| 00:01.16 | *** join/#asterisk KuJaX (n=weee@customtrading.dsl.xmission.com) |
| 00:03.49 | *** join/#asterisk mrtelnet (n=mrtelnet@c-67-173-191-235.hsd1.in.comcast.net) |
| 00:05.05 | mrtelnet | is there a way to have asterisk hint a blf as the state of a variable? |
| 00:05.24 | russellb | mrtelnet: yeah, using func_devstate |
| 00:06.19 | KuJaX | My asterisk server use to e-mail us when a voicemail message was left but now it doesn't. I am using CentOS, where would be the first place to look to troubleshoot this? |
| 00:06.32 | russellb | mrtelnet: http://asterisk.org/node/48325 |
| 00:07.06 | russellb | and for 1.4 ... http://asterisk.org/node/48360 |
| 00:07.31 | mrtelnet | @russelb: Thank you! |
| 00:07.36 | russellb | mrtelnet: you're welcome |
| 00:07.48 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 00:08.46 | ZX81 | someone gimme news :) |
| 00:08.58 | ZX81 | what's happening in the world of Asterisk |
| 00:09.25 | ZX81 | might write up about the jack thing |
| 00:09.28 | ZX81 | looks cool |
| 00:09.46 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 00:09.53 | mrtelnet | ZX81, thanks for helping me with that autodialout last month, it really helped me |
| 00:10.01 | ZX81 | :) sweet as |
| 00:10.08 | mrtelnet | but alas, i have no news |
| 00:10.23 | ZX81 | :) |
| 00:10.26 | russellb | ZX81: there was also the new res_phoneprov committed the other day |
| 00:10.31 | ZX81 | oh yeah |
| 00:10.35 | ZX81 | that looks good too |
| 00:10.40 | ZX81 | there we go 2 stories |
| 00:10.41 | ZX81 | :) |
| 00:10.45 | russellb | heh, yay |
| 00:22.58 | puck | KuJaX: The mail server logs on your asterisk box? |
| 00:23.03 | puck | Can you send email from that box? |
| 00:23.20 | KuJaX | puck: Is there a command I can run to test the mail server? |
| 00:23.48 | puck | use mail |
| 00:23.54 | puck | mail your@email.address |
| 00:26.14 | *** part/#asterisk RoyK (n=roy@91.149.24.225) |
| 00:27.24 | ZX81 | who wrote res_phoneprov? |
| 00:28.33 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 00:28.42 | ZX81 | i.e. who is twilson? |
| 00:30.19 | ZX81 | http://www.venturevoip.com/news.php?rssid=1934 |
| 00:30.35 | ZX81 | 66 more articles and we reach 2000! :) |
| 00:30.56 | DumFuq | why won't a trunk pass through a callers ID? |
| 00:31.57 | ZX81 | maybe there is a caller id on the trunk account? |
| 00:32.10 | ZX81 | :) love the handle :) |
| 00:32.36 | DumFuq | thanx |
| 00:33.15 | DumFuq | ZX81: TBH ... I'm using Trixbox 2.4 ... so I already know I'm asking in the wrong channel ... |
| 00:33.17 | *** join/#asterisk Winkie (n=urmom@149.254.192.192) |
| 00:33.42 | ZX81 | heh yeah - if we tell you to make config changes, they may work but will be overwritten next time you use the gui |
| 00:33.57 | drmessano | wow |
| 00:34.15 | ZX81 | shit meeting |
| 00:34.16 | ZX81 | late |
| 00:34.17 | ZX81 | woops |
| 00:34.18 | DumFuq | ZX81: but whenever I receive a call via BRI (Epygi QuadroISDN) ... I receive the caller ID as the username |
| 00:34.19 | drmessano | lol |
| 00:34.21 | DumFuq | lol |
| 00:34.34 | DumFuq | ZX81: username for the trunk that is |
| 00:34.58 | drmessano | Youre better off asking in FreePBX |
| 00:35.13 | drmessano | Since a lot of those things are GUI related more than Asterisk related |
| 00:37.35 | KuJaX | puck: very strange, I changed email address that it was going to get sent to (changed it to a gmail) and it went through, changed it back to the original email address (which I know works) and it does. |
| 00:37.37 | KuJaX | *doesnt. |
| 00:37.44 | ManxPower | Um, BRIs don't have a username associated with it. |
| 00:38.50 | drmessano | BRIs don't have a future associated with them either |
| 00:38.54 | drmessano | :/ |
| 00:39.03 | ManxPower | drmessano: you must be in the USA or Canada. |
| 00:39.08 | drmessano | yes |
| 00:39.22 | ManxPower | In may parts of the world BRI has basically replaced analog. |
| 00:39.23 | puck | KuJaX: I'd suggest checking the mail logs for your MTA (whatever it might be) |
| 00:39.31 | drmessano | Im also the owner of BRIsux.org |
| 00:39.41 | drmessano | j/k |
| 00:40.25 | drmessano | TJNII: That how much it sux, I wont even waste the time putting a site up |
| 00:40.32 | *** join/#asterisk lhfx21 (n=lhfx21@net-cdd-fw01.cddlasmercedes.com) |
| 00:40.38 | drmessano | Thats* |
| 00:40.43 | lhfx21 | Hello Everyone |
| 00:40.52 | TJNII | I don't know anything about it, I think I skimmed a wikipedia article on it once, that was it. |
| 00:41.19 | drmessano | Actually, the only thing wrong with BRI is the way the telco handles it in the states |
| 00:41.28 | drmessano | It's the black sheep |
| 00:41.35 | lhfx21 | Anyone can helpme with some problems with E1 links? |
| 00:41.45 | TJNII | Don't ask to ask, just ask. |
| 00:42.01 | TJNII | No garuantees you'll get an answer, though. :) |
| 00:43.13 | lhfx21 | Ok, I am trying to configure an TE412P with two E1 links, one for outgoinf calls and the other for incoming calls, 15 lines each |
| 00:43.43 | ManxPower | Channelized E-1 or PRI E-1? |
| 00:43.53 | lhfx21 | PRI E1 |
| 00:44.17 | JT | there is pretty much no such thing as a channelised E1, ManxPower |
| 00:44.31 | lhfx21 | The problem is that astreisk returns "CONGESTION" every times I try to make a outside call |
| 00:45.25 | lhfx21 | Cause 34 - Circuit/channel congestion |
| 00:45.44 | JT | do a pri intense debug |
| 00:45.46 | lhfx21 | zttool reports all ports OK |
| 00:45.49 | ManxPower | lhfx21: chances are it's an issue with the pridialplan setting. unknown is the common setting |
| 00:46.05 | ManxPower | JT: what do you call a voice E-1 that is not PRI? |
| 00:46.53 | lhfx21 | In extensions.conf i have: exten => _9NXXXXXX,1,Dial,Zap/g4/${EXTEN:1} |
| 00:47.04 | ManxPower | lhfx21: what is your pridialplan setting |
| 00:47.31 | lhfx21 | Sorry I don't have much experience with this |
| 00:47.40 | lhfx21 | Where I should find it |
| 00:49.03 | ManxPower | /etc/asterisk/zapata.conf |
| 00:49.58 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 00:51.05 | ManxPower | lhfx21: I assume group=4 is set in /etc/asterisk/zapata.conf right? |
| 00:51.18 | lhfx21 | Yes |
| 00:51.48 | lhfx21 | But I'm looking in zapata.conf and I think I dont have the pridialplan |
| 00:52.03 | puppet | Im tired of coding PHP, So ill make a RealTime Editor in C# instead ;P |
| 00:56.47 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
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| 00:58.26 | *** join/#asterisk l0verb0y (i=l0verb0y@210.1.137.41) |
| 00:58.41 | l0verb0y | hey can anyone recommend a good fax card? |
| 01:01.12 | *** join/#asterisk RoyK (n=roy@91.149.24.225) |
| 01:02.27 | lhfx21 | Any idea? |
| 01:04.04 | puppet | to tired to do a silly gui |
| 01:12.59 | *** join/#asterisk jblack (n=jblack@pool-71-181-138-192.sctnpa.east.verizon.net) |
| 01:16.02 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 01:16.16 | esaym | Anyone here using the HandyTone HT-286 ata adapter with asterisk? |
| 01:16.25 | esaym | I am wondering if it is any good |
| 01:18.45 | *** join/#asterisk techie (n=techie@adsl-76-214-28-29.dsl.lsan03.sbcglobal.net) |
| 01:19.44 | AndyGraybeal | do i need to configure zaptel.conf if i have a linksys 3102 ? |
| 01:19.49 | AndyGraybeal | spa3102 |
| 01:20.56 | TJNII | That's a sip device isn't it? |
| 01:21.13 | AndyGraybeal | i'm not really sure, it has fxo/fxs ports on it |
| 01:21.34 | AndyGraybeal | and it hooks to the network |
| 01:22.11 | AndyGraybeal | go ahead and make fun of me, i don't understand this stuff |
| 01:22.18 | TJNII | Well, I suggest you find out or else you'll never get it configured. |
| 01:22.38 | TJNII | I'm not trying to make fun, if you log into it you should be able to find out very quickly |
| 01:23.23 | AndyGraybeal | well.... i'm logged into it, what are the clues i should be looking for? |
| 01:23.47 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 01:24.00 | _ShrikE | AndyGraybeal: No, zaptel is not required for that device |
| 01:24.01 | *** join/#asterisk RoyKa (n=roy@91.149.26.90) |
| 01:24.12 | AndyGraybeal | _ShrikE: okay thank you. |
| 01:24.17 | _ShrikE | sip.conf |
| 01:24.24 | AndyGraybeal | ah bad ass thanks |
| 01:24.59 | TJNII | _ShrikE: That is the successor to the PAP2, correct? |
| 01:26.16 | AndyGraybeal | says it's the successor to the spa3000 |
| 01:26.22 | AndyGraybeal | but i have no idea about this stuff |
| 01:26.29 | _ShrikE | I believe thats correct |
| 01:29.16 | AndyGraybeal | i think the pap2 is a similiar but different thing |
| 01:29.24 | AndyGraybeal | i dont' think it has fxo/fxs |
| 01:29.41 | *** part/#asterisk RoyKa (n=roy@91.149.26.90) |
| 01:31.47 | TJNII | The pap2 has 2 fxs lines |
| 01:31.54 | AndyGraybeal | ah okay |
| 01:35.37 | [TK]D-Fender | AndyGraybeal, www.voxilla.com <- go read their forums to learn how to configure it with * |
| 01:36.19 | AndyGraybeal | rad, thank you [TK]D-Fender |
| 01:36.28 | jblack | aren't there two pap2s, a locked one, and an unlocked one? |
| 01:38.27 | [TK]D-Fender | jblack, correct |
| 01:38.37 | russellb | everyone be nice to AndyGraybeal ... he's going to help me write some cool Pd patches to mess with the audio of phone calls :) |
| 01:38.51 | jblack | Yay AndyGraybeal! |
| 01:38.56 | jblack | What's a pd? |
| 01:39.03 | russellb | http://puredata.info |
| 01:39.25 | russellb | graphical programming environment for audio/video analysis/manipulation/generation/etc |
| 01:40.00 | AndyGraybeal | ;) |
| 01:40.37 | jblack | please oh please have gsm output. |
| 01:41.17 | AndyGraybeal | [TK]D-Fender: you want me to put you as the referrer on voxilla ? |
| 01:41.35 | *** join/#asterisk d-tech (n=d-dtech@72.245.233.107) |
| 01:41.37 | AndyGraybeal | looks like i have to register to read the configuration wizard |
| 01:41.47 | [TK]D-Fender | AndyGraybeal, I don't have an account there. |
| 01:42.01 | [TK]D-Fender | AndyGraybeal, I didn't say to use a shmuck config tool. |
| 01:42.09 | [TK]D-Fender | AndyGraybeal, I said read their FORUMS |
| 01:42.26 | AndyGraybeal | i think it's like a guide, not a tool |
| 01:42.47 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 01:43.03 | AndyGraybeal | i wasn't sure there was a difference between the guides and the forums |
| 01:43.57 | JT | ManxPower: something that does not exist. |
| 01:46.36 | *** part/#asterisk techie (n=techie@adsl-76-214-28-29.dsl.lsan03.sbcglobal.net) |
| 01:47.13 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-71ecd631ba8b5f61) |
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| 01:57.48 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 02:04.00 | ZX81 | hi, I have to plug in 700 analogue extension - anyone know of a high port count sip gateway? |
| 02:04.12 | ZX81 | I can't imagine doing it with 8 port gateways |
| 02:05.41 | ZX81 | can I daisy chain xorcom units? :) |
| 02:06.03 | _ShrikE | ZX81: WOW.. Audiocodes has a nice 24 port gateway but you would still need almost 30 of them |
| 02:06.08 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 02:06.11 | ZX81 | yeah I know |
| 02:06.44 | InHisName | digium makes 24 port PCI cards. |
| 02:06.54 | DumFuq | ZX81: I don't think you can "daisy chain" the units |
| 02:07.09 | d-k-t | T3 channel bank? |
| 02:07.12 | ZX81 | :) |
| 02:07.13 | ZX81 | yeah |
| 02:07.16 | ZX81 | or max tnt maybe |
| 02:07.16 | InHisName | Tie 8 systems together and 24 of those boards. |
| 02:07.44 | ZX81 | my boss just said the per port price on the 24 ports are too expensive |
| 02:07.47 | InHisName | hmmm maybe 32 |
| 02:07.50 | ZX81 | he prefers the 8 port |
| 02:07.51 | ZX81 | lol |
| 02:08.07 | ZX81 | I'm going to have to friggin wire this shit lol |
| 02:08.11 | ZX81 | should make him do it |
| 02:08.12 | ZX81 | :) |
| 02:08.27 | d-k-t | ZX81, put 100 on each wire |
| 02:08.28 | InHisName | Then buy 24-30 computers to run 4 8 port boards ugggh |
| 02:08.34 | ZX81 | lol nice |
| 02:08.35 | ZX81 | go dundi |
| 02:09.02 | ZX81 | boss is looking at spa8000's |
| 02:09.34 | jblack | ZX81: I have one. |
| 02:09.37 | InHisName | Orrrrr buy a whole lotta ATAs from linksys and one really fast computer for the asterisk. |
| 02:09.41 | d-k-t | hotel or something? |
| 02:09.55 | ZX81 | nah just a business |
| 02:10.02 | ZX81 | lol they only have 1 E1 |
| 02:10.08 | ZX81 | most phones are unused |
| 02:10.11 | ZX81 | big site |
| 02:10.40 | d-k-t | and they don't want to swap out the phones |
| 02:10.48 | ZX81 | nah |
| 02:10.55 | ZX81 | but I'm actually considering it |
| 02:10.59 | ZX81 | is like 4k difference |
| 02:11.24 | jblack | If anyone is intersted, there's a dundi network forming. It covers chicago, northeastern pa, parts of italy, blue ridge GA, kissimmee,fl elburn,il and hartford, CT |
| 02:11.35 | ZX81 | problem is, its probably all twisted pair back to the closet |
| 02:11.51 | ZX81 | jblack, is kissme really a place? |
| 02:11.57 | jblack | It truly is. |
| 02:12.01 | ZX81 | :) sweet |
| 02:12.24 | _ShrikE | right by disney world :) |
| 02:13.21 | jblack | ZX81: want to know what I think about my spa8k ? |
| 02:13.25 | d-k-t | ZX81, had that issue before... well almost, cat5 back to the wiring closet on each floor, but, interfloor provided by bundled pairs... suitable for analogue or avaya digital handsets, but not very useful for ethernet |
| 02:13.32 | ZX81 | jblack, yeah for sure |
| 02:13.41 | ZX81 | compared to other 8 ports maybe :) |
| 02:13.52 | ZX81 | d-k-t, yeah |
| 02:13.59 | d-k-t | pain in the bum |
| 02:14.00 | jblack | It does what it promises, and is incredibly configurable. |
| 02:14.06 | ZX81 | i.e. we used the grandstreams and they were shocking |
| 02:14.13 | jblack | On the downside, it seems a little flakey to me. |
| 02:14.16 | ZX81 | jblack, that's what I like to hear |
| 02:14.31 | ZX81 | the grandstream settings don't always save |
| 02:14.57 | ZX81 | and a couple bricked on firmware update over crossover cable |
| 02:14.58 | ZX81 | lol |
| 02:15.11 | TJNII | "No application 'MeetMe' for extension" .... hmmmmm |
| 02:15.33 | jblack | tjnii: Perhaps you don't have a meetme config file. Also, make sure you have the zt-dummy kernel module. |
| 02:15.43 | TJNII | Oh, I see whats wrong. |
| 02:15.50 | TJNII | jblack: Yea, it's the lack of ztdummy |
| 02:16.25 | jblack | zx81: I also noticed that the spa web interface gets very laggy when it's registering. (It registers each phone port as a seperate sip account) |
| 02:16.43 | ZX81 | yeah same as the grandstream |
| 02:16.55 | jblack | Also, more than once I've noticed that it has greyed out options that should be configurable. |
| 02:17.05 | ZX81 | heh |
| 02:17.18 | jblack | I have one line stuck in nat compatibility mode. Why? No idea. You can turn it on, but not off. |
| 02:17.22 | ZX81 | they better have autoprovision if I'm going to set up 84 of them :) |
| 02:17.46 | jblack | they might. The 8k only has 8 lines, but if you have 8 of them... |
| 02:17.53 | jblack | I found the book not too long ago. Let me get it for you |
| 02:18.26 | d-k-t | The SPA8000 offers key features and capabilities that can enable service providers to offer customized services to their subscribers. The SPA8000 can be remotely provisioned and supports dynamic, in-service software upgrades. |
| 02:18.37 | ZX81 | cool |
| 02:19.07 | *** join/#asterisk Porks (i=Porks@200-148-39-76.dsl.telesp.net.br) |
| 02:20.52 | jblack | Of course I'm having trouble finding it now |
| 02:22.37 | jblack | Found it |
| 02:23.10 | jblack | blasted javascript |
| 02:24.20 | jblack | You can find it here: http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&pagename=Linksys%2FCommon%2FVisitorWrapper&cid=1175235457466 |
| 02:25.19 | jblack | zx81: On the whole, I think it works great for home. I don't know if I'd be comfortable putting it in a production environment without a babysitter. |
| 02:25.38 | ZX81 | yeah - but don't have much choice |
| 02:26.08 | jblack | Ok, well, it's controlled entirely over http, so at least you've got that. |
| 02:26.09 | ZX81 | I don't really want to be install 100 Asterisk boxes at one site |
| 02:26.13 | ZX81 | yeah |
| 02:26.18 | ZX81 | and I'll have ssh to the box |
| 02:26.26 | ZX81 | tunnelling to the cpe |
| 02:28.30 | d-k-t | it's definitely cheaper to use the linksys than the bigger 24/32 port gateways |
| 02:29.38 | d-k-t | $31 per port vs $63 per port for a mediatrix 1124 |
| 02:30.14 | ZX81 | yeah I know |
| 02:30.22 | ZX81 | but then $150 p/h to install |
| 02:30.26 | ZX81 | plus ongoing support |
| 02:30.33 | ManxPower | We usually end up spending about $200 per extension |
| 02:30.53 | ZX81 | yah |
| 02:30.53 | ManxPower | phone + switch + server + PRI card. |
| 02:31.10 | ZX81 | do you do cabling? |
| 02:31.11 | ManxPower | come to think of it, we probably spend more than $200 per extenstion |
| 02:31.34 | ZX81 | we went into a place before christmas and rats had eaten through cables in the wall |
| 02:31.44 | ManxPower | Me? No. |
| 02:31.59 | ZX81 | yeah, we've hooked up with a cabling company now |
| 02:32.24 | sevard | rats? i hate rats they drive me crazy. |
| 02:32.26 | d-k-t | at work we've typically spent probably around $1000 per extension |
| 02:32.32 | sevard | crazy? i've been crazy once, they put me in a round room with tons of rats. |
| 02:32.33 | sevard | rats? i hate rats they drive me crazy. |
| 02:32.34 | sevard | crazy? i've been crazy once, they put me in a round room with tons of rats. |
| 02:32.39 | ZX81 | :D |
| 02:33.08 | ZX81 | we're always upgrading our per port cost at the office, have like 4 phones on the tech desks :) |
| 02:33.23 | ZX81 | 5 1/2 on mine :) |
| 02:33.29 | sevard | ZX81: the techs always get the best toys |
| 02:33.36 | ZX81 | :) yep |
| 02:33.48 | d-k-t | I only have 2 phones on my desk |
| 02:33.56 | d-k-t | and one of those is my own from home |
| 02:33.59 | sevard | d-k-t: wtf, i have more phones in my pocket. |
| 02:34.05 | d-k-t | I need a new job |
| 02:34.15 | jblack | zx81: Here's how I'd put it. I got what i paid for. $250 worth of eqp |
| 02:34.15 | sevard | want to be my cleaning lady? |
| 02:34.35 | d-k-t | I do however have a small mountain of dead avaya sets behind my desk |
| 02:34.41 | ZX81 | :) |
| 02:35.04 | d-k-t | sevard, if the pay is right ;) |
| 02:35.09 | ZX81 | we've been taking dead phone (where the network kit works) and turning them into network test devices :) |
| 02:35.28 | sevard | d-k-t: the pay can be right if services rendered are extraordinary.... wink wink |
| 02:35.38 | ZX81 | brb |
| 02:36.20 | *** part/#asterisk Porks (i=Porks@200-148-39-76.dsl.telesp.net.br) |
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| 02:52.33 | InHisName | ZX81, tcs has 1,2,4,8,16,32 port ATAs found on voip-info.com http://www.telecomchinasourcing.com/ it is in red China, hmmmmm |
| 02:52.46 | ZX81 | :) |
| 02:53.27 | d-k-t | it's quite white is china at the moment |
| 02:53.29 | d-k-t | snow... |
| 02:57.46 | TJNII | Can you bridge meetme conference rooms across machines? Like 2 asterisk boxes handling one conference? |
| 03:03.56 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 03:06.00 | d-k-t | TJNII, can't think of a reason why not |
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| 03:23.30 | drmessano | TJNII Check the Trixbox.org forums.. there was a thread on there I was involved in over it |
| 03:23.45 | drmessano | the results were not FreePBX dependant, so dont worry ;) |
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| 03:29.55 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-198-40.dsl.irvnca.pacbell.net) |
| 03:30.01 | UnixDog | [Jan 13 22:29:09] NOTICE[35818]: chan_sip.c:13805 handle_request_invite: Failed to authenticate user "9998" <sip:9998@192.168.123.101>;tag=3A2666E2-C5FD3AF7 |
| 03:30.14 | UnixDog | its failing when I dial out |
| 03:30.48 | UnixDog | but canreivite=no |
| 03:32.15 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
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| 03:54.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:56.04 | jblack | unixdog: I'd say bad password. |
| 04:05.22 | *** join/#asterisk Oleg (n=Oleg@unaffiliated/) |
| 04:09.34 | *** join/#asterisk jblack (n=jblack@pool-71-181-138-192.sctnpa.east.verizon.net) |
| 04:19.19 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 04:19.42 | mosty | how large can i safely set iaxmaxthreads on machines with 2G and 4G of ram |
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| 04:30.03 | FireMac | can anyone help me with wiring setup. i have a dsl and i want my voip to service my whole house |
| 04:32.13 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:32.58 | jblack | Is there something I could do that would cause "core set debug 10" to now show the dialplan to not show inside of asterisk -r, in * 2.4.15 ? |
| 04:33.08 | jblack | s/now/not |
| 04:33.41 | jblack | Um. Let me rephrase that. I'm no longer seeing dialplan debug, despite setting debug to 10. Is there something I could have done wrong? |
| 04:34.02 | jblack | ahh. -rvvvvvvvvvvv is necessary |
| 04:34.13 | mosty | core set verbose 10 |
| 04:34.23 | mosty | you probably want to see verbose messages, not debug messages |
| 04:34.30 | jblack | yeah, I wanted verbose. |
| 04:41.44 | *** join/#asterisk c4t3l (n=c4t3l@c-98-200-2-74.hsd1.tx.comcast.net) |
| 04:42.26 | c4t3l | hello all |
| 04:43.31 | c4t3l | anyone in here ever heard of intuitive voice technology? |
| 04:44.24 | russellb | yeah, another gui |
| 04:44.32 | russellb | switchvox is much nicer IMO ;) |
| 04:44.49 | c4t3l | well, i dont like either one of them |
| 04:45.24 | c4t3l | how well does switchbox scale up to say 1050 users? |
| 04:45.34 | c4t3l | no no 150** |
| 04:46.19 | russellb | 150, definitely very well |
| 04:46.30 | c4t3l | i've just been noticing a disturbing trend... |
| 04:46.43 | russellb | 1050, i couldn't answer ... you'd have to ask them ... |
| 04:46.46 | russellb | what's that? |
| 04:47.00 | TJNII | Is there a 3 digit extension that is commonly used for conference calls? |
| 04:47.28 | c4t3l | @russellb approx how many instructions per phone call does switchvox generate? |
| 04:47.31 | TJNII | Or I should say, to dial into a conference room? |
| 04:47.35 | c4t3l | on the cli? |
| 04:47.56 | russellb | c4t3l: eh? i don't know what you're asking, and even if i did, i don't think i would know the answer |
| 04:48.08 | c4t3l | well check this out... |
| 04:48.40 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 04:48.59 | *** join/#asterisk s0lid (n=s0lid@210.213.243.178) |
| 04:49.00 | russellb | i'm about to check out bed |
| 04:49.29 | c4t3l | using a "plain vanilla" asterisk installation (ver 1.4 or later) with a simple dialplan, ext-to-ext dialling only generates 2 lines of output on the CLI |
| 04:49.50 | russellb | of course, that depends on your verbose/debug settings |
| 04:50.04 | c4t3l | whereas Intuitive or switchvox use at least 42! |
| 04:50.18 | russellb | i _know_ that you don't know that |
| 04:50.26 | russellb | because you can't even see the asterisk CLI on switchvox, i know. |
| 04:50.27 | *** join/#asterisk marcan (i=1337@host215-248.cvf.fit.edu) |
| 04:50.44 | c4t3l | sorry, I got a little over zealous there |
| 04:50.46 | russellb | and besides, lines of CLI output isn't a good measure of anything ... |
| 04:51.02 | c4t3l | what would be a better measure? |
| 04:51.30 | russellb | actually see how many calls you can process before things start to break? |
| 04:51.42 | piper69 | drmessano: hey man , a gift is in its way to you email :) |
| 04:52.29 | c4t3l | I've used old school * ver 1.2 and handled nearly 200 users |
| 04:52.48 | c4t3l | on IVT I can get to 75 |
| 04:53.19 | c4t3l | I just made an assumption that all the dialplan calls were a good place to start looking |
| 04:53.44 | russellb | gotcha ... |
| 04:53.47 | russellb | well, i'm not really surprised |
| 04:54.13 | c4t3l | @russellb: how's that? |
| 04:54.45 | russellb | well, there are lots of ways to build the system so that it supports all the GUI stuff |
| 04:54.55 | russellb | some options can really hurt performance |
| 04:55.06 | russellb | anyway, i've got to sleep ... |
| 04:55.08 | c4t3l | mysql calls and such? |
| 04:55.11 | c4t3l | goodnight |
| 04:55.12 | russellb | yeah |
| 04:55.18 | russellb | and who knows what else .. |
| 04:55.22 | c4t3l | hehe |
| 04:55.23 | russellb | g'night |
| 04:59.56 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 05:21.19 | *** join/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 05:29.39 | [TK]D-Fender | jblack, her 976 ones are far more.... entertaining ;) |
| 05:29.59 | *** join/#asterisk NWM0nKEY (n=robert@207.47.53.178.static.nextweb.net) |
| 05:30.03 | jblack | Heh. I noticed that there's "moron" in there. and a big pile under 'ha' |
| 05:32.14 | *** join/#asterisk angom (n=Angel@201.170.49.106) |
| 05:32.20 | jblack | With "system.gsm" and "power-failure.gsm", I thought perhaps that meant high availability... But with sump-pump, quiet-mode and baby-sleeping-mode and stove.gsm, I think they are jokes. |
| 05:33.38 | jblack | Nah, I bet this is... x-10 stuff. |
| 05:33.59 | jblack | one-of-these, is not like the other... one-of-these, does not belong. |
| 05:34.06 | drmessano | lol |
| 05:34.16 | drmessano | Some of the allison stuff is hilarious |
| 05:34.31 | drmessano | damn weasels |
| 05:34.55 | jblack | Hmm. with twisty-maze... that implies that there must be a grue recording somewhere? |
| 05:35.43 | drmessano | I want "Ohhh, a sword that smells like baking bread" |
| 05:36.04 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 05:36.09 | jblack | Looks like someone did a gps hookup at one point too. |
| 05:38.06 | InHisName | I am using GoToIfTime() and need to know which time is used for the decision. Seems to NOT be local time. I am in zone -5. I type date at linux prompt and see local time. |
| 05:38.51 | fujin | have you got your tz set correctly, localtime etc? |
| 05:40.31 | drmessano | jblack: I found a toy |
| 05:40.34 | jblack | Oh? |
| 05:40.50 | jblack | btw, can I steal your wakeup? The ones I have found are bit rotted. |
| 05:40.52 | drmessano | 6x3x6 metal two position single line phone switch |
| 05:41.03 | drmessano | Hang on |
| 05:41.57 | jblack | what's the 6x3x6 thing? A way to switch current with a phone line? |
| 05:43.20 | drmessano | 6 inches, 3 inches, 6 inches |
| 05:43.30 | drmessano | Big metal box |
| 05:43.33 | drmessano | 2 positions |
| 05:43.42 | drmessano | Like the old printer switches |
| 05:44.07 | jblack | What do you do with it? |
| 05:44.10 | InHisName | fujin, typing date at linux prompt shows my local time ( & name of zone) |
| 05:44.12 | drmessano | http://www.2l2o.com/asterisk/wakeup.rar |
| 05:44.25 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 05:44.28 | jblack | thanks |
| 05:44.28 | k-man | hello |
| 05:44.38 | drmessano | Switches one RJ-11 between A and B |
| 05:44.39 | jblack | aww, to cwd |
| 05:44.49 | jblack | Oh, ok. like an a/b switch. |
| 05:45.09 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 05:47.23 | *** join/#asterisk squigly (n=bdeluca@203.117.161.222) |
| 05:51.23 | drmessano | yes |
| 05:51.59 | jblack | That sounds so... manual |
| 05:53.00 | drmessano | http://www.mjs-electronics.se/images/Diverse/3db15f.jpg |
| 05:53.08 | drmessano | Like that, but RJ-11s |
| 05:53.23 | drmessano | Dude |
| 05:53.24 | jblack | yeah, I get it. |
| 05:53.31 | drmessano | thats not manual, thats failover trunking |
| 05:53.37 | drmessano | thats not manual, thats manual failover trunking |
| 05:53.46 | drmessano | lol |
| 05:54.00 | jblack | Ok.. But can't we already do that automatically, by setting multiple call routes? |
| 05:54.22 | drmessano | This is more 1.0...thats way too 2.0 |
| 05:54.51 | jblack | Fer instance... when I dial an 800#, first I try over fwd. Then, I use dundi. Failing that, I fall into callwithus. And if _THAT_ doesn't work, I hit up teliax. |
| 05:55.52 | drmessano | Ok.. With this, I pick up the phone and go "... oh crap", switch to Line B, and I am good to go |
| 05:56.34 | jblack | I could see where you could use that. |
| 05:56.43 | jblack | I want something like this: http://www.soundbytes.com/page/SB/PROD/SA200 |
| 05:58.15 | drmessano | Thats cool |
| 05:58.17 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 05:58.50 | jblack | I don't think this is _quite_ what I want. But I do want something where if someone rings the doorbell, the house phones ring, and I get an intercom |
| 05:59.37 | drmessano | I dont think that does it |
| 05:59.47 | drmessano | I think its a doorbell actuator and phone bell |
| 06:00.34 | jblack | I suspect that this thing probably pushes/passes through ring voltage to a connected phone. |
| 06:00.40 | jblack | I need something that'll dial |
| 06:01.31 | drmessano | "This multi-function signaler will alert you to a caller at the door and a caller on the telephone" |
| 06:01.37 | drmessano | Its a phone bell and a doorbell |
| 06:01.40 | drmessano | Doesnt backfeed |
| 06:02.06 | drmessano | But I know what you want |
| 06:02.09 | drmessano | I want the same thing |
| 06:02.35 | jblack | Yeah. a waterproof $3.99 walmart speaker phone with only one button. ;) |
| 06:02.44 | *** part/#asterisk piper69 (n=haiger@unaffiliated/piper69) |
| 06:03.27 | jblack | http://www.voip-info.org/wiki/view/Asterisk+phone+door |
| 06:03.35 | jblack | Sigh. Why do I try anywhere else first |
| 06:03.54 | jblack | Cool! BAT PHONES! http://www.redhotphones.com/ |
| 06:05.09 | jblack | Here's an overpriced version: http://www.redhotphones.com/haenausptryu.html |
| 06:05.50 | drmessano | Yes |
| 06:07.29 | drmessano | http://www.redhotphones.com/hevaredwapha.html |
| 06:07.35 | drmessano | I want that without the keypad |
| 06:07.40 | drmessano | Prison phone FTW |
| 06:08.46 | jblack | actually... I suppose _any_ ringdown phone would work, as long as it's resistant to the elements. |
| 06:09.12 | jblack | Just set up a special context for it, and it should drop into s |
| 06:09.54 | drmessano | jblack |
| 06:10.12 | jblack | Sir? |
| 06:10.17 | drmessano | You dont use FreePBX in any way, shape, or form.. correct? |
| 06:10.24 | jblack | Straight asterisk |
| 06:10.47 | jblack | Here you go.. http://www.ablecomm.com/auriinseupha1.html |
| 06:10.52 | jblack | no dtmf pad |
| 06:10.54 | drmessano | You should open a new window to freepbx and just idle |
| 06:11.09 | drmessano | Got a newb wanting to use Trixbox to make cheap calls |
| 06:11.14 | drmessano | It cheaper, no? |
| 06:11.28 | drmessano | Hes got an ATA and I guess wants free calls |
| 06:11.32 | drmessano | So he installed TB |
| 06:11.56 | fujin | tell him to piss off |
| 06:12.08 | jblack | Asterisk is free software. What's cheaper than that? Does he want someone to pay him to install software? |
| 06:12.37 | drmessano | HE WANT TO MAKE FREE CALL IT VOIP OVER INTERNET FREE LONGER DISTANCE, NO YES NO? |
| 06:12.46 | mosty | is the maximum number of voicemail messages for a particular account hardcoded in 1.2? |
| 06:12.55 | jblack | drmessano: Ohhhh. |
| 06:12.58 | fujin | drmessano: foad |
| 06:13.09 | drmessano | lol |
| 06:13.14 | jblack | Tell him to install... skype. |
| 06:13.22 | drmessano | LOL |
| 06:13.24 | drmessano | Thats cruel |
| 06:13.30 | drmessano | ..and my next step |
| 06:13.34 | jblack | Oh, oh, oh, I know! |
| 06:13.40 | jblack | http://www.payphone.com/shop/catalog/Pay_Phones-p-1-c-253.html |
| 06:14.57 | drmessano | I think I just peed a little |
| 06:15.00 | drmessano | YES.. I need one |
| 06:15.26 | jblack | Me to!! |
| 06:15.42 | jblack | I'd put it on the very edge of my property, next to the neighbor I _HATE_. |
| 06:15.59 | jblack | Perhaps put a sign on it that says "Drug dealers and hookers welcome!" |
| 06:16.34 | drmessano | Id put a PAP2 and gaming adapter in a weatherproof box under it, screw it to the wall at Wally World at 3am |
| 06:16.59 | jblack | http://www.sandman.com/autodial.html |
| 06:17.06 | jblack | Look for "NO-DIAL HANDSET" |
| 06:17.30 | drmessano | Sandman rocks |
| 06:17.46 | jblack | Oh, and on the same page, look for "1 Button ALERT DIALER" |
| 06:18.29 | drmessano | Hell yes |
| 06:19.47 | drmessano | The one button dialer is awesome |
| 06:20.30 | jblack | Of course... I get even less visitors than I do phone calls..... |
| 06:23.11 | drmessano | lol |
| 06:23.53 | jblack | http://cgi.ebay.com/Genuine-pay-phone-coin-operated_W0QQitemZ330202862716QQihZ014QQcategoryZ985QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 06:23.55 | drmessano | Yeah, between IRC and radio, I get enough of my friends |
| 06:24.14 | drmessano | When we see each other it's like "Didnt I just talk to you?" |
| 06:24.46 | jblack | This is annoying. Why are waterproof speakerphones so (#*@*&@ expensive? |
| 06:24.47 | drmessano | YES |
| 06:25.00 | drmessano | I can get those and then red box myself |
| 06:25.11 | drmessano | "Thats not a quarter, ass" |
| 06:25.16 | jblack | lol |
| 06:26.14 | drmessano | I remember reading something in 2600 a few years back about someone patching Asterisk for blue boxing |
| 06:26.38 | drmessano | i dont remember at all what the specifics were.. it was pre-asterisk for me |
| 06:27.05 | jblack | I think what I might do is just rip apart a cheap speakerphone and put it in a waterproof housing. |
| 06:27.30 | drmessano | Here is what you can do |
| 06:27.49 | drmessano | Rip the speakerphone apart, as you said |
| 06:27.57 | drmessano | Run a pair from the ATA to it |
| 06:28.07 | drmessano | put a DIAC in line to auto-answer |
| 06:28.21 | drmessano | and get a $19 wireless doorbell buzzer |
| 06:28.34 | drmessano | Someone buzzes, you call the line, the speakerphone answers, bam |
| 06:28.49 | jblack | Nah, that doesn't sovle the problem. |
| 06:29.00 | drmessano | Whats missing? |
| 06:29.01 | jblack | The problem I have is that in the rare occasion that someone does call, that I can't hear the doorbell. |
| 06:29.15 | jblack | does come calling. |
| 06:29.30 | jblack | I bet that home depot sells something appropriately crappy |
| 06:29.40 | drmessano | So the wireless doorbell buzzer wont work? |
| 06:29.57 | jblack | well, sure, if I get plenty of buzzers for each room of the house. |
| 06:30.01 | jblack | The point, though, is to wire it into * |
| 06:30.09 | drmessano | Lemme think |
| 06:30.12 | drmessano | I got ya |
| 06:30.13 | jblack | so that it calls me. I have plenty of alarms. |
| 06:30.33 | drmessano | You need an autodialer |
| 06:30.36 | drmessano | and a $19 buzzer |
| 06:30.44 | drmessano | small relay |
| 06:30.56 | jblack | Just an autodialer with a speaker. suitable for outside. |
| 06:31.02 | jblack | Like a waterproof batphone. |
| 06:31.04 | drmessano | yes, $$$$ |
| 06:31.15 | drmessano | Spend the money if you want |
| 06:31.26 | jblack | I don't want to spend a lot of money. |
| 06:31.40 | jblack | That's why I'm thinking a $6 walmart phone, with a speaker... |
| 06:32.02 | drmessano | Ok |
| 06:32.07 | jblack | Push a button, which goes to the hotwired dtmf pad. |
| 06:32.16 | drmessano | and a big red button added |
| 06:32.22 | drmessano | Wired to the recall |
| 06:32.28 | jblack | Well, take phone off hook, hitwire dtmf pad. |
| 06:32.47 | jblack | It can't be too difficult to do. |
| 06:32.53 | jblack | The trick is figuring out when the call is "over" |
| 06:33.07 | drmessano | Ok heres what you do |
| 06:33.29 | drmessano | Check the hook switch... see if they used a stupidly large switch |
| 06:33.37 | drmessano | Like, extra contacts |
| 06:33.48 | drmessano | No |
| 06:33.50 | drmessano | NM |
| 06:33.57 | drmessano | Doubt youd get a NO contact |
| 06:34.58 | drmessano | Wait |
| 06:35.00 | drmessano | I GOT IT |
| 06:35.09 | drmessano | But you need to check into Linksys dial plans |
| 06:35.32 | jblack | I bet it does. |
| 06:35.44 | drmessano | Sure you can craft a dial plan where ANY digit dials xxx |
| 06:35.50 | jblack | For some reason, it's decided to not listen to 192.168.2.97, but it's happy to talk to 192.168.2.2 |
| 06:36.10 | drmessano | ummm |
| 06:36.24 | jblack | yeah, I'm telling you. This is a quirky puppy |
| 06:37.25 | drmessano | HA |
| 06:37.31 | drmessano | I got it |
| 06:37.45 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 06:37.49 | drmessano | (<x:ext>) |
| 06:37.52 | drmessano | Thats all you need |
| 06:37.55 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
| 06:37.56 | drmessano | (<x:600>) |
| 06:38.01 | jblack | What's that? |
| 06:38.09 | drmessano | Dial plan for Linksys |
| 06:38.18 | drmessano | Replace ANY single digit with 600 |
| 06:39.14 | jblack | Yeah, I can do that on the * side too. |
| 06:39.23 | jblack | I'm wondering if it's possible to dial *no* digit. ;) |
| 06:39.38 | drmessano | Not without modding the phone |
| 06:39.52 | drmessano | I dont think the ATA will |
| 06:39.56 | drmessano | Lemme check a PAP2 |
| 06:40.18 | jblack | I bet it can. |
| 06:40.22 | jblack | wtf... |
| 06:41.06 | jblack | Hey, this puppy is lynxable. Nice |
| 06:41.29 | drmessano | Nope |
| 06:41.47 | drmessano | PAP2 wont, and SPA3102 wont |
| 06:42.00 | drmessano | Im gonna try something |
| 06:42.42 | drmessano | Ok |
| 06:42.49 | drmessano | the dialplan I gave you works for 1 digit |
| 06:45.55 | jblack | I bet there's a way to do it for off-hook. |
| 06:45.57 | drmessano | The grandstream will |
| 06:46.01 | drmessano | The Linksys wont |
| 06:49.06 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:52.25 | SwK | jblack, dial no digits? like a hotline? |
| 06:52.32 | SwK | just pick up and it sends a call? |
| 06:52.47 | mishehu | bah. |
| 06:52.52 | SwK | mishehu, |
| 06:53.04 | SwK | mishehu, hey whats your landline carrier? |
| 06:53.06 | mishehu | silicon jeezuz kryst |
| 06:53.16 | mishehu | SwK: I use globalcom for my PRI |
| 06:53.23 | mishehu | they're a chitown clec |
| 06:53.32 | mishehu | (northern illinois actually) |
| 06:53.35 | jblack | swk: Yeah |
| 06:53.53 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:53.53 | SwK | jblack, pap2 spa2XXX can do that |
| 06:53.58 | mishehu | SwK: why you ask? looking for something in chicago? |
| 06:54.02 | SwK | its an "advanced setting" |
| 06:54.02 | drmessano | PAP2 can? |
| 06:54.02 | jblack | swk: How? |
| 06:54.04 | drmessano | Where? |
| 06:54.07 | jblack | What's the setting? |
| 06:54.16 | SwK | its in there somewhere I forget and I dont have a pap2 handy |
| 06:54.31 | SwK | I've set them up that way for call boxes and stuff that just go off hook |
| 06:54.35 | drmessano | Hmm |
| 06:54.55 | jblack | That's exaactly my project. A doorbell intercom |
| 06:55.05 | SwK | jblack, it will work... |
| 06:55.23 | SwK | i have 4 pap2s on my desk hah lemme see if I can find the psu for one of them |
| 06:55.45 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 06:56.15 | drmessano | Ive been all through mine |
| 06:56.20 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 06:57.30 | SwK | hah gotta charge the phone |
| 06:58.03 | mishehu | if you'd like to make a call, please hang up and try again |
| 06:58.16 | mishehu | if you need assistance please dial the oooooperator! |
| 07:00.06 | jblack | I suppose all I really need is a ringdown generator |
| 07:00.33 | jblack | I don't really need an intercom |
| 07:00.38 | drmessano | Im not seeing it in my V1 PAP2 |
| 07:00.45 | drmessano | Maybe the PAP2T |
| 07:01.00 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 07:01.04 | SwK | trying to get into one of these pap2's |
| 07:01.14 | jblack | That's ok. I'll check home depot tomorrow |
| 07:05.09 | drmessano | Any luck, SwK |
| 07:05.45 | SwK | Dialing Plan settings from the PAP2 |
| 07:05.53 | SwK | The following implements a Hot Line phone, which automatically calls 1 212 5551234. |
| 07:06.00 | SwK | ( S0 <:12125551234> ) |
| 07:06.03 | SwK | try that |
| 07:06.13 | drmessano | Oh geez |
| 07:06.34 | SwK | i had to grab my manual |
| 07:06.38 | SwK | its been a while heh |
| 07:06.49 | jblack | That's perfect. |
| 07:06.56 | drmessano | I was sniffing around dial plans |
| 07:06.57 | SwK | you can also do warm line stuff too |
| 07:07.11 | jblack | Where does that go? |
| 07:07.12 | SwK | like |
| 07:07.13 | drmessano | Missed the S0 |
| 07:07.25 | SwK | jblack, that goes in the dialplan setting |
| 07:07.31 | SwK | ( P5 <:1000> | xxxx ) <--- this one is cool too |
| 07:07.32 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:08.04 | SwK | the P5 <:1000> means if you dont get digits within 5 seconds dial exten 1000 |
| 07:08.11 | drmessano | Nice |
| 07:08.43 | jblack | I can use that now. |
| 07:09.41 | SwK | that should work on and SPA or PAP device since they all use the same basic firmware |
| 07:09.59 | jblack | yeah |
| 07:10.00 | SwK | a PAP2 and a linksys SPA are just the same f'n stuff in different plastic cases |
| 07:10.49 | drmessano | Thats awesome.. I saw the Linksys didnt specify an off-hook dial, and I was playing with dialplans, but found a few places that said it couldnt be done.. Cool to find that out |
| 07:11.39 | jblack | brb |
| 07:11.43 | drmessano | I need to implement the 5 sec pause on the PAP2s I have |
| 07:12.05 | drmessano | "If you'd like to make a call, DO IT" |
| 07:14.10 | *** join/#asterisk jblack (n=jblack@pool-71-181-138-192.sctnpa.east.verizon.net) |
| 07:14.20 | jblack | This is weird. The spa 8k is no longer talking to my laptop |
| 07:15.15 | jblack | I can get to it from another machine on the same network, but not from _this_ machine |
| 07:17.08 | jblack | hmm. and it's ignoring ****732668. Says it's successful, but it never stops responding to icmp |
| 07:18.00 | SwK | whats 732668? |
| 07:19.18 | SwK | never mind thats reboot |
| 07:19.28 | drmessano | r e b o o t |
| 07:19.28 | drmessano | yeah |
| 07:19.40 | jblack | flip the power switch, and things are good again. |
| 07:19.40 | SwK | i looked at my dtmf pad wrong heh |
| 07:19.46 | drmessano | or http://ip/admin/reboot |
| 07:20.07 | SwK | i used to build ITSPs for a living i should know all the settings on that thing |
| 07:20.22 | drmessano | Ok, so whats the best guide on the PAP2? |
| 07:20.32 | jblack | I need to learn how to read these dialplans |
| 07:20.44 | drmessano | I know the things inside and out.. but skipped dialplans COMPLETELY |
| 07:20.45 | drmessano | Well |
| 07:20.54 | drmessano | Not 100%, but missed some things apparently |
| 07:21.06 | drmessano | Rest I guessed at and made work lol |
| 07:27.19 | jblack | Ahh, I see what S0 does. |
| 07:27.25 | jblack | That means "0 delay" |
| 07:30.57 | drmessano | yep |
| 07:31.29 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.102.159) |
| 07:32.56 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:33.53 | drmessano | jblack, I got the Linksys ATA Admin guide if you want a copy |
| 07:34.23 | SwK | i have the linksys config compilers around here somewhere |
| 07:34.42 | SwK | you can use them to autoconfig your ATAs via tftp, ftp http or whatever |
| 07:34.49 | drmessano | Cool |
| 07:35.00 | jblack | drmessano: Yeah, I hae it in pdf here. |
| 07:35.09 | SwK | (its basically the same thing vonage uses to generate the file that is downloaded to the PAP2) |
| 07:35.12 | jblack | Normally, I don't need the ata for it's dialplan stuff. That's what * is for. |
| 07:35.38 | drmessano | SwK, I will trade you my sister for them |
| 07:35.40 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 07:35.58 | SwK | drmessano, how old is she and what she look like? |
| 07:36.02 | SwK | :P |
| 07:36.14 | jblack | So I figure, except for the doorbell hotline, that i could set my dialplan to X. |
| 07:36.15 | drmessano | She doesnt look anything like me, shes good looking.. 28 |
| 07:36.28 | SwK | pic url? |
| 07:36.29 | drmessano | lol |
| 07:36.30 | SwK | hah |
| 07:36.59 | drmessano | Shes got a big greasy hippie boyfriend though.. he likes to smash bricks on his head as a hobby |
| 07:37.06 | drmessano | But im sure you two will get along |
| 07:37.17 | drmessano | lol |
| 07:37.41 | drmessano | I said I would trade.. Shipping and pickup arrangements are ALL YOU |
| 07:37.56 | SwK | hah |
| 07:38.11 | SwK | FOB location? |
| 07:39.02 | drmessano | Knoxville somewhere |
| 07:41.16 | SwK | thats not too far |
| 07:41.26 | SwK | i'm in huntsville |
| 07:42.13 | drmessano | Cool.. I'll make sure she cleans up before shipping..like you would an eBay sale |
| 07:43.06 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 07:43.20 | SwK | hah |
| 07:43.34 | dacs | hi guys |
| 07:43.40 | SwK | I dont by anything sight unseen tho :P |
| 07:43.44 | dacs | drmessano: you got the email man |
| 07:43.54 | drmessano | I didnt, dacs |
| 07:44.38 | drmessano | Well, SwK, if you run across the provisioning apps, keep me in mind... I love messing with PAP2s and thats right up my alley.. I have 40 or so of them |
| 07:44.46 | dacs | drmessano: its 40 MB maybe thats why |
| 07:44.58 | drmessano | dacs: PDF? |
| 07:45.04 | SwK | drmessano, yeah i'm trying to remember where I stashed them and where I got them from |
| 07:45.11 | SwK | think i got them from the guys at voipsupply |
| 07:45.15 | drmessano | ah |
| 07:45.20 | dacs | drmessano: yes it is? |
| 07:45.28 | drmessano | dacs: zip it up |
| 07:49.15 | drmessano | http://spc.pifiu.com/ |
| 07:50.01 | jblack | Actually, I think I wans S:2,xx. |
| 07:51.03 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 07:52.43 | *** join/#asterisk af_ (n=getsmart@88-149-240-167.dynamic.ngi.it) |
| 07:56.03 | drmessano | SwK are they GUI apps? |
| 08:00.19 | jblack | Yeah, it looks like S:2(xx.) is all I need. |
| 08:04.20 | tzafrir | PDFs are usually reasonably compessed when done well |
| 08:05.56 | drmessano | Yep |
| 08:08.07 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:14.15 | jblack | I would think that ([#*x].) would allow anything though, but it blocks *86 |
| 08:16.48 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:24.56 | *** join/#asterisk kbast (n=KB123@58-65-160-140.nayatel.pk) |
| 08:25.09 | kbast | Hi guys |
| 08:25.16 | jblack | Ohh, I can fax! |
| 08:25.47 | kbast | I have question about Asterisknow... Can we use it for hosting multiple users |
| 08:26.02 | jblack | kbast: Did you see the /topic yet? |
| 08:27.11 | kbast | ohh... can you tell asterisknow channel name ? |
| 08:28.26 | kbast | got it thanks any way! |
| 08:29.39 | *** join/#asterisk oej (n=olle@213.115.215.130) |
| 08:30.59 | *** join/#asterisk Al_WinKiller (i=Alex_Win@83.139.12.190) |
| 08:32.31 | Al_WinKiller | hi guys I have installed radiusclient-ng, how do I know that it is starts , while asterisk stars |
| 08:32.32 | Al_WinKiller | ? |
| 08:41.50 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:45.27 | kbast | you need to configure radiusclient-ng.conf |
| 08:45.35 | kbast | and for asterisk cdr_radius.conf |
| 08:46.11 | *** join/#asterisk dreamydevon10 (n=dreamyde@c-71-198-211-71.hsd1.ca.comcast.net) |
| 08:46.27 | Al_WinKiller | ok, let me see |
| 08:46.29 | dreamydevon10 | hello is there anyone active in this room? |
| 08:47.14 | dreamydevon10 | I have a question about inbound SIP lines |
| 08:48.24 | Al_WinKiller | but, listen,, how do I know, that radiusclient is loaded ? |
| 08:49.05 | tzafrir | dreamydevon10, noone |
| 08:49.17 | nixguy | dreamydevon10: dont ask just ask |
| 08:49.27 | Al_WinKiller | which port does it use ? |
| 08:49.32 | tzafrir | Al_WinKiller, I really have no idea, but maybe you should ask the right questions |
| 08:50.12 | tzafrir | "what port does XXX use" is usually rather simple to answer |
| 08:50.22 | tzafrir | also: a client listens on a port? |
| 08:50.48 | tzafrir | To check what programs listen on local ports: netstat -lnutp |
| 08:50.49 | Al_WinKiller | the right question is "how do I know that radiusclient-ng is active ? or loaded ? " |
| 08:51.58 | dreamydevon10 | ok so the issue is I am using Vitelity inbound with AsteriskNOW, and I am able to get calls coming in fine from three different DID that i have with them, however the problem that I have is i cant seem to get * to distinguish between the different lines, when i look at the CLI it shows this for each DID |
| 08:52.27 | dreamydevon10 | Executing [415963300@DID_trunk_9:1] |
| 08:52.36 | tzafrir | Al_WinKiller, if it comes with a decent package and it is a daemon, it should come with an init.d script |
| 08:53.13 | dreamydevon10 | every call comes in to the same trunk, even tho the system has them set to different trunks for each of the three numbers |
| 08:54.23 | Al_WinKiller | ok trafrir ) thnx ) |
| 08:55.11 | mvanbaak | I think configuring radiusclient-ng.conf and cdr_radius.conf is enough |
| 08:55.30 | mvanbaak | no daemon to run. It will allow asterisk to connect to your radius server |
| 08:55.42 | dreamydevon10 | anyone have any thoughts? |
| 08:56.34 | *** join/#asterisk Schumie (i=SteveWri@87.127.1.8) |
| 08:57.08 | Al_WinKiller | mvanbaak look what I want to do http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html#AEN34 |
| 08:57.13 | Al_WinKiller | not for cdr |
| 08:58.52 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:00.06 | dreamydevon10 | I can give some more information I just dont know what else would be useful in determining the right settings I need to use, it was working fine earlier then now its not |
| 09:01.22 | mvanbaak | Al_WinKiller: so why dont you follow that page ? |
| 09:01.33 | mvanbaak | it gives you a step by step plan to follow |
| 09:01.57 | dreamydevon10 | anyone? |
| 09:02.07 | *** join/#asterisk tiav (n=tiav@inv75-3-82-241-117-16.fbx.proxad.net) |
| 09:03.00 | Al_WinKiller | I do, but it doesn't work :) |
| 09:03.09 | Al_WinKiller | but,, it will I am sure ) |
| 09:04.00 | dreamydevon10 | nixguy any thoughts? |
| 09:07.14 | jochien1 | !mISDN |
| 09:07.49 | tzafrir | dreamydevon10, hint: repeating the same question over and over again is impolite |
| 09:08.07 | dreamydevon10 | well being ignored isnt so polite either |
| 09:08.20 | Stefan1979 | uhoh |
| 09:08.51 | tzafrir | dreamydevon10, maybe noone has a good answer |
| 09:09.17 | dreamydevon10 | is there a good paid forum for asterisk? |
| 09:10.11 | tzafrir | There are a bunch of paid support guys hanging here |
| 09:10.13 | tzafrir | (not me) |
| 09:10.40 | tzafrir | dreamydevon10, try providing useful details |
| 09:11.14 | dreamydevon10 | well i dont want to get too detailed if there is noone here that knows what I am talking about |
| 09:13.39 | nixguy | dreamydevon10: sorry nothing i can answer right now, im still an asterisk beginner :) it would have been more linux-generic i could have helped out more :| |
| 09:14.13 | dreamydevon10 | alright well thanks anyways, im going to go looking on some of the more static forums to see what I can find |
| 09:16.47 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:18.28 | yang | If I had to choose the right BRI ISDN card for asterisk 1.2. version, what would you suggest...I was looking at Sangoma AFT A500 , and Junghanns DUO BRI which has HFC-4S chip, and there are also some Digium cards... |
| 09:25.35 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 09:26.26 | *** join/#asterisk MrMister2 (n=mrmister@195-23-220-61.net.novis.pt) |
| 09:27.23 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-dcd94a4700ec3e39) |
| 09:30.59 | tzafrir | yang, why "for asterisk 1.2"? |
| 09:31.11 | tzafrir | Is this an existing installation? |
| 09:31.36 | tzafrir | Generally everybody supports (drivers-wise) asterisk 1.2 as well as 1.4 |
| 09:31.58 | yang | tzafrir: the debian lenny asterisk |
| 09:32.04 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:32.29 | yang | I was refered to "wanpipe" but I cannot find such package |
| 09:32.29 | tzafrir | Lenny will finally have asterisk 1.4 in a day or two, unless there are big surprises |
| 09:32.42 | tzafrir | There was one in potato |
| 09:32.51 | tzafrir | And removed later. |
| 09:33.03 | alrs | yang: the Sangoma drivers have binary blobs in them, so they probably won't show up packaged for many distros |
| 09:33.21 | tzafrir | that's the BRI ones, right |
| 09:33.35 | yang | alrs: and people told me, that the Digium cards have zaptel issues |
| 09:33.37 | tzafrir | though they should basically work with bristuff as well |
| 09:33.45 | tzafrir | which is in Debian |
| 09:34.27 | yang | alrs: but probably its better to get Sangoma than some "preety unknown" brand Junghanns with HFC4S chip? |
| 09:34.31 | alrs | yang: the old Digium cards have a bad reputation, but the newer ones are supposed to be better. Since I'm in the US I've never touched any of the BRI stuff. |
| 09:34.47 | JT | alrs: they've only released one bri card? |
| 09:35.07 | alrs | JT: that sounds right, I know little of their BRI stuff |
| 09:35.39 | JT | the digium bri card only uses misdn, which is poop |
| 09:35.52 | alrs | JT: what is the popular BRI card in Europe? |
| 09:35.59 | yang | You know I hear different stories here...some favorize Digium, some Sangoma, really hard to decide what to buy |
| 09:36.01 | JT | i have no idea |
| 09:36.09 | JT | but cologne based cards are the most common |
| 09:36.19 | JT | favorize...? |
| 09:36.39 | alrs | I like to stick with Debian packages, so I'd probably go with the cologne-based, myself. |
| 09:37.06 | JT | sticking with packages is a silly critereon to pay any importance to in the asterisk world |
| 09:37.09 | alrs | Sangoma tech support is excellent, but their drivers are semi-proprietary and exist outside of deb |
| 09:37.18 | *** join/#asterisk Jam0r (i=Jamie@87.127.190.82) |
| 09:37.46 | alrs | JT: That's how I do. |
| 09:37.53 | JT | crazy |
| 09:37.59 | JT | you can always make your own packages |
| 09:38.01 | yang | So you say a cologne-Chip Junghanns HFC-4S could be a better choice? |
| 09:38.02 | JT | to keep everything nice |
| 09:38.25 | JT | pre made packages are not the be all and end all, especially with asterisk |
| 09:38.28 | alrs | JT: I'm running Asterisk 1.4.17 in Debian unstable in a Xen domu |
| 09:38.43 | JT | uhuh |
| 09:38.47 | alrs | so I can run my other services on other domUs with stable |
| 09:39.46 | alrs | I ran Slackware for four years in the '90s, I've come to realize my limitations when it comes to keeping up with security alerts for every application |
| 09:39.55 | JT | sure |
| 09:40.30 | nixguy | alrs: does that work for you? |
| 09:40.39 | nixguy | curious about asterisk and virtualization |
| 09:40.46 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:40.48 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 09:40.50 | alrs | mrtelnet:~# lspci |
| 09:40.51 | alrs | comprookie2000:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 09:40.51 | alrs | mrtelnet:~# |
| 09:41.16 | alrs | that was bizarre |
| 09:41.22 | alrs | tel |
| 09:41.31 | alrs | a cut and paste nightmare |
| 09:50.23 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 09:54.54 | *** join/#asterisk DaPrivateer (n=matt7229@66.92.79.218) |
| 09:55.34 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:56.17 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 09:58.47 | FlatFoot | morning all |
| 09:58.54 | mvanbaak | hey |
| 09:59.09 | dacs | morning |
| 09:59.39 | FlatFoot | anyone know where i can get the latest FreeBSD ports version ? i have been trying to get 1.4.x but all i can find is 1.2.x |
| 09:59.47 | FlatFoot | cvsup BTW |
| 10:10.22 | jblack | I feel like I'm so close, yet so far, with asterisk-app-fax |
| 10:25.59 | TJNII | Heh. I just found a problem. In a meetme chatroom if one phone sends the DTMF in-audio it can be picked up by an ATA which then thinks it came from its phone |
| 10:26.29 | TJNII | Oh well, I'll fiddle with it in the morning. It is waaaaay past my bedtime |
| 10:28.33 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:30.12 | jblack | I actually got part of a tiff. |
| 10:34.46 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 10:34.56 | *** part/#asterisk Porks (n=Porks@201.62.79.12) |
| 10:39.26 | MrMister2 | Hi. Whenever I do a attended transfer and the other extension doesn't pick up the phone, it only rings 3 or 4 times before I get the call back. Any ideas on how to increase this timeout? I _think_ it has to do with this line on the log: "res_features.c: We exceeded our AT-timeout" |
| 10:44.49 | *** join/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 10:44.56 | *** join/#asterisk Winkie (n=urmom@87-194-109-4.bethere.co.uk) |
| 10:47.45 | *** join/#asterisk Makenshi (n=makenshi@makenshi.at.furry.be) |
| 10:56.04 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 10:56.34 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 10:56.35 | Chris-NB | hi |
| 10:56.49 | Chris-NB | ist it possible to use a Sangoma card with chan_capi and not with chan_zap ? |
| 10:58.12 | jblack | Oh YEAH. I got faxes! |
| 10:58.27 | puppet | jblack: mailed to you? |
| 10:58.56 | jblack | faxed to my ipkall number, through asterisk, to email. |
| 10:59.01 | jblack | The downside is that they're sideways. |
| 10:59.03 | puppet | what script you use? |
| 10:59.13 | jblack | I'm using asterisk-app-fax |
| 11:00.02 | puppet | to mail it? oh |
| 11:00.49 | jblack | I lied. They're not sideways. |
| 11:00.57 | jblack | I have a fax number!! |
| 11:02.11 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 11:06.16 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 11:08.05 | morlac | Chris-NB: not sure about chan_capi, but if you dont want to use chan_zap, you can use chan_womeera |
| 11:10.16 | *** join/#asterisk orn (n=orn@85.197.193.24) |
| 11:10.29 | orn | Anyone had this problem: [Jan 14 11:05:38] WARNING[2585]: channel.c:718 ast_best_codec: Don't know any of 0x0 formats ? |
| 11:11.51 | jblack | Sounds to me like you couldn't agree on a codec. Did you disallow all, and only allow 1 or 2? |
| 11:12.03 | Chris-NB | morlac, my problem is, that I've to use Q.SIG as protocol with a Siemens HiPath 4000 ... and there are .... many restrictions due to the lack of Q.SIG Implementation in Asterisk |
| 11:12.04 | orn | (I'm having a codec negotiation problem, except that as far as I can tell from the SIP dialog there is codec agreement) |
| 11:12.18 | Chris-NB | and I heared, that Q.SIG is better implemented in chan_capi ? |
| 11:12.19 | mosty | orn, which codec? |
| 11:12.23 | orn | alaw & ulaw |
| 11:12.49 | orn | disallow = all and allow = alaw allow = ulaw |
| 11:12.52 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 11:13.33 | orn | here is the capabilities that the * sends to the SIP phone before i see the error message above in verbose mode: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x1d4c (ulaw|alaw|g726|slin|g729|ilbc|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) |
| 11:13.39 | dacs | Alcatel Lucent support sucks |
| 11:13.56 | orn | so, capabilities seem to be 0xc, and yet * complains about 0x0 |
| 11:14.25 | orn | i'm wondering if this is a bug in the asterisk appliance, because I don't have this problem on custom asterisk compiled from source on a different machine |
| 11:14.43 | orn | it seems to me that the only codec that is working on the appliance is GSM |
| 11:20.47 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:21.21 | morlac | Chris-NB: I have no idea about QSig as I never used it...nor did I use chan_capi....but to my knowledge, Sangoma patches chan_zap when you enable HW accel during sangoma installation...Why dont you try sangoma support? my experience whith them was excellent |
| 11:22.53 | Chris-NB | morlac, okay, thanks for the hint. I'll do that. |
| 11:24.59 | morlac | Chris-NB: and to my knowledge, chan_capi is meant for ISDN cards, and was written for Klaus Peter Junghanns, check http://www.voip-info.org/wiki/index.php?page=Asterisk+How+to+connect+with+CAPI |
| 11:25.14 | Chris-NB | morlac, k, thanks! |
| 11:25.22 | morlac | ur welcome |
| 11:28.10 | jblack | Does anyone know if the sip fax detection docs at http://www.voip-info.org/wiki/view/Asterisk+and+faxes and http://www.voip-info.org/wiki-Asterisk+fax are still current, in that NVBackgroundDetect is still needed? |
| 11:28.36 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:29.02 | badcfe | i got some sound files in format alaw here, but file says its "RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz". how do i extract just the alaw raw data in there? |
| 11:29.52 | mosty | badcfe, use sox? |
| 11:29.58 | badcfe | by the way, is there any howto of how i can convert from 16bit wav 44mhz sound files into alaw files for asterisk? |
| 11:30.19 | mosty | badcfe, again, use sox |
| 11:30.31 | badcfe | mosty: that sox may remove that certain RIFF header for me? |
| 11:30.42 | badcfe | mosty: thanks. ill look into that sox. |
| 11:30.46 | mosty | sox can convert between lots of different formats |
| 11:31.06 | mosty | another free software option is audacity (which has a gui) |
| 11:31.28 | badcfe | mosty: if file extracts all that RIFF stuff, i guess sox probably understands and handles it yes. |
| 11:31.48 | badcfe | mosty: i note that audacity too in case, thanks. |
| 11:32.31 | badcfe | mosty: tho i generally prefer command line tools (have a bunch of files i want to pass thu it) .. |
| 11:32.41 | RoyK | badcfe: sox somefile44korsomething.wav -c 1 -w -r 8000 outfile.wav |
| 11:32.46 | mosty | then sox is definitely what you want |
| 11:32.57 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 11:33.23 | RoyK | badcfe: just try that - works for me (tm) |
| 11:43.05 | badcfe | RoyK: thank you. but even after applying options as i should try to according to the sox man page i cant get sox to read my file of type "RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz". it tells me sox soxio: Failed reading `4.wav': unknown file type `auto' ... |
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| 11:44.33 | badcfe | hmm, maybe my files may already be presentable for asterisk? (even if file reports raw data on the existing asterisk sound files) |
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| 12:00.01 | mosty | you might need some -t option to tell sox what the format is |
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| 12:03.55 | dacs | DarKnesS_WolF: you here |
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| 12:23.11 | RoyK | badcfe: paste the command line used, please |
| 12:23.36 | RoyK | Hi all. Is it possible to somehow use app_chanspy or similar to spy on a whole bridge? |
| 12:25.42 | *** join/#asterisk Daejeo (n=chatzill@211.211.234.81) |
| 12:27.47 | Daejeo | can anyone recommend VOIP provider for asterisk ? I am looking for US/CANADA unlimited calls- home |
| 12:28.58 | tzanger | Daejeo: I like unlimitel very much for canadian termination and dids, their international rates are pretty good too. excellent quality and customer service. nufone's another one that gets my vote |
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| 12:43.45 | Daejeo | tzange: i was researching nufone's i could not find any plan |
| 12:44.22 | Daejeo | tzanger: i was researching nufone's i could not find any plan |
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| 12:47.18 | tzanger | Daejeo: I just contact jerjer directly |
| 12:50.05 | awk | hrm, http://www.pastebin.ca/854609 |
| 12:50.10 | awk | big issue about buffer space, any ideas... |
| 12:50.33 | awk | first paste was from /var/log/messages second was from asterisk messages |
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| 12:57.44 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 12:57.54 | ZaVoid | morning boys |
| 12:58.01 | ZaVoid | anyone else on the east coast? |
| 12:58.06 | ZaVoid | what a BLIZZARD we had last night huh? |
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| 12:59.24 | Rawplayer | is it already possible to authenticate against ldap? |
| 13:04.23 | tzafrir | awk, http://tldp.org/LDP/LG/issue93/TWDT.html#tips.5 |
| 13:04.54 | tzafrir | first hit on search.yahoo.com (why should google get all the credit?) |
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| 13:07.17 | tzanger | tzafrir: because yahoo sucks? :-) |
| 13:08.11 | RoyK | ~book |
| 13:08.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
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| 13:10.26 | tzafrir | tzanger, no, not really |
| 13:10.51 | tzafrir | And I don't like the fact that everybody uses google. |
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| 13:11.24 | tzafrir | Right now they are nice, but I don't really like them as a single point of failure |
| 13:13.19 | tzafrir | That said, Yahoo are not among my favorites, and if anybody can suggest me an interesting alternative, please do |
| 13:13.35 | tzanger | I thought there was some new one that was google-esque |
| 13:13.39 | tzanger | there were a few I think |
| 13:13.58 | tzafrir | altavista is a front to yahoo |
| 13:14.09 | ronr | tzafrir: we'll need someone to fight after we beat microsoft, so we're moving google into position now :) |
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| 13:15.13 | ZaVoid | whatcha trying to do tz? |
| 13:15.23 | tzafrir | which tz? |
| 13:15.28 | ZaVoid | you :) |
| 13:15.30 | ZaVoid | tzafrir: |
| 13:15.36 | ZaVoid | lol |
| 13:15.44 | ZaVoid | didn;'t even realize both of you had tz.. damn it |
| 13:16.09 | tzafrir | I stole his prefix :-) |
| 13:16.29 | tzafrir | See alternatives to google |
| 13:16.34 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:17.02 | tzafrir | One really bad thing about yahoo: the links they give in a results page are not direct links |
| 13:17.33 | tzafrir | Rather: indirect links to some internal script of theirs. |
| 13:18.37 | tzafrir | This is why (a) it is slower to follow their search results and (b) I rarely quote their links |
| 13:19.18 | mvanbaak | looks like digium download server ;) |
| 13:22.39 | tzanger | heh |
| 13:22.44 | tzanger | I hate redirects like that |
| 13:22.49 | Makenshi | The problem is, there aren't any alternatives as comprehensive and fast as Google right now (available to the public) :-/ |
| 13:23.03 | tzanger | generate a static page with the right link for fuck sakes |
| 13:23.05 | tzanger | but that's just me |
| 13:23.13 | Aurs | I always get the wget from digium wrong :) |
| 13:23.46 | tzafrir | search.yahoo.com is quite nice |
| 13:24.06 | ZaVoid | bah don't care for yahoo search |
| 13:24.21 | ZaVoid | google i still like with all 5000000000 results lol |
| 13:25.04 | tzafrir | All the 5000000000 don't really matter. The result I really wanted is what matters. |
| 13:25.35 | tzafrir | Fast comprehensive search engines is what we had in 1998. Many of them |
| 13:26.50 | mikkel | Is it possible to have a ISDN (8 lines) and have for analog phones. What Wildcard should I buy. I need a simple setup where 4 phones separately can call out and there could be call waiting. Have could this be done ? |
| 13:27.12 | mikkel | .. four analog phones .. |
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| 13:30.28 | mosty | mikkel, isdn is digital, you can't connect analogue phones directly to isdn |
| 13:30.51 | awk | tzafrir, i fixed it.. but cant exactly pin point the issue, either to many entries in my arp table..I upped the threshhold on gc_thresh1 to gc_thresh3 values 256 , 512, 1024 ... but also i see who ever set this network up has 2 192.168 networks.. 1 on nic 1 and 1 on nic 2 and subnet mask of 255.255.0.0 and 255.255.255.0 and i think the issue is doing routing decisions with overlapping networks. |
| 13:31.03 | awk | well anyway temp fix, its working.. i'll resolve this network issue now too |
| 13:31.12 | mikkel | mosty: Was hoping that you could connect the line to a Wildcard and it could convert to analog. |
| 13:31.39 | [TK]D-Fender | mikkel: taking in LINES, and letting you plug in PHONES are two completely different thigs. |
| 13:31.39 | mosty | mikkel, you need an isdn card in that case |
| 13:32.20 | tzafrir | mikkel, 8 ISDN lines or 4 ISDN lines (you can have up to 2 calls per line) |
| 13:32.23 | [TK]D-Fender | mikkel: If you wish to do both, you'll need some SIDN interface card(s), and some other means of connecting analog phones (ATA's are usually the cheapest and best) |
| 13:32.42 | mikkel | tzafrir: Then I just need 4 ISDN. |
| 13:32.52 | tzafrir | If you have ISDN (digital) why convert it to analog? |
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| 13:33.31 | [TK]D-Fender | tzafrir : I'm thinking he just wants ISDN in, and analog phones for handsets |
| 13:33.37 | mikkel | tzafrir: Have not bought anything yet. Just that analog phones would lower the cost (I assume) |
| 13:33.51 | mikkel | [TK]D-Fender: Yes like that. |
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| 13:34.30 | [TK]D-Fender | mikkel: Well I can't advise on which ISND cards are best. How many analog phones do yuo want to have connected to *? |
| 13:37.11 | mikkel | I need 4 phone (analog or digital) and 8 lines in (ISDN 4) and people could be on hold for the Secretary and the doctors could call on the other phones ? |
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| 13:39.22 | [TK]D-Fender | mikkel: If you want functionality like MoH for your phones it'd be advisable to use SIP hard phones. |
| 13:40.21 | mikkel | [TK]D-Fender: I could use the ELMEG IP 290, that would be fine. |
| 13:41.45 | [TK]D-Fender | mikkel: I don't know that model and am weary of these cheap unknown imports (Elmeg just rebrands stuff). In your area I suggest Linksys might be a better choice. |
| 13:44.54 | mikkel | [TK]D-Fender: That look fine, $129 that's fine. So If I buy 4 of the linksys IP phones and need 8 simultaneous lines (ISDN 4) what card should I buy ? Would the Digium TE220 be fine ? |
| 13:45.45 | [TK]D-Fender | mikkel: Again I have no personal experience with ISDN BRI cards, others here would be better to advise you on those. |
| 13:45.51 | [TK]D-Fender | tzafrir : ? |
| 13:45.57 | tzafrir | mikkel, the TE220 is a PRI (and dual span) |
| 13:46.11 | J4zen | i'd go for a BRI from Junghans |
| 13:46.28 | J4zen | have had some good expierence with it, although limited. |
| 13:46.43 | mosty | i have a wierd problem with sip->sip->pri calls with asterisk 1.4.17, the call works fine using one particular sip account, but not with another identical (besides username/fromuser settings). SIP config's here: http://pastebin.com/m7635f0a4 anyone know what might be wrong here? |
| 13:46.43 | mikkel | tzafrir: I have no idea what the means. Just that it was digital and assumed that's what I need for ISDN |
| 13:46.55 | tzafrir | Well, I'm of course not impartial as for the selection of a BRI card :-) |
| 13:47.01 | mosty | mikkel, i'd get the sangoma BRI card |
| 13:47.14 | J4zen | mikkel: read up on voipinfo about the topic :) |
| 13:47.38 | mikkel | tzafrir: So what is your impartial advice ? |
| 13:48.11 | mosty | mikkel, does your telco offer BRI or fractional PRI? |
| 13:48.11 | tzafrir | mikkel, you already have 4 BRI lines, right? |
| 13:49.06 | tzafrir | 8 is probably marginal for being cost-effective to fractional-PRI |
| 13:49.19 | tzafrir | but I figure it depends on the telco |
| 13:50.44 | mikkel | mosty: I don't know what that means BRI fractional PRI (Sorry very new to this) |
| 13:50.56 | tzafrir | ~bri |
| 13:50.57 | jbot | bri is, like, the Basic Rate Interface , an ISDN access interface type composed of two B-channels each at 64 kbps and one D-channel at 16kbps (2B+D). |
| 13:51.00 | tzafrir | ~pri |
| 13:51.01 | jbot | pri is probably Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 13:51.50 | tzafrir | mikkel, and if jbot's not arround to answer you, also try looking for technical terms in either wikipedia or voip-info.org |
| 13:52.35 | mikkel | tzafrir: Thanks. I know, just a lot of questions :) |
| 13:52.48 | mosty | mikkel, do you already have these ISDN lines installed? |
| 13:53.02 | mikkel | Really just need a inexpensive setup. |
| 13:53.36 | mikkel | mosty: I have not bought anything, yet. |
| 13:54.37 | mikkel | 4 or 8 lines in and 4 IP phones would do it. I'm in denmark (That is Europe) and would use a telco called TDC (If that help anyone) |
| 13:55.14 | mosty | mikkel, first of all you will have to ask your telco what 4 BRI lines will cost (8 simultaneous calls), and what a fractional PRI with 8 channels will cost. then once you've decided on BRI vs PRI, choose the right kind of ISDN card |
| 13:55.43 | puppet | why not just run pure VoIP? |
| 13:56.45 | mikkel | puppet: This is for medical clinics, a lot of older people need to be able to call. So I need normal phone service. |
| 13:56.58 | puppet | mikkel: ehm? |
| 13:57.01 | [TK]D-Fender | mikkel: what is "normal phone service"? |
| 13:57.04 | puppet | mikkel: and where does VoIP not fit in there? |
| 13:57.19 | [TK]D-Fender | mikkel: I think you are misunderstanding VoIP... |
| 13:57.37 | [TK]D-Fender | mikkel: that is jsut a WAY to place/receive calls. |
| 13:57.40 | [TK]D-Fender | ~itsp |
| 13:57.40 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 13:57.42 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 13:57.43 | nixguy | puppet: im actually curious about that also a sip trunk, dont you risk getting reponse time issues? |
| 13:57.58 | nixguy | since its IP |
| 13:58.14 | puppet | nixguy: most of Skåne (a part of sweden) are running VoIP on all the hospitals |
| 13:58.17 | puppet | run by Siemens |
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| 13:58.35 | [TK]D-Fender | nixguy: Yes, things can "happen" if your traffic goes over the public internet of course |
| 13:58.50 | puppet | And Siemens run alot of SIP |
| 13:59.18 | mikkel | I know what VoIP is, just thought that he meant that all calls would come over the internet. |
| 13:59.30 | mosty | [TK]D-Fender, i'm stuck, can you look over this setup and spot anything wrong? http://pastebin.com/m4fa93f38 i have two near identical sip user/peer pairs, and only one of them works. i can't figure out what the difference is that causes one to fail |
| 13:59.31 | [TK]D-Fender | puppet: And please separate the use of VoIP inside a building vs using it to terminate to an ITSP |
| 13:59.38 | puppet | mikkel: well it will? |
| 13:59.55 | BrokenNoze | Hi all, having problems installing my new BRI sangoma card. after a wanrouter hxprobe I just get a "FATAL: Error inserting wanpipe..." message any ideas? |
| 13:59.56 | dacs | [TK]D-Fender: drmessano help me over the weekend and now i have a running * :) |
| 14:00.11 | mosty | BrokenNoze, pastebin the entire error message |
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| 14:00.32 | mikkel | puppet: So how will a older person only having a landline phone (never even owned a computer) call in ? |
| 14:00.49 | puppet | mikkel: to a number? just like normal |
| 14:00.51 | [TK]D-Fender | mosty: not enought info in your pastebin |
| 14:01.02 | [TK]D-Fender | mosty: SIP debug, higher verbose, etc.... |
| 14:01.13 | [TK]D-Fender | mosty: And description of what exactly is on each end |
| 14:01.32 | [TK]D-Fender | mikkel: this has nothing to do with how calls come in <- |
| 14:01.41 | mikkel | puppet: So it is a service I buy at a company and they forward all connections to with VoIP ? |
| 14:01.54 | puppet | mikkel: kinda yes, but you need a stable internet connection for it to be ok |
| 14:02.01 | BrokenNoze | mosty: http://pastebin.com/m4d0f5c16 |
| 14:02.07 | [TK]D-Fender | mikkel: they dial your phone # and however the call gets to your PBX is irrelevant. |
| 14:02.39 | mosty | [TK]D-Fender, the sip side of things works, but the sip server can only terminate calls to pri using one of the two accounts |
| 14:02.40 | [TK]D-Fender | BrokenNoze: First guess : your kernel just got upgraded |
| 14:02.52 | [TK]D-Fender | BrokenNoze: And you need to recompile your wanpipe modules because of this |
| 14:03.01 | BrokenNoze | recompiled twice |
| 14:03.03 | mosty | BrokenNoze, did you look in dmesg? |
| 14:03.08 | mikkel | puppet: [TK]D-Fender: I see. Will check the prices for that asweel (The internet connection is very stable, so it is an option) |
| 14:03.30 | BrokenNoze | no, but my zaptel drivers ain't loaded either modprobe zaptel fails |
| 14:03.34 | [TK]D-Fender | mikkel: take some time and calculate out all your options. |
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| 14:04.27 | [TK]D-Fender | BrokenNoze: Prehaps you could pastebin your dmesg so tohers can continue to help you on this. |
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| 14:05.09 | BrokenNoze | OK, if i can find dmesg :) |
| 14:05.21 | puppet | just write dmesg |
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| 14:07.36 | BrokenNoze | OK, http://pastebin.com/m50a40c6a |
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| 14:08.45 | BrokenNoze | looks to me like my zaptel install is rubbished? |
| 14:09.07 | [TK]D-Fender | BrokenNoze: Entirely possible. |
| 14:09.22 | [TK]D-Fender | BrokenNoze: I'd suggest starting the process from scratch tarball extractions. |
| 14:09.42 | BrokenNoze | i've recompiled of zaptel, make clean, make, make install again |
| 14:10.04 | BrokenNoze | then rerun the sangoma install. same problem |
| 14:11.04 | [TK]D-Fender | BrokenNoze: Again I suggest trashing the original folders and extracting all over again |
| 14:11.18 | BrokenNoze | ok, ill give that a go |
| 14:11.23 | BrokenNoze | thanks |
| 14:11.41 | BrokenNoze | cd .. |
| 14:13.47 | mosty | [TK]D-Fender, it appears that the PRI isn't happy when the sip client has a digit in its username, so i figure it's a callerid issue |
| 14:14.55 | [TK]D-Fender | mosty: Yes, that is entirely liekly. |
| 14:15.54 | mosty | the unchecked sip callerid is "leaking" into the pri in some sense |
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| 14:21.16 | *** mode/#asterisk [+o d3wayne] by irc.freenode.net |
| 14:21.16 | *** part/#asterisk myiagy (n=Jose@189.34.24.93) |
| 14:22.13 | [TK]D-Fender | *b00m* |
| 14:23.39 | mikkel | [TK]D-Fender, puppet, mosty: Thanks for the help. I will check some more on the net, now that I know a little bit more. |
| 14:23.46 | puppet | mikkel: no worries :) |
| 14:24.34 | *** join/#asterisk Greek-Boy (n=grb@41.221.58.2) |
| 14:25.08 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 14:27.09 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
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| 14:36.33 | *** join/#asterisk riddlebox (n=wmirc_us@70-8-187-119.area4.spcsdns.net) |
| 14:38.43 | l0verb0y | does anyone know any good fax cards |
| 14:38.46 | riddlebox | has anyone put asterisk onto a partner system as a voicemail? |
| 14:39.13 | Daejeo | [TK]D-Fender: happy new year |
| 14:39.18 | [TK]D-Fender | riddlebox: Just ask your direct end question |
| 14:39.30 | [TK]D-Fender | l0verb0y: www.hylafax.org |
| 14:39.44 | l0verb0y | thanks |
| 14:39.57 | Greek-Boy | [TK]D-Fender what was that GUI that you recommended to me the last for asterisk? |
| 14:40.01 | Greek-Boy | you said it was the best one around |
| 14:40.19 | mosty | l0verb0y, eicon diva |
| 14:40.29 | [TK]D-Fender | Greek-Boy: ScopServ |
| 14:40.33 | riddlebox | when a call comes in from a partner extension does asterisk see the caller id? |
| 14:42.17 | [TK]D-Fender | riddlebox: what is a "partner" extension, what is the call coming in via, and what have you done to verify the CID? |
| 14:42.40 | [TK]D-Fender | riddlebox: And some backup might be nice... |
| 14:42.51 | [TK]D-Fender | riddlebox: Can you tell me whats wrong with my car? |
| 14:44.24 | riddlebox | tk i havent tried it yet i am wondering has to find out before i tell my boss that it can be done. I guess I wil snag a processor from the office and try doing it |
| 14:45.13 | riddlebox | it would be coming in on an analog station |
| 14:45.32 | [TK]D-Fender | riddlebox: that tells virtually nothing. |
| 14:48.22 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:48.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:48.36 | riddlebox | nm I will jus try it later |
| 14:48.41 | *** join/#asterisk bmg505 (n=leon@196.209.181.67) |
| 14:49.12 | riddlebox | i was just wondering if anyone has done it before |
| 14:50.43 | [TK]D-Fender | riddlebox: You have no details and we can't help you without them. |
| 14:50.50 | ManxPower | riddlebox: it would depend on the type of connection between the two systems and how the "partner" system is configured. |
| 14:51.28 | ManxPower | Our Nortel box, for example can neither receive nor send CLID information over ports configured as "trunk" |
| 14:51.33 | mosty | i have a call flow that looks like this: sipclient -> asterisk1 -sip-> asterisk2 ->zap. how can i preserve the the sipclient's callerid all the way through? |
| 14:51.46 | ManxPower | mosty: it does that by default. |
| 14:52.10 | ManxPower | Of course once the call hits the PSTN, it's the carrier's issue. |
| 14:52.21 | mosty | ManxPower, in my case, asterisk2 sees the callerid as whatever is set on asterisk1 |
| 14:52.24 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 14:52.45 | ManxPower | mosty: That sounds correct to me. |
| 14:53.29 | mosty | ManxPower, that's not what i want however- i want to know what sipclient's callerid was on asterisk2 |
| 14:53.38 | mosty | not asterisk1's callerid |
| 14:53.44 | ManxPower | mosty: then don't override it. |
| 14:53.57 | mosty | i'm not overwriding it |
| 14:54.01 | ManxPower | don't put callerid= in server 1's info on server 2 |
| 14:54.12 | ManxPower | mosty: you realize that a server does not have callerid, right? |
| 14:54.26 | ManxPower | So I'm not sure what you mean by "asterisk1's callerid" |
| 14:54.55 | *** join/#asterisk postc (n=marquis@gw-corp.postconf.com) |
| 14:55.41 | ManxPower | We always have the correct callerid information forwarded between all SIP devices on all servers on our network. No special configuration required. |
| 14:55.48 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:55.49 | mosty | sip phones have accounts on asterisk1, when i dial a number from a sip phone to asterisk1, asterisk1 dials via asterisk2. asterisk2 sees the callerid of the call as "asterisk" |
| 14:56.14 | ManxPower | mosty: that is what asterisk puts in CLID if it does not receive CLID information. |
| 14:56.46 | ManxPower | mosty: when my SIP phones on server one dial a SIP phone on server two the destination always sees the correct callerid name and number. |
| 14:57.05 | [TK]D-Fender | mosty: I believe your use of "fromuser" is polluting that. |
| 14:57.17 | ManxPower | Sp the question is what are you doing that is breaking that. |
| 14:57.31 | mosty | [TK]D-Fender, hmm ok i'll test changing that |
| 14:57.44 | ManxPower | [TK]D-Fender: I think we use fromuser. |
| 14:58.09 | ManxPower | mosty: paste asterisk1's sip.conf info as listed in server2's sip.conf |
| 14:58.15 | ManxPower | to pastebin, of course |
| 14:58.45 | mosty | one sec, i'll test without fromuser being set on asterisk1 |
| 14:58.53 | *** part/#asterisk postc (n=marquis@gw-corp.postconf.com) |
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| 14:59.44 | *** join/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 14:59.51 | mosty | if i remove the fromuser setting on asterisk1, then it can't authenticate to asterisk2 |
| 15:00.27 | ManxPower | That's what I would expect. Notice [TK]D-Fender did not say remove it, he just said that's what he thinks is causing it. |
| 15:00.28 | mosty | Failed to authenticate user "Unknown" <sip:Unknown@myipaddress> |
| 15:00.34 | mosty | ah ok |
| 15:01.26 | [TK]D-Fender | mosty: you'll likely need to tweak all of your account settings |
| 15:02.00 | ManxPower | now paste the sip.conf entries from both servers. |
| 15:02.14 | mosty | ok, i will paste, one moment |
| 15:02.20 | ManxPower | I have to leave in -2 mins. |
| 15:03.39 | mosty | do you just want the user/peer entries? my sip.conf is split into multiple files |
| 15:05.40 | ManxPower | how you split your files is not my problem. yes, just the user/peer entries for communictaion from asterisk1 to asterisk2 |
| 15:07.51 | mosty | http://pastebin.com/m4af744ed |
| 15:08.23 | mosty | it would have taken me a bit longer to construct the entire sip config if you wanted that, is all |
| 15:08.53 | [TK]D-Fender | mosty: you should has pastebinned a call with sip debug enabled...... |
| 15:09.17 | *** part/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 15:09.35 | mosty | i can get that for you |
| 15:10.42 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 15:11.32 | *** join/#asterisk mocker (n=ksexton@198.247.173.227) |
| 15:11.49 | festr_ | hi |
| 15:12.21 | festr_ | is it possible to disable p2p bridge in SIP (1.4)? I need to make conversion between inband DTMF to SIP INFO |
| 15:13.06 | *** join/#asterisk s0lid (n=s0lid@210.213.243.178) |
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| 15:15.35 | mosty | here are the first couple of sip debug messages: http://pastebin.com/m2b6bf0e6 |
| 15:15.43 | Greek-Boy | scopserv doesn't do billing |
| 15:15.52 | Greek-Boy | I want a GUI mainly for billing |
| 15:16.34 | *** join/#asterisk Op3r (n=edwin@222.127.214.210) |
| 15:17.40 | [TK]D-Fender | festr_: "canreinvite=no" |
| 15:18.01 | [TK]D-Fender | Greek-Boy: I believe they have a billing module as they are designed for ITSP use as well |
| 15:18.03 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 15:19.13 | festr_ | [TK]D-Fender: i'm using this. i'm bridgin one SIP (inband RTP) and SIP trunk with sip info. but in ethereal there is no SIP INFO but RTP DTMF |
| 15:19.26 | [TK]D-Fender | mosty: Incomplete. Please pastebin the COMPLETE call attempt and don't mask anything. Provide 1 for server A, one for server B's response |
| 15:19.47 | [TK]D-Fender | festr_: pastebin your configs |
| 15:20.38 | ManxPower | mosty: at this point I leave you in [TK]D-Fender's capable hands |
| 15:20.49 | [TK]D-Fender | ManxPower: Debateable ;) |
| 15:24.32 | mosty | ok, one moment while i create the files |
| 15:29.28 | *** join/#asterisk O-Mega (i=O-Mega@175-21.wireless.mvn.net) |
| 15:30.41 | mosty | [TK]D-Fender, do you know of a pastebin site that lets you upload files? if not i could dcc them to you if that's ok, otherwise i'll keep looking |
| 15:30.55 | *** join/#asterisk codestr0m (n=asura@ip5451d5cd.direct-adsl.nl) |
| 15:31.01 | [TK]D-Fender | mosty: copy&paste |
| 15:31.20 | [TK]D-Fender | mosty: this is not Raw-Cat science |
| 15:31.22 | codestr0m | I'm having some strange intermittent problems lately.. |
| 15:31.23 | codestr0m | Jan 14 15:30:22 WARNING[11895]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' |
| 15:31.41 | codestr0m | I'm on Asterisk 1.2.17 and worried about the upgrade to 1.4 |
| 15:31.47 | [TK]D-Fender | codestr0m: that error is completely self explanitory |
| 15:32.23 | codestr0m | yeah I don't have a timeout rule, but what I don't understand is why it's timing out |
| 15:32.56 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:32.56 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:33.04 | codestr0m | [TK]D-Fender: sometimes it's working sometimes not.. I'm looking at the sip debug and just don't make sense |
| 15:33.04 | [TK]D-Fender | codestr0m: pastebint he complete CLI output of your failed call attempt at versbose 10 along with your associated dialplan. |
| 15:33.05 | [TK]D-Fender | ~pb |
| 15:33.06 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:33.07 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 15:33.18 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:33.22 | [TK]D-Fender | codestr0m: SIP debug has nothing to do with IVR timeouts. |
| 15:33.43 | [TK]D-Fender | codestr0m: Unless you aren't getting any DTMF period. |
| 15:33.56 | codestr0m | I thought it might be a timeout with the ua for some reason.. I'm stumped |
| 15:34.02 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:35.00 | O-Mega | anyone know of a gui or config generator for LumenVox Speech? |
| 15:35.04 | *** join/#asterisk Egonis (n=root@207.245.216.9) |
| 15:35.25 | Egonis | what does 'bpviol' mean on 'zap show status'? becaue my tor2 span 1 shows 2 bpviol, but an 'OK' under alarms |
| 15:36.11 | [TK]D-Fender | codestr0m: PASTEBIN <- |
| 15:37.39 | mosty | [TK]D-Fender, mind if i /msg you the url's, so my private info isn't available to the entire world? |
| 15:37.57 | [TK]D-Fender | mosty: tahts fine |
| 15:38.12 | [TK]D-Fender | mosty: hold that thought |
| 15:38.28 | [TK]D-Fender | mosty: You haven't PB'd them yet, have you? |
| 15:38.38 | mosty | i have already, yes |
| 15:38.45 | codestr0m | [TK]D-Fender: http://paste.debian.net/46866 |
| 15:39.40 | Egonis | how do I place a test call from the CLI? e.g. dial out on channel 1/span 1 > number? |
| 15:41.51 | *** join/#asterisk ddunavant (n=David@68-244-55-181.area3.spcsdns.net) |
| 15:43.21 | [TK]D-Fender | Egonis: Use at least a soft-phone |
| 15:44.18 | [TK]D-Fender | codestr0m: its trying to treat your exten like an IVR (and it shouldn't) because you also have no priority #4 <- |
| 15:44.29 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:44.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:44.35 | [TK]D-Fender | codestr0m: You cannot just skip priorities like that |
| 15:45.11 | codestr0m | I had a prio 4 before and it was to set callerid and was mutli ringing.. thanks |
| 15:46.30 | [TK]D-Fender | codestr0m: 8 & 90 are also useless |
| 15:47.06 | puppet | non one knows a project with astersik realtime/manager and c#? all i find is outofdate and links dead |
| 15:47.45 | *** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
| 15:48.02 | codestr0m | [TK]D-Fender: no because I have the goto and if it doesn't hang-up before the other condition it'll play the discon-or-out-of-service message |
| 15:48.06 | codestr0m | which is only for my ex |
| 15:48.41 | *** join/#asterisk fainsys (n=adam@c-76-17-121-45.hsd1.ga.comcast.net) |
| 15:48.59 | [TK]D-Fender | codestr0m: Missed the Goto... correct for 90.. 8 however.... |
| 15:49.23 | [TK]D-Fender | codestr0m: and exten => 2062795000,2,GotoIf($["${CALLERIDNUM}" = "${MERVE}"]?90:3) <- ${CALLERIDNUM} is deprecated. Please use the CALLERID function. |
| 15:49.26 | codestr0m | thanks though. I'd only have one other question.. which is ME=SIP/cbergstrom1-1&SIP/gradwell/xxxxxx something was up with the gradwell forwarding.. (used to just work (tm)) and |
| 15:49.45 | codestr0m | anywho. won't take up more of your time. thanks a lot |
| 15:50.01 | [TK]D-Fender | codestr0m: For your last one, whats not working? |
| 15:50.45 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 15:51.33 | codestr0m | [TK]D-Fender: I'd have to enable it again and see, but just started having problems with the multi ring when I put gradwell in there. I'm just going to remove it for now |
| 15:52.24 | [TK]D-Fender | codestr0m: If Gradwell's service "answers" *'s call and passes ringing inband as audio then your other SIP device will ring once at most if you're lucky. |
| 15:52.42 | codestr0m | yes. that was exactly what was happening |
| 15:52.45 | codestr0m | , but it used to work |
| 15:53.04 | [TK]D-Fender | codestr0m: Also watcht he "fun" continue if the # you are calling defaults to Voicemail for some reason |
| 15:53.15 | [TK]D-Fender | codestr0m: maybe Gradwell changed. |
| 15:53.36 | codestr0m | [TK]D-Fender: lol. I had to disable that stupid vm earlier. it's been a mess, but all makes sense now |
| 15:54.22 | [TK]D-Fender | codestr0m: multi-dial including PSTN = predictable mess |
| 15:55.10 | [TK]D-Fender | codestr0m: You'd have to add separate nested local channels each with their own timeout and suing a host that actually provides call-progress instead ot passing progress as audio. |
| 15:55.11 | codestr0m | [TK]D-Fender: yeah. it's hard to know what's actually happening with the call and respond accordingly.. |
| 15:55.27 | *** join/#asterisk ifnotwhynot (n=davidh@196.34.229.130) |
| 15:56.01 | ifnotwhynot | hi there need some help dialing out from asterisk to patton gateway, any help welcome |
| 15:56.18 | codestr0m | good point. I know a lot of voip/sip hosts that answer the call and pass ringback as call progress |
| 15:56.41 | codestr0m | (not sure I worded that correctly. you understand though) |
| 15:57.03 | [TK]D-Fender | codestr0m: yes, thats clear enough... |
| 15:57.19 | [TK]D-Fender | ifnotwhynot: Show us the problem and maybe we can have something to comment on. |
| 15:57.40 | *** join/#asterisk essiene (n=essiene@212.100.73.98) |
| 15:57.54 | codestr0m | [TK]D-Fender: if you don't mind. which company are you working with? |
| 15:59.13 | essiene | hi all... kinda a newbie to asterisk and AGIs. I have a python agi which does some processing that could take time. i'm using fork to let the parent process die quickly, so the caller doesn't experience delay. problem is... on the commandline, the parent returns to me quickly, but when i call it via AGI(), it still takes a lot of time |
| 15:59.22 | ifnotwhynot | k TK regitered Patton Bri gateway with asterisk no problem(using sip), incoming call to asterisk no problem, outgoing using 1,1,Dial(SIP/patton/8888) 8888 being my extension number but no luck |
| 15:59.48 | essiene | is there something i should be aware of? or is there a way to spawn an AGI() call in the background? (hopefully.. BAGI? :D ... hehe... a guy can be hopefull :D) |
| 16:00.12 | ifnotwhynot | patton device connected to pabx for testing |
| 16:00.13 | [TK]D-Fender | codestr0m: I don't work in the VoIP field, I jsut help around here. I am an independent consultant however off and on |
| 16:00.42 | [TK]D-Fender | ifnotwhynot: PASTEBIN is your friend. So is SIP debug. |
| 16:01.03 | codestr0m | ok. just curious. thanks. any b2bua you've worked with lately (non-asterisk) |
| 16:01.05 | ifnotwhynot | TK how much to come to south africa and share your knowledge with me? |
| 16:01.06 | [TK]D-Fender | ~pb |
| 16:01.06 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:01.21 | ifnotwhynot | what is sip debug |
| 16:01.28 | [TK]D-Fender | ifnotwhynot: I can do that remotely.... you couldn't afford my travel expesnses :) |
| 16:01.36 | ifnotwhynot | try me |
| 16:02.02 | ifnotwhynot | sponsored holiday TK |
| 16:02.08 | codestr0m | [TK]D-Fender: africa's cheap. you're overestimating how much you could actually cost them there outside of working hours |
| 16:02.19 | [TK]D-Fender | ifnotwhynot: It really shouldn't require physical presence.... anyways. Do "sip debug" from CLI, and pastebin the complete call output for your failed attempt at verbose 10 |
| 16:02.33 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 16:03.03 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 16:04.55 | ifnotwhynot | http://pastebin.com/m40b16292 |
| 16:05.01 | *** part/#asterisk Makenshi (n=makenshi@makenshi.at.furry.be) |
| 16:05.29 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 16:05.43 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:08.10 | [TK]D-Fender | ifnotwhynot: please pastebin again, you cut off an initial invite. there is also no dialplan output in there, and include your sip.conf entries. |
| 16:09.46 | ifnotwhynot | if you debug do i need to debug the patton gateway or my extension where i am dialing from? TK |
| 16:10.56 | [TK]D-Fender | ifnotwhynot: Don't know.. I'm not seeing enough yet. |
| 16:11.47 | ifnotwhynot | go it working typo |
| 16:12.12 | ifnotwhynot | sorry for the trouble |
| 16:12.43 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584655.dsl.bell.ca) |
| 16:12.43 | ifnotwhynot | so how about the holiday in south africa i will show you lions |
| 16:12.50 | [TK]D-Fender | ifnotwhynot: no problem |
| 16:12.52 | ifnotwhynot | tk |
| 16:13.20 | ifnotwhynot | tk: where you from if you don't mind me asking |
| 16:13.23 | ifnotwhynot | ? |
| 16:14.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:14.51 | [TK]D-Fender | ifnotwhynot: Montreal, QC, Canada |
| 16:14.55 | codestr0m | ifnotwhynot: learn what "whois" is |
| 16:15.04 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 16:16.06 | *** join/#asterisk eldon (n=eldon@216.207.245.1) |
| 16:18.18 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 16:18.27 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 16:18.49 | dacs | good morning all |
| 16:18.57 | Nugget | Moo |
| 16:20.19 | [TK]D-Fender | file: Calories boy! Don't skimp! |
| 16:20.38 | file | I'm counting how many times my roommate's work will call for him before he wakes up |
| 16:20.49 | file | so far only two, but they sounded desperate last time |
| 16:21.29 | ifnotwhynot | whois codestr0m |
| 16:23.04 | codestr0m | ifnotwhynot: you can 'whois' people with irc. which would have shown you [TK]D-Fender was at a .ca domain. digging further instead of merely asking them. where are you from :P (not-withstanding proxies of course) |
| 16:23.14 | variable_office | I am trying to figure out the cause of a number of really strange traffic issues with a sip user (using ulaw) with wireshark. When I analyze the stream I am getting a jitter of something like 28,000,000 ms, even though the call isnt even that long, I am losing next to no packets(less than .1%) and otherwise it all looks great, The user is saying they are getting echoes, and noise, and other crazy things that I dont hear on my side of the |
| 16:23.14 | variable_office | <PROTECTED> |
| 16:23.22 | file | codestr0m: or cloaks. |
| 16:23.31 | *** join/#asterisk alrs (i=non-knav@pozug.com) |
| 16:24.17 | [TK]D-Fender | codestr0m: unless I was connecting via TOR like so many do :) |
| 16:24.37 | [TK]D-Fender | codestr0m: Or was passing through another server, or a few dozen other reasons :) |
| 16:25.14 | codestr0m | [TK]D-Fender: you don't want to use tor anymore unless you can trust the exit node.. |
| 16:25.36 | [TK]D-Fender | codestr0m: was just saying.... don't trust anything... Except me. |
| 16:26.05 | [TK]D-Fender | codestr0m: ";) |
| 16:26.38 | file | he only almost killed me once |
| 16:27.46 | [TK]D-Fender | file: No... every vehicle in line of sight could have been the end for you..... thats COUNTLESS times! ;) |
| 16:27.50 | variable_office | I monitored the conversation and i cant hear any distortion on the asterisk side of things |
| 16:28.35 | dacs | [TK]D-Fender: can you help me with http://pastebin.ca/854819 |
| 16:29.15 | dacs | i get Registration error: 403 -Forbidden |
| 16:29.16 | [TK]D-Fender | dacs: its telling you to your face you shouldn't be putting a host IP added in there if your phone is going to REGISTER <- |
| 16:30.07 | dacs | [TK]D-Fender: what? i don't understand |
| 16:30.47 | *** join/#asterisk BajaEd (n=ednagy@72.168.135.209) |
| 16:31.07 | [TK]D-Fender | dacs: You put "host=[someip]" in your sip.conf instead "host=dynamic". Therefor * does not LET them register. |
| 16:32.11 | [TK]D-Fender | dacs: tahts like phoning home to tell your partner "Hey I'm at the office, you can call me there!" and getting "I ONLY call you at one #, so I don't CARE where you say you are! Get lost!" |
| 16:32.38 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 16:34.02 | dacs | my host is dynamic thu |
| 16:35.22 | [TK]D-Fender | dacs: You did not put "host=dynamic" in sip.conf. Asterisk is not lying. |
| 16:35.28 | [TK]D-Fender | dacs: [Jan 14 08:24:59] ERROR[25447]: chan_sip.c:8575 register_verify: Peer 'Edwardo' is trying to register, but not configured as host=dynamic |
| 16:35.33 | [TK]D-Fender | dacs: Read the big print. |
| 16:37.07 | *** join/#asterisk uwe (n=d46a490f@gateway/web/cgi-irc/ircatwork.com/x-8098102f8e17e757) |
| 16:37.54 | uwe | hello, i have 2 asterisk machines, connecting to a cisco AS5400 , one machine can dial SIP normally, the other cant, what should i be looking at while debugging ? |
| 16:39.53 | dacs | [TK]D-Fender: http://pastebin.ca/854834 |
| 16:40.41 | [TK]D-Fender | dacs: Eitehr the leading whitespace is bad or you haven't applied your changes |
| 16:41.38 | dacs | [TK]D-Fender: when i try phone1 it works fine , but edwardo doesn't |
| 16:41.40 | dacs | hmmmm |
| 16:42.13 | [TK]D-Fender | dacs: Of course dialing Edwardo doesn't work, he's set up wrong and * has no idea where to find him. |
| 16:44.16 | dacs | [TK]D-Fender: it's exten 700 and i have it my extention.conf |
| 16:44.30 | dacs | [TK]D-Fender: exten => 700,1,Dial(SIP/Edwardo) |
| 16:44.44 | [TK]D-Fender | dacs: Doesn't matter, you sip.conf is bad and your phone can't register |
| 16:44.57 | dacs | [TK]D-Fender: bad where |
| 16:45.15 | [TK]D-Fender | dacs: I've told you twice. Pay attention. |
| 16:45.32 | Daviey | keep up back there |
| 16:46.21 | dacs | [TK]D-Fender: you said its not Dynamic and i showed you it was |
| 16:46.33 | [TK]D-Fender | <[TK]D-Fender>dacs: Eitehr the leading whitespace is bad or you haven't applied your changes |
| 16:47.32 | dacs | [TK]D-Fender: am sorry but what do you exactly mean by the leading white space |
| 16:47.42 | [TK]D-Fender | dacs: And there is no such thing as "caninvite=no" |
| 16:48.01 | [TK]D-Fender | dacs: the SPACES in front of "host=dynamic" |
| 16:49.42 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:51.17 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:54.46 | dacs | [TK]D-Fender: thank you, that fixed it |
| 16:55.11 | *** join/#asterisk kamanashisroy (n=root@202.56.7.135) |
| 16:55.51 | elverkilde | [TK]D-Fender: I admire your patience :) |
| 16:56.14 | [TK]D-Fender | elverkilde: believe me... it has limits... |
| 16:57.43 | drmessano | lol |
| 16:58.11 | kamanashisroy | hi , anyone tried to execute dialplans while it is dialing an outgoing channel ? I tried G option of dial application .. I think there is something missing .. or I do not know .. |
| 16:58.11 | [TK]D-Fender | drmessano: Oh? You've been hand-holding him over the simple configuration of FWD yourself.... |
| 16:58.17 | drmessano | lol |
| 16:58.22 | drmessano | Im just messing with you |
| 16:58.22 | elverkilde | lol |
| 16:58.34 | drmessano | My forehead is already dented in |
| 16:58.42 | [TK]D-Fender | kamanashisroy: you cannot do stuff while its dialing. |
| 16:58.54 | [TK]D-Fender | kamanashisroy: that is not what "g" is for |
| 16:59.40 | drmessano | I did more than hand hold.. I rewrote his configs lol |
| 16:59.46 | drmessano | at 3am |
| 16:59.59 | drmessano | or was it PM |
| 17:00.03 | badcfe | can sox generate alaw output for asterisk? how should it be done? |
| 17:00.06 | drmessano | I forget, either way I wasnt sober |
| 17:00.29 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 17:01.11 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 17:01.21 | kamanashisroy | [TK]D-Fender: G(context^exten^pri) - If the call is answered, transfer the calling party to |
| 17:01.22 | kamanashisroy | <PROTECTED> |
| 17:01.22 | kamanashisroy | <PROTECTED> |
| 17:01.22 | kamanashisroy | <PROTECTED> |
| 17:01.22 | kamanashisroy | <PROTECTED> |
| 17:01.27 | kamanashisroy | sorry :-P |
| 17:01.33 | x86 | ouch |
| 17:01.37 | x86 | +b! |
| 17:01.39 | x86 | ;) |
| 17:01.56 | [TK]D-Fender | kasmad that doesn't do stuff WHILE its dialing, that does stuff once it is ANSWERED |
| 17:02.08 | *** join/#asterisk glen2 (n=glen@212.54.184.253) |
| 17:02.26 | [TK]D-Fender | kamanashisroy: rather |
| 17:02.33 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 17:02.35 | dacs | ~pb |
| 17:02.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:02.38 | kamanashisroy | [TK]D-Fender: yes .. answered I mean .. But the outgoing channels is not hanged up .. |
| 17:03.05 | [TK]D-Fender | kamanashisroy: then pastebin your full config and CLI output so we can see what's going on. |
| 17:03.09 | *** join/#asterisk fiXXXerMet (n=meep@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 17:03.15 | glen2 | Rar, I have a question! Can someone recomend a good Asterisk training course in London, Kent or Hampshire. |
| 17:03.24 | glen2 | s/./?/ |
| 17:03.33 | elverkilde | badcfe: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql |
| 17:04.22 | glen2 | s/!/`uname -a`/ |
| 17:04.28 | fiXXXerMet | After installing ztdummy, and recompiling Asterisk with support for it, my sounds no longer play... I can see them playing on the command line, but the person on the phone never hears anything. |
| 17:04.40 | fiXXXerMet | Nice try glen2 :) |
| 17:04.42 | kamanashisroy | [TK]D-Fender: Let me explain, in caller dialplan I dial and the callee answers .. after that I send the caller billing information in each 5 seconds .. I have done this using G() option in dial .. |
| 17:04.54 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:05.07 | *** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net) |
| 17:05.21 | glen2 | I the type of person who voids warrentys, I had to see if I could pick jbot apart. |
| 17:05.59 | [TK]D-Fender | kamanashisroy: PASTEBIN <------ |
| 17:06.09 | kamanashisroy | <PROTECTED> |
| 17:07.06 | glen2 | s/^$/:D/ |
| 17:07.24 | glen2 | s/^*$/:D/ |
| 17:07.26 | drmessano | [TK]D-Fender: the other thing is, once you get FWD working, you get to spend the rest of your life fixing it |
| 17:07.32 | badcfe | elverkilde: my version v14.0.0 doesnt like that -w .. |
| 17:08.12 | glen2 | What, in your opinion, is the best way to learn Asterisk? |
| 17:08.12 | kamanashisroy | [TK]D-Fender: give me a second |
| 17:08.39 | drmessano | My two favorite question |
| 17:08.45 | [TK]D-Fender | glen2: Grab the book and get reading. Then jsut start playing with it. |
| 17:08.46 | [TK]D-Fender | ~book |
| 17:08.47 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 17:08.52 | drmessano | "How do I learn Asterisk?" "How long does it take" |
| 17:09.12 | [TK]D-Fender | drmessano: "How I can werk plaeses!?!??!" |
| 17:09.45 | fiXXXerMet | After installing ztdummy, and recompiling Asterisk with support for it, my sounds no longer play... I can see them playing on the command line, but the person on the phone never hears anything. |
| 17:09.48 | drmessano | [TK]D-Fender: "I just got a job admining an "Asterisk" how can I learn it real quick?" |
| 17:10.30 | outtolunc | you put the horse in *front* of the cart |
| 17:10.44 | *** join/#asterisk robeph (n=robf@24.214.206.254) |
| 17:10.58 | [TK]D-Fender | outtolunc: And the heaad under the wheel ;) |
| 17:11.20 | robeph | anyone know of any problems with polycom phones not forwarding :-\ |
| 17:11.26 | glen2 | OK drmessano I'll rephrase my question. I'm about to try and learn Asterisk, is there anything I should know before I start so I can avoid common mistakes that plague new users> |
| 17:11.31 | robeph | they were working fine, and now suddenly, several stop forwarding... |
| 17:11.44 | drmessano | Mabe you shud join #AKERISK-BEGINAR |
| 17:11.47 | drmessano | lol |
| 17:11.49 | choogaistir | http://slil.ru/25354596 best MOH )))) |
| 17:11.51 | drmessano | I R BEGINAR |
| 17:12.05 | robeph | :-s |
| 17:12.09 | glen2 | Akerisk? |
| 17:12.15 | drmessano | lol |
| 17:12.15 | robeph | glen2: he's jerkin ya |
| 17:12.23 | drmessano | yes |
| 17:12.24 | drmessano | Sorry |
| 17:12.43 | drmessano | Had a bad night with a newb, and wasted so much of my time, that *I* actually got pwned |
| 17:12.45 | drmessano | :( |
| 17:13.17 | robeph | eh...anyhow bout these polycoms...its driving us nuts...cos we cannot figure out wtf could possibly be wrong with it |
| 17:13.24 | drmessano | <Dude> I got ERRAR: Passward NO WORKY |
| 17:13.28 | drmessano | Then 12 hours later |
| 17:13.34 | drmessano | <Dude> I got different ERRAR: Passward NO WORKY |
| 17:13.40 | robeph | asterisk is giving 100 trying, and then nothing |
| 17:14.15 | robeph | amd then 304 not modified. |
| 17:14.51 | robeph | nm wtf are these guys doing... thats the http error... lemme go smack these guys |
| 17:17.38 | elverkilde | badcfe: Must be deprecated. Anyway, u need to resample to 16b 8kHz mono... check the sox docs, maybe? |
| 17:18.41 | kamanashisroy | [TK]D-Fender: here is my problem http://paste.uni.cc/18093 |
| 17:18.49 | [TK]D-Fender | fiXXXerMet: I've heard of things going wrong with ZTDUMMY like that. stop *, rmmod it, then restart if audio comes back, thats the issue. I'm not sure what to full reason or proper correction is however |
| 17:19.26 | [TK]D-Fender | kamanashisroy: and I said I wants the full CLI OUTPUT. |
| 17:19.32 | *** join/#asterisk pLr (n=bobo@unaffiliated/plr) |
| 17:19.38 | fiXXXerMet | [TK]D-Fender: I'll give it a shot - thanks. |
| 17:20.18 | kamanashisroy | [TK]D-Fender: within a minute :-P |
| 17:23.49 | kamanashisroy | [TK]D-Fender: Here is it http://paste.uni.cc/18094 |
| 17:24.29 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 17:25.03 | dacs | from x-lite to my ATA rings but no voice? |
| 17:25.33 | *** join/#asterisk vrtk (n=bruno@201.9.57.7) |
| 17:31.59 | [TK]D-Fender | kamanashisroy: That is jsut looping around in circles.. what are you trying to do? |
| 17:33.28 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 17:33.40 | dacs | [TK]D-Fender: http://pastebin.ca/854905 |
| 17:34.30 | [TK]D-Fender | dacs: Read up : |
| 17:34.32 | [TK]D-Fender | ~sipnat |
| 17:34.33 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:36.42 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 17:37.51 | kamanashisroy | [TK]D-Fender: yes it loops and shows the billing information to the caller .. |
| 17:38.15 | [TK]D-Fender | kamanashisroy: All it seems to be doing is going around in circles doing nothing though..... |
| 17:38.42 | kamanashisroy | [TK]D-Fender: But the two loops(called and caller) does not hangup .. |
| 17:38.50 | kamanashisroy | [TK]D-Fender: The problem is I cannot hangup the calling party on called party hangup and vice varsa. |
| 17:39.09 | kamanashisroy | [TK]D-Fender: it loops and sends text messages to the caller .. |
| 17:39.12 | [TK]D-Fender | kamanashisroy: Why not? Grab the phone. Hang up. |
| 17:39.20 | [TK]D-Fender | kamanashisroy: Whats the problem? |
| 17:40.08 | kamanashisroy | kamanashisroy: SendText(you have talked 12 minutes and your bill is 12BDT); bala bla in each 5 seconds in a loop .. |
| 17:40.39 | kamanashisroy | [TK]D-Fender: problem is I cannot hangup the caller when the called party hangs up .. |
| 17:41.12 | kamanashisroy | [TK]D-Fender: any idea ? |
| 17:41.15 | [TK]D-Fender | kamanashisroy: Why is it you can't just hangup? |
| 17:41.52 | [TK]D-Fender | kamanashisroy: And are the 2 ends talking during all this looking? |
| 17:41.55 | [TK]D-Fender | looping* |
| 17:42.26 | kamanashisroy | [TK]D-Fender: I send message to caller until the called party has hangup .. But how do I know that the called party has hangup ? |
| 17:42.36 | kamanashisroy | [TK]D-Fender: yes |
| 17:42.42 | kamanashisroy | [TK]D-Fender: certainly .. |
| 17:43.04 | kamanashisroy | [TK]D-Fender: two channels are up and bridged .. |
| 17:43.16 | [TK]D-Fender | kamanashisroy: Thats up to the endpoint you are calling. |
| 17:43.48 | kamanashisroy | [TK]D-Fender: and eventually , suppose the called party hangs up ... the calling party goes on while I want him hangup too .. but how ?? |
| 17:44.02 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 17:44.46 | kamanashisroy | [TK]D-Fender: yes the caller still loops when the "endpoint I am calling" hangs up :( .. |
| 17:45.00 | [TK]D-Fender | kamanashisroy: No idea... |
| 17:47.29 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 17:49.07 | kamanashisroy | [TK]D-Fender: I think we can write an application 'stillTalking' which will return 0/1 |
| 17:49.29 | kamanashisroy | 0 when the bridged channel is not talking ..else 1 |
| 17:51.19 | kamanashisroy | if(bchan = ast_bridged_channel(chan)) { return !(ast_check_hangup(bchan)) } return 0; |
| 17:51.39 | kamanashisroy | [TK]D-Fender: what do you think ? |
| 17:53.15 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 17:54.04 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 17:54.16 | [TK]D-Fender | kamanashisroy: No idea... |
| 17:54.24 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 17:54.47 | kamanashisroy | [TK]D-Fender: Thank you for your time |
| 17:54.49 | *** join/#asterisk murdmath (n=vircuser@209.181.82.1) |
| 17:54.53 | *** part/#asterisk kamanashisroy (n=root@202.56.7.135) |
| 17:55.21 | TJNII | Hmmmm.... is there a way to have app-meetme remember my name between conference calls? I'd like to only have to record my name once on my desk phone. |
| 17:55.37 | ZPertee | does anyone know the telephone line color codes? I can't seem to find it on google. I am using 4 conductor (26 guage) with 4 conductor plugs |
| 17:55.57 | TJNII | Bell operators give better service |
| 17:56.18 | ZPertee | the colors that it uses are yellow, green, red, and black |
| 17:56.44 | TJNII | That's old. |
| 17:56.49 | TJNII | But still common |
| 17:57.45 | TJNII | Red - Line 1 ring (-), Green - Line 1 tip (+), Yellow and black are line 2, I forget tip + ring |
| 17:58.54 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 17:59.02 | ZPertee | TJNII ok. can you put that a little more in lehman's terms for me. Basically I am trying to run line from an fxs port to message waiting indicator light |
| 17:59.11 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 17:59.23 | TJNII | Hook up red & green |
| 18:00.26 | dacs | ~sipnat |
| 18:00.26 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:01.08 | ZPertee | ok. so there is 4 prongs in the plug where do I put the red and green wire? and do I need to reverse the order or anything? |
| 18:02.02 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 18:03.09 | TJNII | http://www.inventgineering.com/telephone_wiring_codes.htm |
| 18:04.46 | TJNII | Line one is on the middle two pins. Polarity really shouldn't matter, though it is good practice to try and maintain it. |
| 18:05.23 | *** join/#asterisk pepse (n=pepse@71-223-194-69.phnx.qwest.net) |
| 18:08.47 | hmmhesays | ~seen coppice |
| 18:08.51 | jbot | coppice <n=chatzill@230.202.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 4d 1h 42m 49s ago, saying: 'and with some revisions of the handytone firmware T.38 even works'. |
| 18:08.55 | jwh | uh |
| 18:09.06 | jwh | why am I geting forbidden messages from my own ip? |
| 18:09.13 | jwh | (calls in via h323, out via sip provider) |
| 18:09.29 | [TK]D-Fender | jwb : Because |
| 18:09.46 | jwh | it was ok on iax |
| 18:09.50 | [TK]D-Fender | jwh: rather |
| 18:09.52 | [TK]D-Fender | . |
| 18:10.01 | jwh | but i've changed to sip as iax can't do so many calls |
| 18:10.20 | jwh | but now no calls work |
| 18:11.12 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 18:11.42 | hmmhesays | Heh I guess I'm diving into the embedded asterisk world again |
| 18:11.44 | hmmhesays | GREATE |
| 18:13.07 | tzanger | hmmhesays: come on, it's fun |
| 18:14.06 | hmmhesays | tzanger: about as fun as pulling your hair out |
| 18:14.25 | tzanger | nah, the embedded asterisk stuff isn't too bad, rowe et al have done the hard work there |
| 18:14.35 | tzanger | it's adding new channel drivers and the kernel stuff ot drive 'em that's nasty :) |
| 18:15.01 | hmmhesays | yeah |
| 18:15.10 | hmmhesays | the astfin project is pretty good on the blackfin it seems too |
| 18:15.21 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 18:15.37 | hmmhesays | I had it compiled on an audiocodes tulip about 6 months ago |
| 18:15.40 | hmmhesays | what a nightmare |
| 18:18.26 | [TK]D-Fender | hmmhesays: Trying to run * on a Mediant? |
| 18:19.41 | [TK]D-Fender | hmmhesays: OMG.. an ATA?! |
| 18:19.52 | [TK]D-Fender | hmmhesays: Whats the point? |
| 18:21.05 | tzanger | yes that''s wha my embedded asterisk stuff is on |
| 18:21.07 | hmmhesays | the point was I was getting paid to do it |
| 18:21.31 | *** join/#asterisk l2trace99 (n=asd@fl-67-76-209-28.sta.embarqhsd.net) |
| 18:21.34 | tzanger | hmmhesays: that's one of the best points :-) |
| 18:21.38 | hmmhesays | the ultimate goal of that project was a bad one but I didn't care |
| 18:21.45 | jwh | wtf seriously, so I have incoming h323 peers, which send in calls, which get passed to sip provider, why on earth would it be showing forbidden ;/ |
| 18:22.05 | hmmhesays | sip debug |
| 18:22.19 | mosty | jwh, which host is saying forbidden? your sip provider, or your own host? |
| 18:22.28 | jwh | it appears to be coming from my own host |
| 18:22.38 | jwh | <dstnumber@pbxip> |
| 18:25.02 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:25.23 | [TK]D-Fender | jwh: it is appears you're showing us nothing of value... |
| 18:25.34 | jwh | [TK]D-Fender: sec, stuck on gprs in the hospital atm |
| 18:26.00 | jwh | what info do you nee?d |
| 18:26.30 | [TK]D-Fender | jwh: all of the SIP/H.323 debug from CLI in a pastebin and your configs to match |
| 18:26.37 | [TK]D-Fender | ~pb |
| 18:26.38 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:26.47 | jwh | righto |
| 18:28.20 | jblack | I got incoming faxes last night. |
| 18:31.32 | robeph | what could cause something like "WARNING[4537]: app_meetme.c:1541 conf_run: Unable to write frame to channel: Inappropriate ioctl for device" just anything come to mind readily? |
| 18:34.56 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 18:35.30 | magic_hat | hey everyone. Is there a way that I can use a different dtmf setting depending on the caller ID of an incoming call? |
| 18:37.36 | jwh | heh, asterisk can do less sip calls than it can iax |
| 18:37.38 | jwh | interesting one |
| 18:37.46 | Jam0r | jwh! |
| 18:37.54 | jwh | Jam0r: msn |
| 18:37.56 | Jam0r | was just trying yo phone you |
| 18:37.58 | jwh | before I hurt someone |
| 18:38.33 | _ShrikE | magic_hat: core show applicatoin SIPDtmfMode |
| 18:38.39 | Jam0r | jwh: you aint on |
| 18:38.45 | jwh | on company one |
| 18:38.48 | _ShrikE | s/applicatoin/application/ |
| 18:38.48 | Jam0r | ah |
| 18:39.45 | magic_hat | jbot: not following this... can you explain more? |
| 18:41.15 | _ShrikE | magic_hat: You may be able to do a gotoif on the callerid and then set the appropriate dtmf mode with SIPDtmfMode |
| 18:43.46 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 18:43.47 | magic_hat | will this work in sip.conf: dtmfmode=${IF("${CALLERIDNUM}" = "7739891234"?inband:rfc2833)} |
| 18:44.17 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 18:46.59 | magic_hat | hrm.... seems to be okay. dtmfmode=IF("${CALLERIDNUM}" = "7739891234"?inband:rfc2833) |
| 18:47.29 | *** join/#asterisk tobias (n=tobias@rrcs-70-62-101-155.midsouth.biz.rr.com) |
| 18:47.43 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 18:49.45 | mvanbaak | magic_hat: dont use ${CALLERIDNUM}, it's deprecated. use ${CALLERID(num)} instead |
| 18:51.49 | *** join/#asterisk Greek-B0y (n=email@41.221.58.2) |
| 18:52.09 | magic_hat | mvanbaak: okay....so dtmfmode=IF("${CALLERID(num)}" = "7739891234"?inband:rfc2833) |
| 18:52.32 | mvanbaak | yup |
| 18:53.41 | *** join/#asterisk CCFL_Man2 (i=bbc01338@pool-70-105-198-92.scr.east.verizon.net) |
| 18:54.14 | magic_hat | cool...thx. |
| 18:54.44 | dacs | my phone has 1/8000 of a second delay |
| 18:54.54 | dacs | hahahahha |
| 18:54.56 | Qwell | dacs: I'm quoting you |
| 18:55.11 | dacs | Qwell: no i was just kidding |
| 18:55.25 | dacs | giving someone here something to bit on |
| 18:55.32 | dacs | s/bit/bite/ |
| 18:55.55 | TJNII | I'm getting app_meetme.c:772 build_conf: Unable to open pseudo device |
| 18:56.07 | TJNII | It is a permission issue, I assume with the zaptel drivers |
| 18:56.17 | TJNII | But I can't seem to find which files I need to chown |
| 18:57.01 | hmmhesays | I'm guessing you didn't compile as root |
| 18:57.11 | TJNII | I did, actually |
| 18:57.19 | TJNII | The problem is I don't want to run * as root |
| 18:57.27 | hmmhesays | I ee |
| 18:57.29 | hmmhesays | *see |
| 19:00.09 | *** join/#asterisk Wall (n=sceptik@200.70.24.107) |
| 19:00.13 | Wall | hola |
| 19:00.25 | Wall | alguno que escriba español ? :) |
| 19:00.47 | TJNII | Oooh, is the psude device accessed through /dev? |
| 19:03.05 | badcfe | i just dont get the sox man page .. anyonw know how i convert _from_ the format of "RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 44100 Hz" ? |
| 19:08.40 | TJNII | Yea, it was an ownership of /dev/zap issue |
| 19:10.44 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83.static-pool-4.pool.hargray.net) |
| 19:12.26 | nny_1 | wonder if anyone has tried to make a hold while on hold app.. guess it would be hard to distinguish between the two.. (thinking a way to put a call on hold if the other side has you on hold, and notify when the call is back) |
| 19:15.28 | hmmhesays | nny_1 it is a mess |
| 19:15.50 | hmmhesays | we need a manager event when a call puts another call on hold |
| 19:15.59 | hmmhesays | which at this point does not exist |
| 19:16.20 | nny_1 | hmmhesays: yeah thats what i was thinking |
| 19:16.30 | nny_1 | hmmhesays: no big deal, just came up in a situation |
| 19:16.37 | hmmhesays | however right now at least when I put a call on hold with my polycoms there is no manager event |
| 19:17.20 | hmmhesays | if there was we could easily send a notification when someone who has put you on hold has taken you off hold, obviously asterisk knows |
| 19:17.31 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 19:17.32 | nny_1 | yeah |
| 19:17.42 | nny_1 | no real audio cues, would be a load of false positives |
| 19:18.04 | mvanbaak | nny_1: I think there was some sample on the wiki that does that |
| 19:18.58 | mvanbaak | http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+reverse+hold |
| 19:19.03 | mvanbaak | there it is |
| 19:19.09 | *** join/#asterisk tsabi (n=tsabi@gw.creditexpress.hu) |
| 19:20.33 | badcfe | with the sox convertion program, is one to specify -A in order to get the alaw as read by asterisk? |
| 19:20.51 | hmmhesays | i would have to remap my hold key |
| 19:22.27 | nny_1 | cool reading |
| 19:22.56 | nny_1 | ahh |
| 19:23.13 | nny_1 | nice engineering skills on whomever wrote that |
| 19:23.20 | nny_1 | i was overthinking the solution |
| 19:23.34 | mvanbaak | I like it |
| 19:24.39 | nny_1 | gonna set that up and test it |
| 19:26.38 | robeph | what could cause something like "WARNING[4537]: app_meetme.c:1541 conf_run: Unable to write frame to channel: Inappropriate ioctl for device" just anything come to mind readily? |
| 19:27.31 | nny_1 | just got our new in house pbx stuff.. dual core amd 4000 1 GB RAM 2u rackmount etc. etc. Gonna make it a gateway firewall with Snort ID and IP |
| 19:30.11 | elverkilde | badcfe: still stuck? |
| 19:30.33 | *** part/#asterisk Wall (n=sceptik@200.70.24.107) |
| 19:30.59 | nny_1 | wth is the name of the simple nice cdr analiyzer? |
| 19:31.01 | hmmhesays | you can run a lot of phones off that |
| 19:31.05 | nny_1 | er analyzer* |
| 19:31.08 | hmmhesays | asterisk-stat-v2 |
| 19:31.49 | nny_1 | hmmhesays: thanks |
| 19:32.22 | *** part/#asterisk fiXXXerMet (n=meep@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 19:32.29 | hmmhesays | np |
| 19:32.57 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 19:34.16 | deeperror | does asterisk need to be restarted every few days still? |
| 19:35.29 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 19:36.07 | elverkilde | badcfe: try: sox foo-in.wav -r 8000 -c 1 -s -2 foo-out.wav resample -ql |
| 19:37.45 | iratik | seems that asterisk is listening for IAX2 "asterisk 11646 asterisk 16u IPv4 167147 UDP *:iax |
| 19:37.45 | iratik | " but that doesn't tell me the port number its listening for |
| 19:37.47 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 19:37.52 | iratik | how can I be sure that its listening to the right port? |
| 19:38.32 | [TK]D-Fender | iratik: "netstat -an|grep UDP |
| 19:38.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:39.35 | iratik | hmmmm... might be different |
| 19:39.44 | deeperror | anyone ever seen 'core show channels' return thousands of calls and 'sip show channels' returns back the correct amount? I restarted the server and everything seems to work fine so far any clues? |
| 19:39.48 | iratik | I found out that lsof gets its port aliases from /etc/services |
| 19:42.02 | [TK]D-Fender | iratik: I seriously doubt you have an iddue there... |
| 19:42.07 | [TK]D-Fender | isssue* |
| 19:42.19 | iratik | No I don't |
| 19:42.29 | iratik | I'll bother the voicepulse people about it |
| 19:43.14 | [TK]D-Fender | iratik: Whats the problem? |
| 19:43.28 | l0verb0y | hey does anyone know how to change the name of voicemail folders in asterisk? |
| 19:43.28 | iratik | Dialing a DID and not getting anything |
| 19:43.34 | iratik | nada..... |
| 19:43.40 | [TK]D-Fender | jblack: You around? |
| 19:44.03 | [TK]D-Fender | l0verb0y: "man mv" <- |
| 19:44.20 | l0verb0y | you can just mkdir or mv a dir and asterisk doesn't care? |
| 19:44.37 | [TK]D-Fender | l0verb0y: Of course ti will, but that depends on what you're changing it to. |
| 19:44.50 | [TK]D-Fender | l0verb0y: And the full intent of your original question. |
| 19:47.03 | iratik | Just because 4569 is open and listening... that might not necessarily mean that something isn't blocking that port.... I tried to telnet into port 4569 and got connection refused..... does this mean that if my IAX2 provider tried to contact me on 4569... they might get connection refused? |
| 19:48.43 | [TK]D-Fender | iratik: Telnet was never going to work... |
| 19:49.20 | mvanbaak | iax is udp |
| 19:49.33 | [TK]D-Fender | mvanbaak: SHH! |
| 19:49.42 | l0verb0y | ahh thanks |
| 19:50.12 | iratik | how can i test if my box is receiving iax2 connections properly .........oh.... connect an iax softphone via the external ip |
| 19:50.38 | [TK]D-Fender | iratik: Yeah actually trying to use it would be a good idea. |
| 19:50.51 | mvanbaak | nah |
| 19:50.56 | mvanbaak | use sip |
| 19:51.01 | [TK]D-Fender | iratik: have you even checked to see if * was getting packets? |
| 19:51.04 | outtolunc | how can i tell if a car will fit in my living room... drive one into it |
| 19:51.23 | [TK]D-Fender | outtolunc: As you wish ;) |
| 19:51.31 | Jam0r | measure the car, and the space it has to fit into would be easier ;) |
| 19:51.34 | TJNII | outtolunc: Well, it would workm wouldn't it? |
| 19:51.36 | Jam0r | saves getting stuck |
| 19:51.42 | outtolunc | <G> |
| 19:54.46 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 19:55.18 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 19:56.46 | *** join/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 19:57.23 | jochien1 | hi hw do i set postfix to send voicemail to my extension holders? |
| 19:57.35 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 19:58.47 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 19:59.00 | *** join/#asterisk mcargile (n=mcargile@rrcs-24-173-134-222.se.biz.rr.com) |
| 19:59.54 | hmmhesays | I cannot get this freaking presence to use my 3rd sip account |
| 19:59.58 | hmmhesays | and it is really irritating me |
| 20:00.29 | Netgeeks | if you have two entries for the same channels in zapata which actually is used, the first or the second? |
| 20:00.58 | Netgeeks | in other words... group =0 channel => 1-23 followed by group=1 channel => 1-23 |
| 20:01.19 | Netgeeks | does one take precedence over the other, or did I just put the same channels in group 0 and 1? |
| 20:01.44 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:04.21 | jameswf | Netgeeks: the second will override the first |
| 20:04.35 | Netgeeks | jameswf: thanks |
| 20:04.38 | [TK]D-Fender | jochien1: Since when does an e-mail server send VOICEMAIL to *? |
| 20:13.27 | *** join/#asterisk saftsack (n=saftsack@p4FC761DA.dip.t-dialin.net) |
| 20:13.30 | [TK]D-Fender | <PROTECTED> |
| 20:13.44 | [TK]D-Fender | </selfish> |
| 20:13.44 | elverkilde | clever... |
| 20:14.25 | deeperror | is zapbarge and chanspy loaded when asterisk loads or are these just functions that can be used? Any way to make sure these features are diabled? |
| 20:15.14 | elverkilde | [TK]D-Fender: what's your time? u seem to be here all hours... |
| 20:15.52 | [TK]D-Fender | elverkilde: EST (GMT -5) |
| 20:17.29 | *** join/#asterisk saftsack (n=saftsack@p4FC761DA.dip.t-dialin.net) |
| 20:18.14 | *** join/#asterisk apocn (n=htejeda@unaffiliated/apocn) |
| 20:18.59 | apocn | Hello all, I want to install a second instance of asterisk (on a new folder (e.g.: asterisk2)) and I dont want it to have conflicts with the current binary... what option should I pass to ./configure ? |
| 20:20.23 | [TK]D-Fender | apocn: Good luck.... * will fight over ports and any zaptel hardware..... |
| 20:21.35 | apocn | well its just changing 5060 to 5070, no hardware on this pc |
| 20:21.48 | apocn | a softswitch sends packets directly to this pc, it doesnt have any hardware on it |
| 20:22.14 | tzafrir | apocn, point it to a different config set with -c |
| 20:22.46 | apocn | tzafrir: the currently installed one? |
| 20:23.03 | tzafrir | you can configure everything at run time, no need to set things at configure time |
| 20:23.08 | apocn | I can run two instances of the same binary pointing to different config |
| 20:23.38 | tzafrir | apocn, I do so regularily, but in a way intended only for testing: |
| 20:23.45 | apocn | yes, its only for testing |
| 20:24.58 | tzafrir | http://bugs.digium.com/11680 . But as I usually need to test zaptel hardware, I just kill the main asterisk and dodge the problem |
| 20:25.12 | apocn | ok |
| 20:25.23 | tzafrir | Feel free to try to automate using different ports |
| 20:26.03 | tzafrir | or edit the config manually |
| 20:26.09 | apocn | cool |
| 20:28.36 | drmessano-LT | Friend if mine asked me if I can make a PAP2 make him a sandwich |
| 20:28.43 | drmessano-LT | PAP2: Sudo make me a sandwich |
| 20:29.21 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 20:30.08 | [TK]D-Fender | drmessano-LT: Possible.... |
| 20:30.23 | dacs | what does it mean, my softphone can ring my ata , the other end can hear me , but ican't hear them |
| 20:30.30 | puppet | gah hard to find firmware for the vood322 |
| 20:30.50 | [TK]D-Fender | dacs: Typical side effect of running * from behind NAT and not being set up properly |
| 20:32.55 | dacs | [TK]D-Fender: is it at my end or the other end |
| 20:33.31 | [TK]D-Fender | dacs: You tell me. Is * behind NAT? |
| 20:34.12 | dacs | [TK]D-Fender: yes , but if the other end can ring me what does that mean |
| 20:34.21 | [TK]D-Fender | dacs: Means nothing. |
| 20:34.30 | [TK]D-Fender | dacs: Now go follow the guide : |
| 20:34.31 | [TK]D-Fender | ~sipnat |
| 20:34.32 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:34.52 | drmessano-LT | dacs: did you change those settings I added to sip.conf? |
| 20:35.10 | dacs | drmessano-LT: no, |
| 20:35.12 | drmessano-LT | externhost, localnet |
| 20:36.09 | drmessano-LT | You didnt edit those? |
| 20:36.31 | dacs | drmessano-LT: to what? |
| 20:37.13 | drmessano-LT | Have you edited them in any way? |
| 20:37.30 | dacs | drmessano-LT: no! |
| 20:38.11 | drmessano-LT | Ports still open? |
| 20:38.37 | dacs | yes |
| 20:39.37 | drmessano-LT | Where is this X-lite phone at? |
| 20:39.40 | drmessano-LT | Offsite? |
| 20:40.27 | dacs | drmessano-LT: yes |
| 20:40.47 | drmessano-LT | nat=yes on that extension? |
| 20:41.00 | dacs | yes |
| 20:41.15 | drmessano-LT | Hmm |
| 20:41.20 | drmessano-LT | It may be broke |
| 20:42.15 | drmessano-LT | You still running the no-ip app? |
| 20:42.22 | dacs | yes |
| 20:42.41 | dacs | the other part can dail me |
| 20:42.46 | dacs | they can hear me |
| 20:42.49 | dacs | but i can't |
| 20:44.12 | drmessano-LT | Id go over that NAT guide again..It SOUNDS like you have it all configged, but something is obviously missing |
| 20:44.21 | drmessano-LT | Id double check all you told me too |
| 20:44.26 | drmessano-LT | Something isnt right |
| 20:44.30 | drmessano-LT | It SHOULD be working |
| 20:44.51 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:44.51 | variable_office | would .3% packet loss be enough to be noticeable? |
| 20:45.21 | elverkilde | dacs: can u call the other part directly? |
| 20:45.29 | *** join/#asterisk lzhang (n=lzhang@67.95.13.186) |
| 20:45.43 | *** join/#asterisk apocn (n=htejeda@unaffiliated/apocn) |
| 20:45.47 | lzhang | hey guys, what are some commands I can do in the cli to check if my t1 pri is hooked up correctly |
| 20:46.23 | apocn | Hello again, I've used the live_ast and it installed well under the "live" subfolder. But when I try using ./live_ast run I get the error: Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect. |
| 20:46.23 | lzhang | I'm running version 1.4.17 |
| 20:46.27 | apocn | any hints? |
| 20:48.19 | dacs | elverkilde: * show the user is unreachable , but i was able to talk |
| 20:48.50 | *** join/#asterisk ZX81 (n=ZX81@202.49.106.158) |
| 20:49.29 | elverkilde | dacs: I was thinking you could dial outside *, to check if thats where its wrong.. just an idea :) |
| 20:50.18 | elverkilde | dacs: (I mean bypassing *) |
| 20:51.36 | *** part/#asterisk codestr0m (n=asura@ip5451d5cd.direct-adsl.nl) |
| 20:51.38 | [TK]D-Fender | elverkilde: Nope.... |
| 20:51.55 | [TK]D-Fender | all so very sad... |
| 20:52.09 | elverkilde | [TK]D-Fender: what? |
| 20:52.25 | [TK]D-Fender | elverkilde: Wrong path to debugging.... |
| 20:52.38 | drmessano-LT | Note to channel |
| 20:52.55 | drmessano-LT | ONE WAY AUDIO DOES NOT MEAN "HE HEAR ME GOOD, BUT HE SOUND BAD TO ME" |
| 20:53.01 | drmessano-LT | Thats "One way internet" |
| 20:54.25 | elverkilde | [TK]D-Fender: u da boss :-D |
| 20:54.42 | fugitivo | drmessano-LT: one way audio = i can talk to god but he doesn't reply, what did i do wrong? |
| 20:56.29 | drmessano-LT | OFL |
| 20:56.32 | drmessano-LT | ROFL too |
| 20:56.39 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 20:56.56 | drmessano-LT | He forwarded his calls to DOG |
| 20:57.00 | drmessano-LT | pwn3d |
| 20:57.05 | jblack | drmessano-LT: I succeded in getting faxes. :) |
| 20:57.22 | drmessano-LT | Awesome.. Whats a Fax? |
| 20:58.02 | [TK]D-Fender | drmessano-LT: Fax of life are all about you! |
| 20:58.19 | drmessano-LT | Oh |
| 20:58.20 | drmessano-LT | Duh |
| 20:58.35 | drmessano-LT | A fax is a modem thing |
| 20:58.39 | drmessano-LT | I had one of those once |
| 20:59.03 | elverkilde | I guess I'm not old enough... |
| 21:00.37 | robeph | fax = facsimile its just a point to point copier usually via analogue mod/dem... |
| 21:00.39 | fugitivo | fax is that thing that you can't hack because it breaks |
| 21:00.40 | robeph | :-d |
| 21:02.58 | *** part/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 21:03.00 | apocn | using live_ast, when I do ./live_ast run, it creates the files asterisk.ctl and asterisk.pid but when I try using ./live_ast run -r I get the error: Unable to connect to remote asterisk (does /root/asterisk-debug/asterisk-1.4.4/live/var/run/asterisk.ctl exist?) |
| 21:03.08 | apocn | any help? |
| 21:04.11 | Qwell | tzafrir: ^^ |
| 21:04.35 | tzafrir | apocn, this means asterisk isn't running |
| 21:04.45 | tzafrir | or at least that local asterisk instance |
| 21:04.49 | apocn | ok |
| 21:05.03 | tzafrir | did you run: ./live_ast run |
| 21:05.10 | apocn | yes |
| 21:05.11 | tzafrir | or: ./live/asterisk |
| 21:05.17 | apocn | ./live_ast run |
| 21:05.29 | apocn | then it created these two files normally (no error appeared, etc..) |
| 21:05.31 | tzafrir | if so, check the logs: live/var/log/asterisk/messages |
| 21:05.37 | apocn | but when I tried to get into the console ./live_ast run -r |
| 21:05.39 | apocn | ok, let me see |
| 21:05.53 | *** part/#asterisk elverkilde (n=jon@h55eb18aa.c45-01-09.dyn.perspektivbredband.net) |
| 21:05.57 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-74-61.tor.primus.ca) |
| 21:06.00 | apocn | uh |
| 21:06.02 | apocn | [Jan 14 17:01:25] ERROR[13209] pbx_dundi.c: Unable to bind to 0.0.0.0 port 4520: Address already in use |
| 21:06.07 | apocn | didng know about this port |
| 21:06.41 | apocn | dundi... |
| 21:06.43 | [TK]D-Fender | apocn: NEXT!!!!@!@ (C) BKW |
| 21:07.16 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:07.35 | apocn | ok, worked now |
| 21:07.36 | apocn | thanks |
| 21:08.21 | asr33 | does anyone here use a type of solid state hard drive with Asterisk? |
| 21:09.22 | asr33 | what kind is reliable or is compact flash the way to go? |
| 21:09.39 | drmessano-LT | SSHD is far better |
| 21:10.00 | asr33 | what brand? |
| 21:10.15 | drmessano-LT | Not sure who has the best one right now |
| 21:11.35 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 21:16.05 | asr33 | http://www.tigerdirect.ca/applications/SearchTools/item-details.asp?EdpNo=330278 |
| 21:16.08 | asr33 | 7&CatId=2503 |
| 21:16.17 | asr33 | sorry |
| 21:16.39 | asr33 | does this look any good? |
| 21:17.12 | asr33 | #http://www.tigerdirect.ca/applications/SearchTools/item-details.a |
| 21:17.25 | asr33 | kick I deserve it |
| 21:17.39 | asr33 | anyway thanks for your help |
| 21:17.43 | [TK]D-Fender | asr33: No, thats far from kick-worthy... |
| 21:18.05 | *** join/#asterisk BBHoss (n=hoss@c-71-207-173-38.hsd1.al.comcast.net) |
| 21:18.06 | robeph | what would be the best method for determining the cause of this error? app_meetme.c:1541 conf_run: Unable to write frame to channel: Inappropriate ioctl for device |
| 21:18.16 | [TK]D-Fender | asr33: I would probably suggest a SSD drive that lets you plug in a REMOVABLE Flash source.... |
| 21:18.58 | asr33 | [TK]D-Fender: like a usb thumbdrive? |
| 21:19.15 | asr33 | I have a usb port on my soekris |
| 21:19.34 | [TK]D-Fender | asr33: Something like that, yes |
| 21:20.04 | asr33 | that would be very much cheaper |
| 21:20.58 | asr33 | great have you ever done this [TK]D-Fender ? |
| 21:21.14 | [TK]D-Fender | asr33: a question of cost effectiveness and reusability. a fixed function 1-piece flash SSD is a dead-end. On that lets you plug in your own cards/etc is far more valuable in the long-term (maybe even shorter) |
| 21:21.32 | [TK]D-Fender | asr33: No, but like a million other things, I think about constantly ;) |
| 21:21.45 | puppet | rofl alot of sites in sweden getting hacked ;P |
| 21:21.46 | puppet | one biig picture/diary site, on partypicture site, tv3 page got hacked too, and then aftonbladet (newspaper) got leaked passes to there email. and then some stupid cop had saved material in a ongoing investigation on his private gmail account, that he had same pass on as the leaked one from the picture/diary site. |
| 21:23.36 | twisted | mein Milchshake holt alle Jungen zum Rasen |
| 21:24.04 | [TK]D-Fender | twisted: Damn right.... |
| 21:24.09 | drmessano-LT | Hell yes lol |
| 21:24.42 | hmmhesays | [TK]D-Fender: guitar world reader poll, guitar youtube video of 2007 andy mckgee drifting |
| 21:24.56 | [TK]D-Fender | hmmhesays: yay :) |
| 21:24.59 | twisted | [TK]D-Fender: verfluchtes Recht, ist es besser als Ihr |
| 21:25.03 | [TK]D-Fender | hmmhesays: I love my new acoustic btw... |
| 21:25.12 | murdmath | asr33: I have two systems with these in them. So far so good: http://www.innodisk.com/industrial/edc2k+.htm |
| 21:25.27 | cappiz | how can i make certain devices/extensions use different outbound routes=? |
| 21:25.33 | [TK]D-Fender | hmmhesays: I have decent access through 19th fret :) |
| 21:25.47 | asr33 | murdmath: thanks |
| 21:25.48 | robeph | mmm milkshakes... |
| 21:25.55 | [TK]D-Fender | cappiz: assign them different contexts in your dialplan. |
| 21:25.59 | drmessano-LT | I need a damn beer |
| 21:26.08 | cappiz | [TK]D-Fender oki:) |
| 21:26.22 | drmessano-LT | and a weekend of my life back |
| 21:26.39 | [TK]D-Fender | drmessano-LT: Would you like fries with that, sir? |
| 21:26.50 | [TK]D-Fender | drmessano-LT: If so.... I think I know a waiter ;) |
| 21:27.36 | [TK]D-Fender | alrighty... check-out time.... BBIAB |
| 21:30.00 | nny_1 | heh this is the strangest thing i have ever seen |
| 21:30.05 | nny_1 | so we ordered a 2u case |
| 21:30.25 | nny_1 | and (facing the rear) the screws and bracket spots for the pci card are on the RHS |
| 21:30.40 | nny_1 | the #2 slot is where the riser card is supposed to go |
| 21:30.52 | nny_1 | which puts the screw and angle ont he pci card on the LHS |
| 21:30.53 | jblack | Is anyone else in a dundi network? |
| 21:30.58 | nny_1 | with no apparent way to mount the pci card |
| 21:31.19 | nny_1 | i am looking at ribbon cable pci risers |
| 21:31.34 | nny_1 | but seriously, wtf was the engineer(s) thinking |
| 21:33.41 | BBHoss | jblack: you got dundi working? |
| 21:34.06 | jblack | <PROTECTED> |
| 21:34.37 | BBHoss | thats cool, i've never had the balls to get dundi working |
| 21:34.42 | drmessano-LT | jblack: But, no friends? |
| 21:34.45 | BBHoss | much less had someone else to peer with |
| 21:35.11 | jblack | drmessano: As always. :) |
| 21:35.26 | drmessano-LT | lol |
| 21:35.36 | drmessano-LT | HAH.. Trixbox is gonna put ads in the GUI |
| 21:35.43 | jblack | Seriously? |
| 21:35.54 | puppet | HAHAHA |
| 21:35.55 | drmessano-LT | Theyre "Talking" about it |
| 21:36.10 | drmessano-LT | Which means, "next version, checking for damage first" |
| 21:36.14 | jblack | Trixbox is gpl, right? So what stops everyone from yanking them out. |
| 21:36.27 | drmessano-LT | Nothing, its the principle |
| 21:36.28 | puppet | well |
| 21:36.34 | puppet | time to change the .org to .com? |
| 21:36.36 | drmessano-LT | Phoning home, ads.. good stuff |
| 21:36.48 | Greek-B0y | i dont get it. I put asterisk in my run levels using make install conf but its not auto starting. starting it manually works fine. |
| 21:36.54 | drmessano-LT | Its Tricksbox alright |
| 21:37.06 | drmessano-LT | "Look, I can make a popup!" |
| 21:37.14 | drmessano-LT | *POOF* |
| 21:37.30 | drmessano-LT | "Look, stole some stats from you!" |
| 21:37.32 | drmessano-LT | *POOF* |
| 21:37.40 | jblack | There's too many * forks anyways. Suicide is as successful method of rectifying that as any |
| 21:37.54 | puppet | jblack: trixbox isnt a fork is it? |
| 21:37.57 | drmessano-LT | Forks isnt the issue |
| 21:37.57 | puppet | isnt trixbox just a gui? |
| 21:37.58 | drmessano-LT | No |
| 21:38.04 | jblack | Oh, it's not? |
| 21:38.06 | drmessano-LT | Trixbox is FreePBX + Poo |
| 21:38.18 | drmessano-LT | Trixbox is Asterisk+ FreePBX + Poo |
| 21:38.26 | drmessano-LT | In a nice bundle |
| 21:38.40 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:38.42 | twisted | voulez-vous détruire mon chandail ? |
| 21:38.55 | drmessano-LT | Full of damn problems |
| 21:39.22 | drmessano-LT | Cant even Yum update without crashing the pinball machine at the bar down the street |
| 21:39.24 | drmessano-LT | "oops" |
| 21:39.35 | jblack | Heh. I hae what I want with * already, except for sending faxes. And I know how to do that. |
| 21:39.48 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.100.130) |
| 21:43.40 | robeph | wait you trying to send faxes digitally or analogue over voip |
| 21:44.19 | robeph | I stopped using faxes when I could email attachments =s |
| 21:44.31 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:44.41 | jblack | i'll send them over voip, of course |
| 21:44.57 | robeph | jblack: don't confuse trixbox with fluxbox =p |
| 21:45.04 | cappiz | haha |
| 21:45.26 | jblack | I need to send/receive faxes 2-3 times a year. Each time, it costs 5 bucks and a trip down to officemax. |
| 21:45.59 | jblack | Now, all the faxes I could get for nothing, delivered to my mailbox in pdf format. |
| 21:46.07 | jblack | Thanks * and ipkall. ;) |
| 21:46.11 | robeph | jblack: could you not simply use a asterisk -> rj11 -> fax/modem + software? |
| 21:46.18 | robeph | that really sounds ugly :( |
| 21:46.38 | jblack | robeph: With an ata that works well with t.38, yeah. That's how I plan on sending them, in fact. |
| 21:46.48 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 21:47.33 | robeph | that'd be a nice plug in though, something to automagically receive faxes without having to resort to external software for a* |
| 21:48.03 | jblack | That's basically what I have now. |
| 21:48.08 | ZPertee | does anyone know where I can find information on linkys voip gateway codes? I am trying to control the mwi dial tone on the spa8000 voip gateway and in the text box beside it says (350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2) |
| 21:48.39 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 21:48.50 | jblack | zpertee: I haven't looked at those tones yet. Did you check the book? |
| 21:49.01 | jblack | There's a big book for the linksys atas at linksys.com |
| 21:49.54 | *** join/#asterisk ToTo (n=ToTo@host126-207-dynamic.2-87-r.retail.telecomitalia.it) |
| 21:49.57 | robeph | ls |
| 21:49.59 | robeph | oops. |
| 21:50.01 | ZPertee | jblack. ok Im new to this but I'll try to see if I can find that book |
| 21:50.32 | jblack | zpertee: Find the product page for the spa 8000 on linksys.com. There's a link for the pdf. It covers everything, including the internal dialplans |
| 21:50.48 | jblack | I can try dcc'ing it to you if you like |
| 21:50.50 | puppet | anyone succeded in reflashing a vood322 to sip? |
| 21:51.57 | jblack | zpertee: Let me know if dcc didn't work |
| 21:52.29 | ZPertee | it didn't not sure if that is because of the inability of my irc or not |
| 21:52.41 | jblack | it's proably me. I'm behind nat |
| 21:52.58 | *** join/#asterisk ToTo (n=ToTo@host126-207-dynamic.2-87-r.retail.telecomitalia.it) |
| 21:53.41 | jblack | Let me reconnect |
| 21:54.14 | *** join/#asterisk jblack (n=jblack@pool-71-181-138-192.sctnpa.east.verizon.net) |
| 21:54.19 | jblack | who was I talkign with? |
| 21:54.44 | ZPertee | me |
| 21:54.56 | *** join/#asterisk phocus (n=phocus@www.healthtech.net) |
| 21:55.30 | jblack | try /dcc get jblack |
| 21:55.41 | jblack | There it goes |
| 21:55.42 | phocus | does anyone know if there is a plug in for Vista meda center , to do caller id with asterisk, maybe even voip |
| 21:55.50 | *** part/#asterisk asr33 (n=asr33@dsl-207-112-74-61.tor.primus.ca) |
| 21:55.51 | ZPertee | got it thanks a lot jblack |
| 21:56.28 | *** join/#asterisk pLr (n=bobo@unaffiliated/plr) |
| 21:57.23 | fujin | phocus: good luck with that buddy |
| 21:58.28 | phocus | thanks, i dont really want to have to write it myself |
| 21:58.31 | phocus | but it looks like i will |
| 21:58.34 | twisted | phocus: http://vistamccallerid.oabsoftware.nl/ |
| 21:59.19 | fujin | lol |
| 21:59.21 | fujin | how about that^ |
| 21:59.42 | *** join/#asterisk Shaun2222 (n=Shaun222@ip68-4-127-67.oc.oc.cox.net) |
| 22:00.07 | twisted | run client and server on vmc, use TAPI for asterisk to get data to it :) |
| 22:00.09 | twisted | simple as cake. |
| 22:00.13 | jblack | Ohh. I recorded "Terminator: The Sarah Connor Chronicles" last night. |
| 22:00.29 | phocus | not a bad idea |
| 22:00.42 | jblack | please don't suck. please don't suck. please please please |
| 22:00.55 | Greek-B0y | which run levels should asterisk run in? |
| 22:01.01 | fujin | jblack: I got the 720p ac3 5.1 |
| 22:01.06 | fujin | haven't watched it yet either |
| 22:01.24 | jblack | greek-boy: That depends upon your preference and the way your distro is designed. |
| 22:01.33 | Greek-B0y | jblack i'm on debian |
| 22:01.37 | jblack | I pull it up in runlevel 2, along with most other system installed things. |
| 22:01.48 | jblack | Odds are it'll default to starting in runlevel 2 |
| 22:02.06 | jblack | Now shush. You're interrupting Sarah connor. She'll terminate you. |
| 22:02.11 | twisted | phocus: another idea: http://www.vistacallerid.com/ <-- install a tapi driver on VMC directly, and choose it as your modem in this software |
| 22:02.15 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:02.35 | Greek-B0y | update-rc.d asterisk start 75 2 3 4 5 . |
| 22:02.37 | phocus | trying to stay away from modem |
| 22:02.39 | Greek-B0y | will that do it? |
| 22:02.47 | Shaun2222 | is this normal... I have a macro that announces the caller to the callee, the caller chooses a extension, says his name, gets put into a queue. The queue then starts calling agents using dial+macro. The agent awnsers the phone and while the macro is playing asterisk times out the call saying nobody answered... shouldnt it wait for the macro? |
| 22:03.09 | *** part/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 22:04.16 | Greek-B0y | [TK]D-Fender u know everything, so let me ask u |
| 22:04.28 | Greek-B0y | i want to add asterisk to the run levels in asterisk |
| 22:04.33 | Greek-B0y | update-rc.d asterisk start 75 2 3 4 5 . |
| 22:04.36 | Greek-B0y | is that good enough? |
| 22:05.54 | drmessano-LT | 14 Jan 2008 - Asterisk becomes self aware |
| 22:06.47 | [TK]D-Fender | Greek-B0y, Don't see why not... |
| 22:07.53 | [TK]D-Fender | drmessano-LT, 14 Feb - Qwell's year-old chan_skinny bot-net launches Russian nukes at the Mediterranean island of Mepos... |
| 22:08.52 | Shaun2222 | [TK]D-Fender: any idea about my q above/ |
| 22:09.00 | drmessano-LT | LOL |
| 22:09.06 | drmessano-LT | Poor Balki |
| 22:09.32 | [TK]D-Fender | Shaun2222, Of course not... |
| 22:09.36 | drmessano-LT | Now Cousin Larry will never get rid of him |
| 22:09.56 | Shaun2222 | [TK]D-Fender: umm, course not you dont know, or course not it shouldnt wait |
| 22:10.40 | [TK]D-Fender | Shaun2222, Of course I have no idea... you haven't SHOWN me anything. |
| 22:11.02 | Shaun2222 | [TK]D-Fender: you know you have a serious hard-on for pastebin you know that ;) |
| 22:11.33 | Shaun2222 | one sec, let me put a presentation together for you :)) |
| 22:13.16 | jblack | I'm going through the front of the phone book, checking out the rates for lcoal service. |
| 22:13.44 | jblack | local, that is. It's frigging amazing how much they try to charge. 16 cents a minute for a call that's 1 mile away? |
| 22:14.09 | Shaun2222 | [TK]D-Fender: http://pastebin.ca/855281 |
| 22:14.41 | [TK]D-Fender | jblack, don't ask about their LONG DISTANCE rates ;) |
| 22:15.00 | [TK]D-Fender | Shaun2222, half-way there |
| 22:15.06 | Shaun2222 | bah, sorry logs.. |
| 22:15.18 | jblack | That's just incredible. 16 cents for 1 minute, then 9 cents a minute after... |
| 22:15.21 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 22:15.32 | drmessano-LT | jblack: I hear Skype is good |
| 22:15.38 | jblack | I can call... brazil for less then that. |
| 22:15.43 | tzafrir | Greek-B0y, why not use defaults? |
| 22:15.57 | jblack | drmessano-LT: I came to * from skype. They're nudging up a little. Playing billing games. |
| 22:16.41 | jblack | With callwithus.. I think I'm paying 1.27 cents a minute |
| 22:16.54 | tzafrir | Greek-B0y, update-rc.d asterisk defaults # or maybe: defaults 5 30 # or so |
| 22:17.20 | *** join/#asterisk metfan2007 (n=metfan20@201.103.91.58) |
| 22:17.47 | metfan2007 | anyone with experience using Asterisk and H.323??? pls... |
| 22:18.01 | Shaun2222 | [TK]D-Fender: http://pastebin.ca/855288 |
| 22:18.27 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 22:18.27 | jblack | metfan2007: I haven't used it, just know it's possible. h.323 is the old way, as I understand things |
| 22:19.25 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 22:19.25 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 22:19.29 | Greek-B0y | alright tzafrir. and i'll run safe_asterisk too, i'll put it into /etc/rc.local |
| 22:19.38 | Greek-B0y | but this is what the startup script outputs |
| 22:19.39 | Greek-B0y | cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory |
| 22:19.39 | Greek-B0y | Automatically restarting Asterisk. |
| 22:19.45 | metfan2007 | jblack, I know that, I'm working in a trunk between an Avaya and Asterisk, but I cannot dial from Asterisk to Avaya, from Avaya to Asterisk everything is Ok |
| 22:19.57 | tzafrir | Greek-B0y, no. Choose one and use it. |
| 22:20.13 | tzafrir | Otherwise when you need to disable it, you'll have no idea what to disable |
| 22:20.31 | Greek-B0y | i chose one |
| 22:20.37 | Greek-B0y | but the script itself outputs that |
| 22:20.42 | Greek-B0y | instead of starting asterisk :( |
| 22:21.09 | Greek-B0y | and also says |
| 22:21.09 | Greek-B0y | Asterisk ended with exit status 127 |
| 22:21.09 | Greek-B0y | Asterisk died with code 127. |
| 22:22.04 | tzafrir | Greek-B0y, see link to Debian init.d script in http://bugs.digium.com/9843 |
| 22:22.08 | tzafrir | (last comment) |
| 22:22.13 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 22:22.32 | [TK]D-Fender | Shaun2222, Executing [s@macro-announcecaller:4] WaitExten("SIP/306-1ef10610", "30") in new stack |
| 22:22.39 | [TK]D-Fender | Shaun2222, Nobody picked up in 30000 ms |
| 22:22.56 | tripps | ~where Qwell |
| 22:22.56 | jbot | [qwell] a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 22:23.01 | Shaun2222 | ya, it says that while 306 is picked up... just the macro is running.. |
| 22:23.04 | tripps | mm |
| 22:23.04 | [TK]D-Fender | Shaun2222, Was this 30s from the start of Waitexten, or 30s from start of ringing? |
| 22:23.14 | tripps | Qwell you around |
| 22:23.17 | Qwell | no |
| 22:23.19 | tripps | 'heh :) |
| 22:23.23 | Qwell | oh, guess ia m |
| 22:23.24 | Shaun2222 | [TK]D-Fender: has to be queue... |
| 22:23.24 | Qwell | I am* |
| 22:23.34 | [TK]D-Fender | Shaun2222, Because at the same time you don't seem to have answered the "Waitexten" |
| 22:23.35 | Shaun2222 | [TK]D-Fender: happens few seconds after the macro starts.. |
| 22:23.45 | Greek-B0y | thanks tzafrir |
| 22:24.02 | Shaun2222 | [TK]D-Fender: right, thats because i was waiting for this error to happen. |
| 22:24.07 | [TK]D-Fender | Shaun2222, perhaps you should change your queue ring time. |
| 22:24.10 | tripps | Qwell: any resolution to bug 0010712? |
| 22:24.22 | tripps | or any findings |
| 22:24.22 | Shaun2222 | [TK]D-Fender: another problem i'm having is when i push 2 it connects the call rather than goign through the loop i have setup in that macro. |
| 22:24.41 | Qwell | tripps: doesn't ring a bell |
| 22:25.03 | tripps | audio drops out over IAX2 trunks |
| 22:25.23 | Shaun2222 | [TK]D-Fender: well the problem with that is it will play forever and say voicemail pics up on a cell phone it's ringing or somthing it's going to play them a year long message.. |
| 22:25.39 | [TK]D-Fender | Shaun2222, CELL PHONE? |
| 22:25.46 | [TK]D-Fender | Shaun2222, Where is this? |
| 22:26.13 | Shaun2222 | [TK]D-Fender: this used to work, i'm almost positive... i wrote somthing simular long time ago with 1.2 but never implemented it, this is it but updated alittle bit. |
| 22:26.42 | Shaun2222 | [TK]D-Fender: shouldnt the queue see that the call was answered? |
| 22:26.45 | [TK]D-Fender | Shaun2222, Then it is still untested. |
| 22:27.02 | [TK]D-Fender | Shaun2222, Not until bridged I don't think. |
| 22:27.05 | Shaun2222 | [TK]D-Fender: cell phone part isnt there yet... but it will be. |
| 22:27.11 | tripps | Qwell: of course in my case the problem is on 1.2 - looking at http://bugs.digium.com/view.php?id=8325 now |
| 22:27.19 | [TK]D-Fender | Shaun2222, Queues weren't made with the idea that agents took calls when they felt like it. |
| 22:28.04 | Shaun2222 | [TK]D-Fender: ya, thats not realy what this is about, mostly to identify the caller and to send them to VM if needed. |
| 22:28.52 | drmessano-LT | Looks like theres some huge bug in tb thats nuking 2.4 systems running yum updates |
| 22:28.59 | Shaun2222 | [TK]D-Fender: also if i press 2, it bridges the call. |
| 22:29.05 | Shaun2222 | rather than going into that loop |
| 22:29.28 | [TK]D-Fender | Shaun2222, Do I see that? |
| 22:29.42 | Shaun2222 | nope, let me paste it |
| 22:30.05 | *** join/#asterisk magumbade (n=magumbad@ppp-82-135-5-188.dynamic.mnet-online.de) |
| 22:30.08 | drmessano-LT | going home.. poof |
| 22:31.43 | Shaun2222 | [TK]D-Fender: http://pastebin.ca/855304 |
| 22:33.35 | [TK]D-Fender | Shaun2222, I don't see #2 run.... |
| 22:34.10 | Greek-B0y | lol |
| 22:34.27 | Greek-B0y | i copied tzafrir's script from http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/asterisk.init?op=file&rev=0&sc=0 |
| 22:34.30 | Greek-B0y | pasted into vim |
| 22:34.42 | Shaun2222 | i hit it, and it bridged the calls... what else cased it to bridge? |
| 22:35.13 | [TK]D-Fender | Shaun2222, But your dialplan code for that exten is not executed. |
| 22:35.50 | *** join/#asterisk aiurea (n=k9@arcadia.timisoara.roedu.net) |
| 22:36.05 | aiurea | Hi |
| 22:36.35 | aiurea | is it possible to assign the return value of a System call to an asterisk variable and then use it with GotoIf calls? |
| 22:37.00 | Shaun2222 | hmm, ya maybe this is the pos linksys wifi phone i'm using for testing. |
| 22:37.10 | tripps | is it possible (or relevant) to show which build version an * box is from the CLI? For example, show version shows 1.2.23, but I want to see what revision number (e.g., 52264) a build was derived from for purposes of cross referencing bug reports |
| 22:37.18 | Shaun2222 | it's sending somthing... * must just not be reconizing it |
| 22:37.45 | Shaun2222 | what i dont get is why it's defaulting to bridge... |
| 22:37.50 | Shaun2222 | shouldnt i see the rest of the dial plan run... |
| 22:37.58 | Shaun2222 | maybe for invalid? |
| 22:39.14 | De_Mon | tripps yeah when you compile it |
| 22:39.51 | tripps | De_Mon: thanks - i'm looking at someone else's system . . . that information isn't available |
| 22:40.23 | De_Mon | somethin you have to add before compiling |
| 22:40.52 | tripps | De_Mon: in that case it would show up with show version? |
| 22:40.59 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:42.37 | magumbade | aiurea: it is possible to grab that value with an external script called by the AGI, save this value in the asterisk database and then work with gotoif |
| 22:43.04 | tripps | De_Mon: i've also got a box that I complied from tarball - I still have the source; how do I tell which revision that one is? |
| 22:43.13 | tripps | s/complied/compiled |
| 22:45.23 | De_Mon | im not sure where exactly you can modify the string 'show version' returns, but you have to update it by hand. |
| 22:45.45 | De_Mon | asterisk doesn't know what revision it was compiled from |
| 22:45.56 | De_Mon | you have to tell it what version it was compiled from |
| 22:49.42 | De_Mon | looks like a compiletime arg ASTERISK_VERSION |
| 22:52.11 | Shaun2222 | [TK]D-Fender: i just had it call my cell rather than that linksys phone... option 2 still doesnt work.. |
| 22:52.21 | Shaun2222 | it's like it doesnt know what to do. |
| 22:53.52 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:58.01 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585286.dsl.bell.ca) |
| 23:01.56 | Shaun2222 | oh how lovely... |
| 23:01.56 | Shaun2222 | WaitExten does not work in a Macro! |
| 23:02.17 | Greek-B0y | what are the permissions supposed to be on /var/log/asterisk/* |
| 23:02.31 | Greek-B0y | i'm running asterisk as user:group asterisk:asterisk |
| 23:02.56 | Shaun2222 | they are root:root 644 |
| 23:03.09 | Shaun2222 | eww... |
| 23:04.11 | *** join/#asterisk duckz (n=duckz@81-31-157-49.vm.dnshosting.it) |
| 23:04.27 | Greek-B0y | why ewww? coz i'm running it as user asterisk? |
| 23:04.48 | Shaun2222 | because i just noticed the perms are 644 |
| 23:05.14 | Qwell | it's not a bug. you should use Read(). There was just a post on the asterisk-users list about this |
| 23:05.18 | Qwell | Shaun2222: ^ |
| 23:06.31 | Shaun2222 | bahh, read sucks... :) |
| 23:07.18 | BBHoss | anybody need a TE210P card cheap? |
| 23:08.14 | Shaun2222 | Qwell: with read() nobody can enter anything until it's done playing right? |
| 23:09.16 | Greek-B0y | Logger Warning: Unable to open log file '/var/log/asterisk/messages': Permission denied |
| 23:09.22 | *** join/#asterisk Winkie (n=urmom@87-194-109-4.bethere.co.uk) |
| 23:09.38 | Greek-B0y | whether i set the owner to root or asterisk, same thing |
| 23:10.26 | Shaun2222 | check the permissions down the line... |
| 23:10.26 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:10.45 | Nugget | yeah, make sure you've got +x as appropriate on /var and /var/log and /var/log/asterisk |
| 23:12.21 | Greek-B0y | i ran chmod -R +x /var /var/log /var/log/asterisk |
| 23:12.35 | Shaun2222 | jeez... |
| 23:12.52 | Greek-B0y | whoops |
| 23:12.53 | Greek-B0y | lol |
| 23:13.00 | *** join/#asterisk defsmac (n=andy@defsdoor.gotadsl.co.uk) |
| 23:13.15 | defsmac | recommended card for 6 FXO ? |
| 23:13.22 | Shaun2222 | Greek-B0y: i think you should stay away from the recursive switches.... |
| 23:13.30 | Greek-B0y | reversed it with chmod -R -x /var /var/log /var/log/asterisk |
| 23:13.38 | Shaun2222 | no you didnt |
| 23:13.42 | Qwell | defsmac: Digium TDM800P or TDM2400P (if you plan to expand) |
| 23:13.42 | Shaun2222 | you probably just made it worse |
| 23:13.57 | Shaun2222 | in fact i guarantee it |
| 23:14.18 | defsmac | Qwell, what do you get with it for connection to DP/sockets etc.. |
| 23:14.32 | Qwell | DP? I'm not familiar with the acronym.. |
| 23:14.35 | Shaun2222 | Greek-Boy: now you just removed execute permission from every file and dir in /var |
| 23:14.42 | mvanbaak | DialPlan ? |
| 23:14.56 | defsmac | Qwell, in uk it's where BT terminate incoming wires |
| 23:15.06 | *** part/#asterisk aiurea (n=k9@arcadia.timisoara.roedu.net) |
| 23:15.19 | defsmac | distribution point or some other abbreviation |
| 23:15.20 | mvanbaak | Brit Telecom ? |
| 23:15.24 | Qwell | the card has 8 rj11 plugs on it, if that answers your question |
| 23:15.36 | defsmac | cool it does :) |
| 23:15.38 | Qwell | the tdm800p does, anyhow. the tdm2400p uses an amphenol type connector |
| 23:15.41 | Greek-B0y | lol |
| 23:15.44 | Greek-B0y | now i'm stuffed |
| 23:15.59 | Greek-B0y | damn -R is dangerous |
| 23:16.00 | Greek-B0y | lol |
| 23:16.09 | Qwell | defsmac: any it's modular too, so you can get fxo or fxs on there |
| 23:16.32 | mvanbaak | Greek-B0y: -R with rm -f ? |
| 23:16.45 | Qwell | the 800 has 2 quad port modules, and the 2400 has 6 |
| 23:16.49 | defsmac | Qwell, yeah - i see now - I just need 6 lines atm |
| 23:16.51 | Greek-B0y | lol |
| 23:16.54 | Greek-B0y | no thank u |
| 23:17.09 | Shaun2222 | Greek-Boy: i hope this isnt a production machine? |
| 23:18.09 | Greek-B0y | luckily not |
| 23:18.23 | Greek-B0y | but |
| 23:18.30 | Greek-B0y | still want to treat it as one |
| 23:18.47 | Shaun2222 | well if it were me i would just wipe and do a clean install on it.. |
| 23:18.49 | Greek-B0y | now I will have to compare the permissions of a similiar machine |
| 23:18.52 | Greek-B0y | lots of work |
| 23:19.12 | Stefan1979 | "sudo chmod 777 -R /" fixes every possible possible problem ;) host (just don't do it ;) |
| 23:19.20 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.17 (2008/01/02), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
| 23:20.16 | defsmac | Qwell, so I need TDM808B - 2 quad FXO |
| 23:20.21 | mvanbaak | Greek-B0y: you dont backup your acl's ? |
| 23:20.36 | *** join/#asterisk anthm (n=anthm@mb50736d0.tmodns.net) |
| 23:20.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:20.50 | _ShrikE | how would I determine what version of asterisk-addons is running on a box? |
| 23:20.56 | defsmac | damned expensive - and I still need echo cancel on top :o |
| 23:21.01 | Greek-B0y | mvanbaak not on non-productions boxes |
| 23:21.35 | Stefan1979 | defsmac: I used a OpenVox clone of the tdm card.. works fine (for me) |
| 23:21.58 | mvanbaak | Greek-B0y: try this: /usr/bin/getfacl -R / | /bin/bzip2 -9 -c > $DEST/system/acl-$TIMESTAMP.txt.bz2 |
| 23:22.25 | mvanbaak | that's taken from my backup script |
| 23:22.33 | JT | defsmac: why do you have so many FXO ports? |
| 23:23.07 | defsmac | JT, existing system - they have 6 analog lines (8 actually but I'm not touching 2 of them) |
| 23:23.26 | defsmac | I love them to go to PRI but no chance |
| 23:23.44 | defsmac | Stefan1979, I dont see a 6-8 port one |
| 23:23.47 | Qwell | defsmac: I don't know about the exactly bundle model number, but yes, 2 quad FXO |
| 23:24.05 | Stefan1979 | defsmac: OpenVox A800P-01 8 FXO |
| 23:24.12 | mvanbaak | get a sangoma card |
| 23:24.23 | mvanbaak | they have a stackable FXO card |
| 23:24.38 | defsmac | mvanbaak, I might - I've used sangoma PRIs before |
| 23:24.54 | mvanbaak | me too |
| 23:25.06 | Greek-B0y | mvanbaak i dont have getfacl |
| 23:25.13 | mvanbaak | but only because they are easier to get then digium cards here in .nl |
| 23:25.21 | Greek-B0y | i will install acl now |
| 23:25.35 | mvanbaak | Greek-B0y: install it. the same package also has setfacl |
| 23:25.43 | JT | defsmac: no chance, why not? |
| 23:25.46 | mvanbaak | you can feed it the output of getfacl |
| 23:25.56 | mvanbaak | to fix acl/permission in a single run |
| 23:25.57 | mvanbaak | it rox |
| 23:26.45 | fujin | damnit, I've having some random issues where asterisk dies when someone transfers a call |
| 23:26.48 | fujin | and I can't work out why |
| 23:26.52 | Stefan1979 | mvanbaak: http://www.mapleleaf-technologies.com/ in germany probarely ships to nl (they ship for free to denmark) |
| 23:27.09 | mvanbaak | Stefan1979: I'm talking about it with Speakup |
| 23:27.32 | mvanbaak | fujin: recompile with DONT_OPTIMIZE and grab a backtrace |
| 23:27.38 | mvanbaak | that way you can open a ticket |
| 23:27.46 | fujin | its' not dumping a core |
| 23:27.57 | mvanbaak | start asterisk with -g |
| 23:28.40 | Greek-B0y | mvanbaak so in my case I can just read the acl's from a similiar box? |
| 23:28.40 | fujin | where will -g drop cores to? |
| 23:28.40 | mvanbaak | Greek-B0y: prolly |
| 23:28.40 | mvanbaak | Greek-B0y: if the setup is simular install acl there as well |
| 23:28.40 | fujin | and how do I recompile with DONT_OPTIMIZE? |
| 23:28.44 | fujin | make -D DONT_OPTIMIZE |
| 23:29.06 | mvanbaak | Greek-B0y: use getfacl to get the permission settings, transfer that file to the b0rked box and run setfacl there |
| 23:29.15 | mvanbaak | fujin: make menuselect |
| 23:29.19 | Shaun2222 | Greek-B0y: you could do somthing ghetto like this..... |
| 23:29.24 | mvanbaak | go to compiler options |
| 23:29.32 | Shaun2222 | Greek-B0y: go to a working machine and run.... for i in `find /var`;do echo "chmod `stat $i |grep -i uid|awk '{print $2}'|cut -d'/' -f1|sed 's/(//g'` $i" >> fixvar.sh;done |
| 23:29.41 | Shaun2222 | copy over the fixvar.sh file to the borked machine |
| 23:29.45 | Shaun2222 | and run it |
| 23:29.57 | Stefan1979 | Shaun2222: wauv bash expert.. |
| 23:29.58 | defsmac | JT - 1200 for E1 install |
| 23:30.17 | defsmac | and no perceived benefit to the end user |
| 23:30.26 | Shaun2222 | Stefan1979: not quiet, go into the bash channel with that command and they would probably laugh at me :) |
| 23:30.40 | Greek-B0y | wow Shaun |
| 23:30.44 | *** join/#asterisk husimon (n=nhuisman@dhcp52.IfA.Hawaii.Edu) |
| 23:30.51 | Greek-B0y | thanks man |
| 23:30.53 | defsmac | Shaun2222, that;s hideous |
| 23:31.00 | Shaun2222 | defsmac: lol |
| 23:31.02 | fujin | lol aye |
| 23:31.05 | Shaun2222 | defsmac: hey it works |
| 23:31.06 | JT | defsmac: wow that sucks, i can get pri installs for free here |
| 23:31.07 | fujin | could at least use $() instead of `` |
| 23:31.16 | defsmac | JT, :o |
| 23:31.18 | husimon | hey does anyone know if the sip 4.4 images for cisco phones are ok or are they buggy? I need to use 4.4 because I want to make sure I can roll back to to sccp in case of a problem. |
| 23:31.21 | Shaun2222 | fujin: thats just how i get down. |
| 23:31.25 | husimon | once I know things are ok i'll upgrade to 8.x |
| 23:31.30 | JT | defsmac: on a 24 month contract anyway |
| 23:31.54 | husimon | because I know once you goto 5.0 sip or sccp images they won't let you downgrade |
| 23:32.25 | Shaun2222 | Greek-B0y: that command was created on a centos 5 machine... so look at the fixvar.sh file before you run it to see if it looks ok. |
| 23:33.11 | Shaun2222 | should work fine, but i havnt been on a debian machine is a while. |
| 23:33.15 | Stefan1979 | husimon: I'm tired of my cisco 7940s - if you brick your i have 3 for sale cheap (and they have never been used for other than testing a couple of days) |
| 23:33.38 | Stefan1979 | Shaun2222: It seemed to work on a ubuntu desktop |
| 23:33.42 | husimon | yeah i'm scared of bricking them |
| 23:33.48 | mvanbaak | husimon: chan_skinny in trunk is making progress |
| 23:33.58 | husimon | yeah that's the thing |
| 23:33.59 | mvanbaak | husimon: I run it in production and it's great |
| 23:34.05 | husimon | is it lacking features ? |
| 23:34.11 | husimon | i read that the chan_sccp was alot better |
| 23:34.24 | mvanbaak | husimon: chan_sccp is dead |
| 23:34.27 | Nugget | That's what Qwell says. :) |
| 23:34.39 | mvanbaak | chan_skinny is actually being maintained |
| 23:34.46 | husimon | that's what I get for reading wikis |
| 23:34.48 | mvanbaak | and,... it's in the default asterisk |
| 23:34.57 | fujin | is there anyway I can cause asterisk to dump a core while running? |
| 23:34.59 | husimon | k well i'll try to use it |
| 23:35.01 | fujin | without recompiling |
| 23:35.04 | mvanbaak | and there are actually some ppl working on it |
| 23:35.29 | mvanbaak | Nugget: there has been a lot of activity on chan_skinny lately |
| 23:35.46 | drmessano | http://consumerist.com/344547/microsoft-customer-service-calls-back-10-years-later |
| 23:35.49 | drmessano | HAW! |
| 23:35.57 | *** join/#asterisk RoyK (n=roy@91.149.27.45) |
| 23:35.59 | mvanbaak | Nugget: I can know. I wrote some of the stuff, and I tested loads of stuff done by DEA and Qwell |
| 23:36.32 | defsmac | anyone used aastra 9133i and 9112i know how they differ? |
| 23:36.36 | Shaun2222 | is there a cmd that works like background that lets the user enter somthing before the play finishes? |
| 23:36.47 | defsmac | (as far as setup is concerned) |
| 23:36.54 | [TK]D-Fender | defsmac : difference is obvious, and neither suggested |
| 23:37.01 | drmessano | lol |
| 23:37.18 | husimon | well I have to use sccp version 3 or some crap |
| 23:37.22 | husimon | hopefully that's ok |
| 23:37.36 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:37.36 | husimon | just until i'm sure the system is all up at working then I'll slowly upgrade phones to sip 8.x |
| 23:37.48 | drmessano | lol |
| 23:38.01 | defsmac | [TK]D-Fender, I thinking of suggest a customer goes for the 9112i but I've not used them before - only the 9133i and 480i - just want to make sure I don't trip over |
| 23:38.02 | drmessano | Youre gonna fight to get sccp working to upgrade to sip later? |
| 23:38.05 | husimon | do you know if chan_skinny has support for like 3 way calling, call forwarding, etc etc? |
| 23:38.12 | mvanbaak | husimon: if you really want sip why bother with chan_skinny |
| 23:38.18 | husimon | drmessano: the reason why is because we currently have a call manager system in place |
| 23:38.31 | husimon | I can't upgrade the sccp for the call manager because it is actually a backup which is limping along in a read only state. |
| 23:38.32 | mvanbaak | husimon: callforwarding has just been committed to trunk |
| 23:38.50 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 23:38.56 | plik | greetings |
| 23:38.57 | husimon | so if I go ahead and upgrade all the phones to sip, our system is down, and if I can't get asterisk working properly or there is an issue, i can't back out |
| 23:38.58 | drmessano | Just get Asterisk working and cut over fully |
| 23:39.00 | husimon | then we are totally dead. |
| 23:39.22 | [TK]D-Fender | defsmac, Linksys is a better choice in your market than those 2. |
| 23:39.30 | husimon | just scared of losing the whole phone system because i do something n00b wrong in asterisk. |
| 23:39.35 | mvanbaak | drmessano: sccp works with * as well |
| 23:39.39 | *** join/#asterisk Victor_Yure (n=Victor_Y@201.9.1.95) |
| 23:39.45 | drmessano | mvanbaak: I know |
| 23:39.54 | [TK]D-Fender | defsmac, 480i would be better than Linksys, but I'm sure it comes at a premium comparatively |
| 23:39.56 | drmessano | My point was, get Asterisk working and use SIP |
| 23:40.04 | mvanbaak | drmessano: I have a 7905 and a 7960 here at home running sccp with chan_skinny |
| 23:40.07 | drmessano | Emphasis on, "getting it working" |
| 23:40.10 | defsmac | [TK]D-Fender, yeah - you said the other day - but I'm trying not to venture too far out of my comfort zone |
| 23:40.13 | husimon | drmessano: yeah |
| 23:40.19 | drmessano | His only reason for not using SIP is needing the backup |
| 23:40.20 | mvanbaak | drmessano: no, dont use sip with the cisco phones |
| 23:40.31 | mvanbaak | why use sip if you can use the native format ? |
| 23:40.36 | mvanbaak | ugh |
| 23:40.47 | husimon | drmessano: because the support for it in chan_skinny isn't as good as the sip features |
| 23:40.48 | [TK]D-Fender | defsmac, sure tie our hands.... my advise stands |
| 23:40.53 | jblack | My ex is so lazy, that she called me to transfer her to our kid, because she didn't feel like hitting any buttons. |
| 23:40.55 | husimon | err that was meant for mvanbaak |
| 23:41.31 | plik | jblack: she just wanted to check up on you on the sly ;) |
| 23:41.43 | jblack | she was here just 3 hours ago! |
| 23:41.50 | mvanbaak | husimon: did you even try it ? |
| 23:41.54 | defsmac | [TK]D-Fender, what specific linksys models would you recommend ? |
| 23:42.01 | plik | see,definitely then |
| 23:42.03 | drmessano | You should have boned her, that would have bought you a week |
| 23:42.14 | jblack | ugh. |
| 23:42.23 | jblack | Her running off was the best thing that ever happened to me. |
| 23:42.30 | drmessano | Amen, bro |
| 23:42.35 | husimon | mvanbaak: i'm just about to. I'm pretty sure SIP is pretty native to the cisco phones these days seeing as they release image for them and call manager uses sip. |
| 23:42.40 | [TK]D-Fender | defsmac, the all configure the same. Depends on your needs.... I'd prefer a 941/942 personally over the lower ones as you get lit line-keys |
| 23:43.10 | mvanbaak | husimon: skinny is the native format. it's still the most stable and fast on the cisco phones |
| 23:43.30 | mvanbaak | ugh, I cant stand ppl judging something without even bothering to try |
| 23:43.31 | husimon | mvanbaak: well I'll give it a shot, what ever works is fine with me. |
| 23:43.51 | JT | can't stand people saying ppl |
| 23:43.59 | drmessano | I can't stand people |
| 23:44.00 | mvanbaak | sorry JT |
| 23:44.08 | mvanbaak | drmessano: I totally agree ! |
| 23:44.09 | JT | anyway, chan_skinny or sccp seems like a pretty limited route |
| 23:44.22 | JT | only if you need backwards compatibility for sccp ccm |
| 23:44.25 | mvanbaak | JT: ok, tell me what you mess |
| 23:44.26 | drmessano | Ow, damnit |
| 23:44.30 | drmessano | I WANT MY WEEKEND BACK |
| 23:44.30 | jblack | ppl r ppl, so y shuld it be. U and I should tolerate, people typing aufully. |
| 23:44.33 | Qwell | JT: or phones that work well |
| 23:44.36 | mvanbaak | s/mess/miss |
| 23:44.42 | JT | Qwell: if you think so |
| 23:44.53 | JT | it's a pretty asterisk only solution |
| 23:44.57 | defsmac | [TK]D-Fender, can I daisy chain a PC off a 942? |
| 23:45.19 | [TK]D-Fender | defsmac, either. |
| 23:45.27 | mvanbaak | common people. have you even tried chan_skinny.so in trunk ??? |
| 23:45.34 | [TK]D-Fender | defsmac, 942 has PoE and backlit screen |
| 23:45.49 | JT | mvanbaak: you're ignoring the point, it's not an open standard, sccp |
| 23:45.55 | JT | it's a limiting solution |
| 23:46.00 | JT | so you can use it with ccm |
| 23:46.04 | JT | or asterisk |
| 23:46.05 | JT | yay |
| 23:46.07 | husimon | mvanbaak: one other reason i was a little hesitant was i have asterisk business edition, it doesn't have the latest up to date chan_skinny. |
| 23:46.09 | defsmac | yeah - I need POE - the site is getting a real patch panel so a new switch is required - might as well be POE |
| 23:46.09 | [TK]D-Fender | mvanbaak, You are the only one with the psycho deals on Cisco, the rest of us prefer to AVOID trouble rather than trying to "make the best of it" |
| 23:46.10 | husimon | they are using asterisk 1.2 |
| 23:46.21 | mvanbaak | JT: erm, you _ARE_ in #asterisk here |
| 23:46.23 | JT | husimon: my condolences ;) |
| 23:46.26 | JT | mvanbaak: and? |
| 23:46.32 | defsmac | (at the moment they have a bunch of cables under a desk) |
| 23:46.34 | husimon | JT: they will be upgrading to 1.4 in a few moths |
| 23:46.35 | drmessano | [TK]D-Fender: Do you have the SCCP firmware for my GXP-2000? |
| 23:46.35 | husimon | months |
| 23:46.43 | JT | husimon: ABE, cough |
| 23:46.55 | mvanbaak | 00:46 < JT> so you can use it with ccm |
| 23:46.55 | mvanbaak | 00:46 < JT> or asterisk |
| 23:46.56 | [TK]D-Fender | drmessano, ZING! |
| 23:46.59 | mvanbaak | yeah |
| 23:47.06 | husimon | JT: yeah I keep forgetting to use the acronym :) |
| 23:47.13 | mvanbaak | you can use it with asterisk, the main topic of this channel |
| 23:47.15 | JT | mvanbaak: exactly, sccp is NOT an open standard that's widely supported |
| 23:47.20 | JT | mvanbaak: so? |
| 23:47.40 | mvanbaak | JT: so? you ever worked with nvidia, ati or intel hardware ? |
| 23:47.45 | JT | many people have much greater needs than asterisk being the be all and end all |
| 23:47.51 | mvanbaak | that's non-openstandard even |
| 23:47.59 | *** join/#asterisk tsabi (n=tsabi@pool-5972.adsl.interware.hu) |
| 23:48.04 | JT | some people have sip proxies |
| 23:48.08 | JT | multiple b2buas |
| 23:48.12 | mvanbaak | JT: this is #asterisk, it's about asterisk |
| 23:48.17 | JT | they need standards |
| 23:48.23 | JT | mvanbaak: no, it's about opening your mind |
| 23:48.29 | JT | just because you use asterisk |
| 23:48.34 | JT | doesn't mean it's the only thing you use |
| 23:48.36 | mvanbaak | if you are talking about ser or $random_other_system go to #ser or #random_other_system |
| 23:49.00 | mvanbaak | JT: they asked about their cisco phone _AND_ asterisk |
| 23:49.01 | drmessano | I think the point is that if my phone does OU812 or SIP, why bother with the OU812 module for Asterisk.. Forget the phone... surely the chan_sip is bit more stable in Asterisk than chan_skinny? I assume people use chan_sip? ;) |
| 23:49.09 | JT | mvanbaak: waaa |
| 23:49.22 | husimon | sorry for starting a war :P |
| 23:49.30 | JT | mvanbaak is just having a cry |
| 23:49.33 | JT | nothing to worry about |
| 23:49.46 | mvanbaak | yeah whatever |
| 23:49.51 | fujin | uh, anyone else noticing that svn.digium.com is broken? |
| 23:49.56 | drmessano | Im still waiting for that SCCP firmware for my Grandstream |
| 23:50.08 | JT | and mvanbaak was implying that everyone who does not use sccp with asterisk is an idiot |
| 23:50.11 | JT | which is wrong |
| 23:50.14 | mvanbaak | fujin: svn. and bugs. are down |
| 23:50.18 | fujin | agh. |
| 23:50.24 | fujin | what a pita. |
| 23:50.25 | mvanbaak | JT: that's what you make of it |
| 23:50.30 | fujin | mvanbaak: any ETA? |
| 23:50.33 | mvanbaak | I was talking about the cisco phones |
| 23:50.37 | mvanbaak | fujin: nope |
| 23:50.53 | JT | even with cisco phones, there are plenty of valid reasons to not use sccp |
| 23:51.00 | JT | of course it's better to avoid cisco phones |
| 23:51.01 | JT | but still |
| 23:51.10 | mvanbaak | JT: tell us and we can fix it |
| 23:51.16 | husimon | yeah too bad we have 60 of them already :P |
| 23:51.35 | JT | husimon: a lot of businesses make that mistake ;) |
| 23:51.38 | drmessano | I love Cisco phones.. So much that I got to McDonalds and give them an extra $8 for every hamburger.. just because. |
| 23:51.47 | tsabi | hi, i just bought a Granstresm 101 phone. dont you know how reliable is this pohone? |
| 23:51.47 | husimon | JT: they were purchased with ccm before my time. |
| 23:51.47 | drmessano | s/got/go |
| 23:51.56 | JT | tsabi: not very |
| 23:51.58 | JT | ~gs |
| 23:51.59 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 23:52.09 | drmessano | ~cisco |
| 23:52.09 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 23:52.16 | drmessano | ftw |
| 23:52.23 | fujin | Fuck a cisco phone |
| 23:52.24 | fujin | imho. |
| 23:52.29 | tsabi | ahh, and linksys 421 one? is that better choice? |
| 23:52.29 | drmessano | imho.. lol |
| 23:52.39 | fujin | You can't go past a Polycom, and even if you do; you'd pick an aastra or a Linksys |
| 23:52.45 | defsmac | any such thing as a sip tannoy ? |
| 23:52.47 | husimon | imo you should never use imho |
| 23:52.48 | fujin | tsabi: the Linksys 9xx series are nice. |
| 23:52.51 | husimon | ;) |
| 23:52.51 | JT | fujin: agreed |
| 23:52.59 | tsabi | thanks :) |
| 23:53.05 | JT | not a fan of the char matrix on the older aatras |
| 23:53.11 | JT | but still |
| 23:53.15 | fujin | we went with 942's and 962's for receptionist/highvolume people |
| 23:53.32 | fujin | although the 962+932 sidecart hasn't been working (rather, I haven't bothered to try get it working yet) |
| 23:53.33 | JT | some people favour snoms over linksys phones |
| 23:53.35 | JT | they're crazy |
| 23:53.45 | mvanbaak | JT: what are the things with chan_skinny that dont work for you ? |
| 23:53.45 | fujin | ha. |
| 23:53.54 | tsabi | ohh, the 962, have you tryed the additional button, the 935 or what module? |
| 23:54.01 | JT | linksys units actually render nicely and look professional |
| 23:54.05 | fujin | tsabi: I have 962=(32's |
| 23:54.05 | JT | and are easier to use |
| 23:54.05 | tsabi | ohh |
| 23:54.08 | defsmac | bed& |
| 23:54.09 | tsabi | i juist see the answer :) |
| 23:54.09 | fujin | 932 |
| 23:54.25 | JT | mvanbaak: the fact that using sccp would limit my options in future does not make it viable |
| 23:54.43 | Qwell | JT: you can still flash the phones to SIP... |
| 23:54.50 | mvanbaak | JT: and how does that relate to asterisk ? |
| 23:55.10 | JT | mvanbaak: because asterisk is a platform with VoIP interconnectivity, duh |
| 23:55.51 | JT | Qwell: yes but if you have a sip proxy, and a few different b2buas, you can't use sccp |
| 23:56.14 | JT | most big deployments have either a sip proxy, a h.323 gatekeeper, or zillions of dollars worth of cisco ccm gear |
| 23:56.27 | lzhang | JT: be a pal and help me figure out how to get this T1 working :) |
| 23:56.28 | mvanbaak | or asterisk boxen |
| 23:56.52 | JT | mvanbaak: and how do you proxy those boxes? |
| 23:57.03 | JT | lzhang: what's wrong with it? |
| 23:57.27 | mvanbaak | JT: asterisk is my border, no need for a proxy |
| 23:57.32 | lzhang | I just hooked up the pri this afternoon, I've got a sangoma card and the drivers installed... ztfcfg shows 24 channels as expected |
| 23:57.48 | lzhang | for some reason I see no indication of the channels in asterisk cli... |
| 23:57.59 | husimon | you setup the zapata.conf? |
| 23:58.00 | lzhang | I'm not sure which commands I should be using to find out |
| 23:58.03 | JT | mvanbaak: how do you do failover and load balancing? |
| 23:58.18 | husimon | yeah how do you folks handle failover in asterisk? |
| 23:58.22 | husimon | i was planning on using heartbeat |
| 23:58.28 | husimon | with rsync script |
| 23:58.32 | mvanbaak | JT: a cluster of openbsd carp+pfsync+relayd boxen in front of it |
| 23:58.40 | lzhang | ya I got zapata.conf with switchtype, context, group, signalling, channel, etc |
| 23:58.53 | husimon | i think you can do like show pri channel <> |
| 23:58.55 | husimon | 1 2 3 4 |
| 23:58.57 | husimon | or something |
| 23:59.05 | husimon | to see the channels |
| 23:59.20 | lzhang | ya I've been trying that, for some reason in the cli it says no command pri show? |
| 23:59.31 | husimon | what about show pri? |
| 23:59.31 | JT | mvanbaak: that doesn't sound nearly as nice as proxying at the sip level |
| 23:59.51 | JT | mvanbaak: and is only good in a hetrogenous environment |
| 23:59.54 | mvanbaak | JT: we have sccp, sip and h323 endpoints |
