads | num000: It depends on what type of channel you are using. |
| 22:08.14 | num000 | hads how is this defined? |
| 22:09.01 | num000 | what defines the format of a channel? is it defined by the phone, or by configuration of asterisk? |
| 22:09.22 | hads | num000: And do you need to? Asterisk should try to play an alaw file and if one doesn't exist then it will play a gsm |
| 22:09.39 | hads | num000: Type of channel i.e SIP or Zap or what? |
| 22:09.45 | num000 | hads but it doesn't play the gsm file |
| 22:10.00 | num000 | hads not the format of a channel, like gsm or alaw |
| 22:10.34 | *** join/#asterisk TrevorSHarrison (n=trevorsh@24-49-36-218-st.chvlva.adelphia.net) |
| 22:10.56 | hads | What type of phone are you using? |
| 22:11.12 | num000 | it is a nokia e60 with sip support |
| 22:12.28 | hads | OK so that won't support GSM most likely. |
| 22:12.37 | JT | the irony |
| 22:12.38 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:12.46 | num000 | the irony, you ar right jt |
| 22:12.50 | hads | heh |
| 22:13.00 | num000 | what can i do now? |
| 22:13.09 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 22:13.10 | hads | So Asterisk will either need alaw sound files or it will need to transcode from something to alaw. |
| 22:13.12 | num000 | find the appropriate files in alaw format? |
| 22:13.15 | Hmmhesays | ok wtf kind of bank doesn't have a swift code |
| 22:13.31 | num000 | Hmmhesays all of them do have swift codes |
| 22:13.32 | hads | It should transcode automatically. |
| 22:13.44 | num000 | hads but it doesn't how can i find this out? |
| 22:13.52 | num000 | or where can i find the alaw files? |
| 22:13.58 | Hmmhesays | num000, no they don't |
| 22:14.03 | JT | diagnostics/verbosity? |
| 22:14.25 | num000 | verbosity is level 3, will higher number tell me more? |
| 22:14.55 | *** join/#asterisk topping (n=topping@207.47.6.207.static.nextweb.net) |
| 22:15.07 | Skyelar | num000: if you do a "show translation" in the console, what's in the "alaw" to "gsm" columns? |
| 22:15.13 | hads | Pastebin the general section and relavant phone section of your sip.conf and a console log with verbose 5 |
| 22:15.35 | hads | Yeah, what Skyelar said |
| 22:15.59 | hads | Good thinking |
| 22:16.03 | JT | building it without g711 support would be pretty neat, too |
| 22:16.10 | hads | heh |
| 22:16.17 | Skyelar | num000: if you're not 100% sure what you're looking at, pastebin the entire output |
| 22:16.45 | num000 | Skyelar ok, moment i'm going to paste it to a pastebin server |
| 22:18.56 | Hmmhesays | ok, anyone ever transfered money to a us bank without a swift code? |
| 22:19.52 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 22:20.07 | num000 | here we go: http://channels.debian.net/paste/3482 |
| 22:20.37 | num000 | i've done a show translation and also pasted the log during calling the echotest |
| 22:20.54 | Skyelar | num000: whoah. That's an extremely sparse translation list |
| 22:21.20 | num000 | Skyelar it is on a openwrt box, just 3mb of space |
| 22:21.29 | hads | So no alaw then :) |
| 22:21.32 | Skyelar | num000: you don't have alaw support :) |
| 22:21.44 | num000 | hads no alaw support yes ;( |
| 22:21.58 | num000 | so what am i going to do? god oh god |
| 22:22.01 | hads | Possibly the modules aren't loaded? |
| 22:22.16 | num000 | mhhh |
| 22:22.23 | num000 | the asterisk modules which support alaw? |
| 22:22.26 | num000 | uyyy |
| 22:22.48 | num000 | will alaw module be enough? |
| 22:23.31 | Skyelar | num000: codec_alaw.so - that should do it (or you could force the e60 to ulaw) |
| 22:23.31 | Skyelar | ... that's *if* the alaw module is available on the openwrt build |
| 22:23.31 | num000 | load codec_alaw.so |
| 22:23.32 | num000 | <PROTECTED> |
| 22:23.33 | num000 | <PROTECTED> |
| 22:23.33 | num000 | <PROTECTED> |
| 22:23.53 | num000 | ohhh it works |
| 22:23.55 | JT | it looks like num000 is in germany, so there'd bo no reason he'd want ulaw :) |
| 22:24.05 | num000 | jt yes |
| 22:24.15 | JT | only north america and japan uses ulaw |
| 22:24.42 | num000 | ok, so why wasn't it loading the module? since noload => codec_adpcm.so ; Adaptive Differential PCM Coder/Decoder |
| 22:24.43 | num000 | ; load => codec_alaw.so |
| 22:24.47 | num000 | it starts like this |
| 22:24.59 | Skyelar | uncomment that "load => codec_alaw.so" line |
| 22:25.00 | JT | good to see it works now |
| 22:25.13 | num000 | yes, i'm happy ;) |
| 22:25.40 | num000 | hads we do have alaw support now ;) |
| 22:25.45 | *** part/#asterisk sebatk (n=sebatk@r200-40-61-230.ae-static.anteldata.net.uy) |
| 22:25.50 | hads | Cool |
| 22:26.11 | *** join/#asterisk Grnd-Wire (i=GrndWire@67-40-17-231.tukw.qwest.net) |
| 22:26.52 | *** join/#asterisk intralanman (n=intralan@pool-72-82-74-171.nrflva.east.verizon.net) |
| 22:29.18 | num000 | JT: so you are joking about nokia that they bastards did not implement gsm coding for sip into a gsm phone? |
| 22:29.28 | JT | i didn't say that |
| 22:29.37 | hads | There are different types of GSM |
| 22:29.41 | num000 | no? ;) or something similar |
| 22:29.47 | JT | hads said it was a possibility |
| 22:29.49 | Skyelar | num000: they e60's I've seen only have alaw/ulaw and g729, and g729 seems to be broken |
| 22:30.18 | quid246 | hmmhesays: are you getting involved in Nigerian 419 scams? |
| 22:30.23 | JT | that sounds horrible if you wanted to do voip over gprs or similar |
| 22:30.29 | num000 | Skyelar i see, but it works actually very well here, i'm very happy with this phone, except that is shows me a wrong wlan mac adress as it actually has |
| 22:31.07 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 22:31.32 | num000 | Skyelar what do you think of the e60? |
| 22:31.59 | Skyelar | num000: my boss has one, and quite likes it |
| 22:32.27 | num000 | Skyelar do you know or can you find out if his equipmentalso shows him a wrong wlan mac adress? |
| 22:32.28 | Skyelar | No STUN support though :-( |
| 22:32.37 | num000 | yes no nat traversal |
| 22:33.38 | Skyelar | num000: call me a prat, but I'm just picking out interesting questions I feel like looking at. The wrong MAC one isn't one of them :-) |
| 22:34.35 | num000 | it wasn't thought for you comfort, i asked you to get this information for me |
| 22:34.45 | num000 | but thanks anyway |
| 22:36.42 | *** join/#asterisk jhamlyn (i=jhamlyn@203.33.186.199) |
| 22:39.05 | Hmmhesays | I get to install centos on this dell 2950 today |
| 22:39.06 | Hmmhesays | woot |
| 22:39.14 | Skyelar | num000: sorry, I wasn't actually meaning to be as rude as that sounds on a re-read |
| 22:39.34 | JT | argh centos |
| 22:39.36 | num000 | Skyelar : no, i didn't think that. No problem Skyelar |
| 22:39.37 | JT | :) |
| 22:39.49 | inv_arp[work] | awww |
| 22:40.27 | num000 | cheeky bugger? what is that? |
| 22:40.34 | num000 | i need to improve my english |
| 22:40.36 | JT | i just had to /whois Skyelar |
| 22:40.42 | JT | no-one in america says that |
| 22:40.50 | JT | new he was from this part of the world :P |
| 22:41.01 | JT | australia/nz |
| 22:41.03 | justinu|laptop | cheeky bugger == smartass |
| 22:41.14 | Skyelar | num000: hmm, NZ/AU slang I guess - as justinu|laptop says |
| 22:41.34 | JT | except we'd say smartarse, heh |
| 22:41.49 | num000 | jt ohhh smartarse, cool |
| 22:43.21 | hads | A NZer. No Wonder ;) |
| 22:43.24 | num000 | cheeky bugger is new zealandish |
| 22:43.36 | JT | and australian too |
| 22:43.40 | JT | pretty interchangeable |
| 22:44.07 | Skyelar | hads: I probably should introduce myself (since I know of you) - privmsg ok? |
| 22:44.12 | quid246 | ya, though NZers aren't as brash |
| 22:44.12 | hads | Sure. |
| 22:44.24 | JT | quid246: lies! |
| 22:45.00 | num000 | does anyone know if these test files do exist in different languages? |
| 22:45.09 | quid246 | no, Aussies are more rough around the edges... NZers are more refined |
| 22:45.11 | num000 | like new zealandish |
| 22:45.29 | quid246 | Apologies to any Aussies, bt I've met tonnes who are always looking for soem kind of freebie. |
| 22:45.44 | num000 | freebie? |
| 22:45.49 | JT | quid246: and americans (?) are quick to make generalisations? :) |
| 22:45.56 | quid246 | yeah, free place to stay, free rides, free meals, etc. |
| 22:46.08 | num000 | jt thats true, generalisations |
| 22:46.13 | quid246 | aussies love to do Round The World trips... and are always looking for freebies |
| 22:46.22 | quid246 | I work in an airport, so I've seen it countless times. |
| 22:46.38 | JT | everyone loves a good bargain |
| 22:46.39 | quid246 | JT: Problem is, I ain't America. :) |
| 22:46.44 | quid246 | American |
| 22:46.52 | JT | what are you? |
| 22:46.55 | num000 | human is human |
| 22:46.59 | quid246 | Canadian |
| 22:47.04 | num000 | cool |
| 22:47.06 | JT | north american |
| 22:47.14 | num000 | you guys are from all over the world |
| 22:47.29 | quid246 | Yeah, we are all "African" |
| 22:47.53 | num000 | african? no i'm not |
| 22:47.59 | quid246 | "I ain't from Africa, you African Booty Scratcher" - Boyz 'N' The Hood |
| 22:48.18 | num000 | booty scratcher??? |
| 22:48.20 | crochat | Hello ! |
| 22:48.23 | quid246 | hehe |
| 22:48.32 | quid246 | had to have been there |
| 22:48.38 | justinu|laptop | num000: you need to listen to more rap music |
| 22:48.39 | crochat | I'm trying to convert an mp3 file to native Asterisk format with mplayer |
| 22:49.04 | num000 | justinu|laptop you are probaply right |
| 22:49.30 | num000 | but can someone tell me what the hell means booty scratcher, i suppose even a dictionary couldn't tell me this |
| 22:49.31 | crochat | I'm nearly ok, but I can't merge the two channels in one... it seams on channel is lost, and not merge in the mulaw stream :-( |
| 22:49.39 | justinu|laptop | num000: ubrandictionary.com is also your friend :) |
| 22:49.39 | crochat | mplayer -really-quiet -quiet -shuffle -ao pcm -af format=mulaw,channels=1,resample=8000 /usr/share/asterisk/mohmp3/Babylon_Zoo_-_Spaceman.mp3 |
| 22:49.50 | justinu|laptop | s/ubran/urban/ |
| 22:50.25 | Grnd-Wire | Can anyone tell me why there isn't a WCFXS module installed on this Trixbox by default? |
| 22:51.28 | num000 | crochat is ulaw the native asterisk format? |
| 22:52.07 | russellb | num000: no, it would be slin |
| 22:52.15 | russellb | Grnd-Wire: try wctdm |
| 22:52.20 | crochat | num000: With my mplayer commandline, it works fine with moh-native |
| 22:52.23 | russellb | wcfxs is from 1.0 days |
| 22:52.45 | crochat | [native] |
| 22:52.53 | crochat | mode=files |
| 22:53.07 | crochat | directory=/usr/share/asterisk/moh-native |
| 22:54.09 | num000 | does anyone know if the files do exist in different languages? |
| 22:54.59 | Grnd-Wire | russellb: err - Does that mean TrixBox 1.1 is really old? or is it reasonably current? |
| 22:55.06 | crochat | num000: It works, but it's like only one channel can be heard... sometimes, there's music, and sometimes, there's silence... like if you hear to a file with the balance positioned right or left, but not centered |
| 22:56.16 | russellb | Grnd-Wire: i'm saying try wctdm instead of wcfxs. in any case, try #trixbox |
| 22:56.33 | infinity1 | i'm having hell with dtmf. the connection to my provider is via IAX and i have jitterbuffer=no |
| 22:56.49 | infinity1 | iax debug shows the #'s i press on the fone, but the calling party doesn't get it |
| 22:58.36 | *** join/#asterisk intralanman (n=lanman@pool-72-82-74-171.nrflva.east.verizon.net) |
| 22:58.43 | Grnd-Wire | russellb: Yeah, there's a wctdm module that is being loaded properly - but there's a wcfxo that's on the filesystem too.. So I'm just confused about why they'd be trying to load something so old. <sigh> I guess the fact I'm the only person in #trixbox is a bad sign, eh? |
| 22:59.06 | russellb | er, #freepbx maybe :) |
| 22:59.13 | russellb | wctdm and wcfxo are fine |
| 22:59.28 | russellb | i can explain why they're named the way they are if you care, heh |
| 22:59.40 | Grnd-Wire | russellb: I actually do.. |
| 22:59.52 | Grnd-Wire | russellb: So spill it. :D |
| 23:00.08 | russellb | ok, so, back in the day, Digium had the X100P as an FXO card, and the TDM400P as an FXS card (it didn't support FXO yet) |
| 23:00.15 | russellb | so the moduels for them were wcfxo and wcfxs |
| 23:00.26 | Grnd-Wire | ahh.. ok.. |
| 23:00.27 | russellb | later, the X100P was retired, and FXO support was added to the TDM400P |
| 23:00.35 | russellb | and the module got renamed to wctdm |
| 23:01.11 | russellb | and that is that :) |
| 23:01.15 | Grnd-Wire | yes! Makes sense.. ok.. |
| 23:02.15 | Grnd-Wire | Well - I just got my TDM11 from Digium today, and I'm so excited to make it work.. but so far I've had no luck. :( I've got the modules loading correctly, the lights are green on the card, I have talk voltage on my analog phone, presumably Asterisk is setup right.. |
| 23:02.55 | Grnd-Wire | russellb: I'm just trying to get lousy dialtone.. I've been reading all sorts of stuff - but the problem is none of it explains a troubleshooting strategy.. Since everything is "working", I don't have any error messages to work with! |
| 23:03.25 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 23:03.58 | russellb | Grnd-Wire: well, you get free support from support@digium.com |
| 23:04.25 | russellb | or, it's probably easy enough, I could just log in and fix it ... |
| 23:04.44 | Grnd-Wire | russellb: ya, but I don't learn anything then.. |
| 23:05.02 | russellb | screen session! |
| 23:05.17 | crochat | Seems to work better with: mplayer -really-quiet -quiet -shuffle -ao pcm -af format=alaw,pan=1:0.5:0.5,resample=8000 /usr/share/asterisk/mohmp3/Babylon_Zoo_-_Spaceman.mp3 |
| 23:06.01 | crochat | alaw or mulaw... both work |
| 23:06.23 | Grnd-Wire | russellb: oooh, right.. hmm - That sounds good to me.. |
| 23:07.00 | Grnd-Wire | russellb: I have to install screen though.. I'll get back to you in a second :D |
| 23:07.17 | russellb | k, just /msg me the login info |
| 23:07.21 | Grnd-Wire | sure |
| 23:08.35 | quid246 | hmm, is there a good howto on getting * to run as non-root? Only can find stuff from 2004... so I dunno if any changes have been made to the source since then for that |
| 23:09.24 | hads | quid246: There isn't much you need to do. |
| 23:09.30 | justinu|laptop | it's just setting permissions |
| 23:09.44 | hads | chown -R asterisk:asterisk /var/lib/asterisk |
| 23:10.16 | quid246 | okay, the stuff I found talks about recompiling and setting up a /var/run/asterisk |
| 23:10.40 | quid246 | but evidently it's already in place |
| 23:10.41 | hads | Oh yeah, /var/run/asterisk and /var/log/asterisk too |
| 23:10.55 | quid246 | okay, thanks |
| 23:11.02 | hads | setup in asterisk.conf |
| 23:11.12 | *** join/#asterisk _fenlander (n=fenlande@82.152.81.57) |
| 23:12.05 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-120-136-36.gdrpmi.dsl-w.verizon.net) |
| 23:14.04 | num000 | ohh god, my asterisk crashes when i make a call out |
| 23:14.10 | *** join/#asterisk Alystair (i=Alystair@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
| 23:14.23 | Alystair | What's the general opinion of Grandstream SIP phones? |
| 23:14.39 | Alystair | are they one of the better brands? |
| 23:15.03 | file | they're cheap, but what one are you referring to? |
| 23:15.15 | joe | Alystair: they suck imho |
| 23:15.25 | joe | Alystair: polycom |
| 23:15.35 | joe | is the way to go |
| 23:15.46 | Alystair | anything... cheaper? |
| 23:15.49 | quid246 | gransdstreams aregood for experimination, but I'd hesitate to put that into a prdouction environment |
| 23:16.08 | Alystair | the polycom would dent the wallet a bit hard |
| 23:16.30 | joe | Alystair: the 301 and 501 are not that that much |
| 23:17.10 | hads | Outside of the US Polycom's are expensive. |
| 23:17.29 | joe | ah |
| 23:17.30 | Alystair | we need about 7 "standard" phones and a single dashboard for the front |
| 23:17.34 | Alystair | and I'm in Canada |
| 23:18.00 | joe | Alex: I paid 124 from cdw iirc for the 301's and 189 for the 501 |
| 23:18.16 | hads | Well, maybe outside of America is more accurate :) |
| 23:18.51 | tzanger | well I am discovering something |
| 23:18.54 | tzanger | nokia 6265i ignores all other bluetooth devices when it's in headset mode |
| 23:18.55 | joe | hads: how much are they in down in nz? |
| 23:18.57 | tzanger | but it's not a nokia 6265i limitation |
| 23:18.59 | tzanger | the motorola razr does the exact samet hing |
| 23:19.02 | tzanger | when it's in headset/handsfree mode it is not discoverable nor does it participate in any SDP browsing |
| 23:20.05 | hads | joe: As a comparison; IP601 = $835 / snom 360 = $510 / Aastra 480i = $505 |
| 23:20.10 | *** part/#asterisk angom_w (n=angom@red-corp-200.79.148.126.telnor.net) |
| 23:20.24 | joe | Alystair: what do you want out of the "dashboard" btw and what sorts did you have in mind? |
| 23:20.34 | JT | tzanger: probably because it'd running in realtime mode |
| 23:20.42 | Alystair | just to transfer/hold/etc |
| 23:20.45 | joe | hads: yikes |
| 23:20.57 | hads | Yeah :/ |
| 23:21.00 | Alystair | Anyone heard of Zultys? |
| 23:21.15 | JT | geebus, just buy from .au, hads |
| 23:21.19 | JT | cheaper than that |
| 23:21.29 | JT | or the us |
| 23:21.42 | tzanger | JT: but even if the phone is not in an active audio connection, it's still ignoring everything |
| 23:21.45 | hads | JT: I sell them all |
| 23:22.00 | hads | JT: So that's going through the official channels |
| 23:22.22 | JT | i'm guessing ebay isn't an official channel |
| 23:23.16 | hads | Heh. Also that includes 12.5% GST, which you have to pay on anything you bring them into the country. |
| 23:24.07 | JT | do you have to pay 12.5% GST on EVERYTHING imported into nz? |
| 23:24.38 | hads | Yah, unless it's value is under a couple of hundred dollars. |
| 23:24.49 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:24.52 | hads | (total shipment) |
| 23:24.58 | JT | yeah |
| 23:25.16 | JT | in .au, it's AUD$1000 via postal service, AUD$250 via courier (like fedex) |
| 23:25.20 | JT | before they start charging |
| 23:25.40 | hads | And only 10% :) |
| 23:25.53 | JT | unless duty applies... |
| 23:26.03 | JT | which depends on type of goods and country or origin |
| 23:26.10 | JT | s/or/of/ |
| 23:29.48 | crochat | num000: I had the same problem (Asterisk crash when a call was finished) when I had the Ubuntu Breezy version (1.0.9)... now, with Dapper version (1.2.7.1), no problem anymore ! |
| 23:31.27 | Alystair | hmm |
| 23:31.50 | Alystair | why is there no site with VOIP phone reviews! >:| |
| 23:31.59 | Alystair | I'll have to take it into my own hands |
| 23:32.13 | Alystair | and start a professional review site for VOIP equipment |
| 23:32.14 | justinu|laptop | it's a conspiracy to sell more voip phones |
| 23:32.32 | Alystair | so has no one here used Zultys phones? |
| 23:35.01 | *** join/#asterisk intralanman (n=lanman@pool-72-82-74-171.nrflva.east.verizon.net) |
| 23:35.22 | crochat | Alystair: I don't know Zultys phones, but I have a AT-320 from atcom.cn (low cost, a lot of firmwares (different languages and protocols like SIP, IAX2, H323, MGCP)... great ! |
| 23:35.22 | *** join/#asterisk sumasuma (n=chumma@cm233.omega181.maxonline.com.sg) |
| 23:35.34 | sumasuma | shall i replace fxs module in iaxy with fxo module ? |
| 23:35.38 | sumasuma | will it work ? |
| 23:36.22 | joe | sumasuma: well depends what you are trying to do fxs are for stations ie phones andthe like fxo are connectong to pstn ie phone lines |
| 23:44.54 | Alystair | has anyone here used Axon? |
| 23:46.31 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-155-98.dyn.embarqhsd.net) |
| 23:51.26 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:52.33 | orlock | JT: apparently vic police went digital 10 or so days ago |
| 23:52.49 | Hmmhesays | talkin bout looooove |
| 23:53.54 | JT | longer than that i believe orlock |
| 23:54.17 | JT | heard they're still having trouble with some areas |
| 23:54.25 | JT | i have the gear to monitor digital though :) |
| 23:54.29 | orlock | cool |
| 23:55.18 | [hC] | JT: isnt it encrypted? |
| 23:56.31 | JT | no |
| 23:56.35 | JT | maybe one day |
| 23:56.48 | JT | digital encoded |
| 23:56.59 | JT | encryption is something they can add relatively easily |
| 23:57.19 | Juggie | can you listen to CDMA/GSM phones? |
| 23:58.24 | *** join/#asterisk mtaht4 (n=m@dsl-63-249-108-250.cruzio.com) |
| 23:58.46 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 23:59.46 | justinu|laptop | GSM was cracked pretty easiliy |
| 23:59.56 | justinu|laptop | CDMA is tough... spread spectrum/frequency agile |
