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00:12.48 | torrancew | thanks all the same, jdoe |
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01:30.42 | *** part/#asterisk NormT_ (~NormT@bas4-oshawa95-1242434647.dsl.bell.ca) |
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01:41.37 | ruben23 | hi anyoe used phone booth |
01:42.20 | Kyosh | yea back in the 80s |
01:44.11 | ruben23 | <PROTECTED> |
01:45.33 | *** join/#asterisk coppice (~chatzilla@153.166.232.220.dyn.pacific.net.hk) |
01:51.14 | carrar | FREE |
01:51.29 | carrar | Be that comes with some excellent SLA's |
01:51.31 | carrar | Bet |
01:59.21 | *** join/#asterisk lepine (~leprechau@69-196-166-209.dsl.teksavvy.com) |
02:00.09 | lepine | I'll be going to the docs shortly, but can someone confirm there's something I'm not grasping yet? http://pastie.org/951037 |
02:01.02 | lepine | I get as far as the new context, but am told the extension is invalid |
02:01.30 | lepine | (and no handler) |
02:01.54 | carrar | remove those spaces |
02:02.16 | carrar | are you sure they are sending you a +1 |
02:03.47 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
02:03.51 | carrar | don't need quote marks in the Noop also |
02:04.22 | lepine | Thanks, that worked (spaces) |
02:04.46 | lepine | however, it irks me, considering i program most of the time and have spaces between parens |
02:04.57 | lepine | and yes, send me the 1 |
02:05.18 | carrar | Brace yourselffor more IRKS |
02:05.23 | lepine | taking note for NoOp, but I think it makes sense to keep it |
02:05.24 | lepine | lol |
02:05.29 | lepine | anything in mind? |
02:05.52 | carrar | I'm sure you'll find more |
02:06.28 | lepine | eh, I've only been fiddling w/ asterisk about 30 hours (very sporadically) ... |
02:06.32 | lepine | I'll get back to you :) |
02:06.42 | carrar | it's fun |
02:07.15 | lepine | yeah, my boss was like "choose, asterisk or ironport (bulk mailing)" ... asterisk, please. |
02:07.51 | lepine | like I care about learning to spam people, i'd rahter play around w/ voip |
02:08.28 | carrar | What version are you running |
02:08.42 | lepine | 1.4 |
02:09.03 | lepine | running debian, and that's what they've got. don't love bypassing apt |
02:09.31 | carrar | Should also have a underscore before 1888 |
02:09.36 | carrar | for pattern matching |
02:09.53 | lepine | tried AsteriskNow, trixbox and asterisk/deb + freepbx 3 ... |
02:10.13 | lepine | _ is asterisks */% ? |
02:10.24 | lepine | (wildcard) |
02:10.45 | lepine | nm, I better read about it |
02:11.16 | carrar | search underscore pattern matching |
02:11.18 | carrar | bbl |
02:11.25 | lepine | will do, thanks |
02:17.54 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
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02:43.17 | *** mode/#asterisk [+o twisted] by ChanServ |
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03:31.52 | dandate2 | what is thee cheapest way to make outbound calls to the UK? |
03:32.33 | coppice | steal |
03:32.50 | jql | works for me |
03:32.55 | TJNII | Yell out the window. |
03:33.10 | jql | it's a company phone, so it's merely misappropriation of resources |
03:33.19 | TJNII | "Hey I can call my maw while I'm up here. HEY MAW! GET OFF THE DANGED ROOF!" |
03:33.24 | TJNII | ~itsp-uk |
03:33.29 | TJNII | ~itsplist-uk |
03:33.30 | infobot | rumour has it, itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
03:46.47 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
03:50.18 | carrar | Give whoever you want to call in the UK a SIP Phone! |
03:51.20 | *** join/#asterisk sourcode (~code@ppp-58-11-73-234.revip2.asianet.co.th) |
03:54.50 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
04:03.52 | coppice | you mean a UKIP phone? |
04:03.56 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
04:04.46 | D-Boy | Hi, how to get a list of connected sip phones (hard phones and soft phones) ? |
04:06.56 | dandate2 | sip show peers |
04:08.25 | D-Boy | dandate2 : thx ! |
04:10.38 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
04:13.05 | carrar | ~book |
04:13.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
04:13.17 | carrar | download that D-Boy |
04:16.56 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
04:16.59 | D-Boy | Oh nice ! i guess now i need just a thousand cups of coffee and patience to get the spirit of asterisk :p |
04:17.07 | D-Boy | carrar : thank you :) |
04:17.46 | carrar | worth it |
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05:54.30 | *** join/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz) |
06:02.49 | Vin73 | hi, i am relatively new to asterisk and would request for some help on an error |
06:03.18 | Vin73 | hi |
06:03.49 | Vin73 | is anyone there? |
06:04.02 | Vin73 | quit |
06:04.05 | *** part/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz) |
06:04.08 | *** join/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz) |
06:04.15 | Vin73 | hi there |
06:04.30 | Vin73 | I am a new user seeking some help |
06:06.52 | Vin73 | #asterisk- [freenode-info] channel |
06:07.05 | *** part/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz) |
06:07.13 | carrar | read this |
06:07.16 | carrar | ~book |
06:07.16 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
06:07.36 | *** join/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz) |
06:07.39 | carrar | ~book |
06:07.39 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
06:07.42 | carrar | read that |
06:08.09 | Vin73 | hi I have read that book, but this is a specific error |
06:08.18 | Vin73 | In fact I have that book |
06:08.54 | Vin73 | The error is X-Asterisk-HangupCauseCode: 20 |
06:09.20 | Vin73 | has anyone seen this error before? |
06:10.38 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
06:11.10 | Vin73 | hello? |
06:11.37 | carrar | https://issues.asterisk.org/view.php?id=14684 |
06:11.42 | *** join/#asterisk voxter (~voxter@c-66-176-249-104.hsd1.fl.comcast.net) |
06:12.57 | Vin73 | ok thanks |
06:13.02 | Vin73 | I will have a look |
06:13.10 | Vin73 | bye |
06:13.53 | sawgood | If I wanted my Asterisk box to 'drop' all current registrations with my ITSP carriers, and try to re-authenticate ... wouldn't reload now from the CLI do that? |
06:16.11 | *** join/#asterisk farkus_ (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
06:16.19 | carrar | restart would |
06:18.13 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-zkccxelspdimjzji) |
06:20.15 | *** join/#asterisk biscuit314 (~michael@s72-38-112-54.static.comm.cgocable.net) |
06:20.16 | sawgood | restart now |
06:24.29 | *** join/#asterisk biscuit314 (~michael@s72-38-112-54.static.comm.cgocable.net) |
06:24.49 | biscuit314 | Hello! I'm trying to get into AGI using pyst. Got a PERL script working fine, but my attempts to get a python script end in "Exec format error" Can anyone help? |
06:25.30 | p3nguin | sawgood: "reload" should never be used. |
06:27.17 | carrar | biscuit314, sounds like you have a python issue |
06:27.50 | carrar | debug your script |
06:29.13 | biscuit314 | I lifted the following from a sample described in the pyst project: |
06:29.18 | biscuit314 | from asterisk.agi import * |
06:29.19 | biscuit314 | def main(): |
06:29.19 | biscuit314 | <PROTECTED> |
06:29.19 | biscuit314 | <PROTECTED> |
06:29.48 | biscuit314 | running directly with python interpreter (for whatever that's worth) reveals no issues |
06:31.50 | carrar | How about you write a simple script to tell you what variables asterisk is sending your script |
06:32.01 | carrar | start with that |
06:32.16 | carrar | then try setting some variables |
06:32.43 | carrar | or join #python and ask there |
06:32.59 | biscuit314 | Perhaps the problem is in the pyst libraries. I'll take your advice and start simple, build up using stdout/stderr, take it from there. |
06:33.18 | biscuit314 | But not tonight.... time for bed. Nice bumping into you! g'nite! |
06:40.26 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
06:49.40 | sawgood | I have two ITSP carriers ... when I do a sip show peers from the CLI one ITSP shows up with two listings and the other ITSP shows up with only one listing |
06:50.08 | sawgood | for the ITSP with two listings ... one is for outgoing settings and the other is for incoming settings |
06:50.31 | sawgood | I was wondering 'why' the 2nd ITSP only shows up once ... with sip show peers, but the other ITSP shows up twice |
06:50.38 | sawgood | strange to try and explain ... |
06:51.40 | sawgood | never mind I see why now |
07:01.16 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
07:03.56 | ChannelZ | Exactly. |
07:10.00 | *** join/#asterisk VanDyke (~scotch@kernel-panic/vandyke) |
07:10.18 | VanDyke | good evening |
07:14.58 | ChannelZ | right |
07:15.00 | VanDyke | has anyone here set up before an asterisk box with a SIP trunk to CommPartners ? |
07:17.39 | carrar | Sounds impossible! |
07:18.59 | VanDyke | carrar, almost |
07:19.00 | tzafrir_laptop | looks like biscuit314 was missing the #! line on the top of the script |
07:27.40 | ChannelZ | she bangs! she bangs! |
07:30.17 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
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07:42.24 | *** join/#asterisk lhz (~shrekz@c-b9aa72d5.021-158-73746f34.cust.bredbandsbolaget.se) |
07:46.12 | sawgood | I have an ITSP carrier who sends all incoming DID traffic to me on a dedicated IP address (which is my Asterisk box) ... |
07:46.26 | sawgood | Can anyone help me with what my 'incoming settings' for this trunk should be like? |
07:46.38 | sawgood | their tech support is closed after 6 PM and on the weekends |
07:46.47 | sawgood | I really don't feel like waiting until Monday to fix this |
07:46.57 | sawgood | I can make calls just fine, but I cannot receive calls |
07:47.28 | sawgood | The 'only' real instructions I got from them is to 'allow' all traffic from IP xxx.xxx.xxx.xxx |
07:47.45 | *** join/#asterisk Tim_Toady (~moi@188.4.66.69.dsl.dyn.forthnet.gr) |
08:25.27 | *** join/#asterisk halkun (~halkun@officegateway.ethostream.com) |
08:26.37 | halkun | I'm done planning out my new home network. Does this look like a good setup with an SPA 3102? ----> http://img85.imageshack.us/img85/5852/neta.png |
08:38.19 | ChannelZ | not sure about the phone splitter, you could probably put the FXO into the DSL modem which should automatically filter that port |
08:44.29 | carrar | Why is the internet plugged into a phone jack? |
08:44.35 | carrar | makes no sense |
08:44.52 | ChannelZ | heh yeah arguably that line should come from the DSL modem |
08:46.20 | carrar | Where is your firewall? |
08:46.38 | carrar | Also why do you have two devices handing out DHCP? |
08:47.25 | ChannelZ | What, you don't wear two condoms? |
08:47.33 | carrar | Turn off DHCP on your AP and let your DSL modem hand out leases |
08:47.44 | carrar | used to |
08:47.45 | carrar | too |
08:50.32 | carrar | Should add that to www.ratemynetworkdiagram.com |
08:51.47 | carrar | http://www.ratemynetworkdiagram.com/?i=10993 |
08:51.51 | carrar | see they have a firewall there |
08:51.54 | carrar | you are missing tghat |
08:52.45 | ChannelZ | he really has 2.. |
08:53.12 | ChannelZ | possibly |
08:53.40 | carrar | halkun, I would make your asterisk box your firewall so that it has a external outside IP address and put your DSL modem in bridge mode |
08:54.06 | carrar | then the second NIC on your Asterisk box would be your internal network |
08:54.24 | carrar | run a real DHCP server (ISC) on yoru asterisk box |
08:54.48 | halkun | Oh, I didn't see the replies. :) |
08:54.57 | carrar | keep up |
08:55.05 | halkun | I was busy working :P |
08:55.11 | carrar | So you don't FAIL IRC |
08:55.19 | carrar | no one works on weekends |
08:55.38 | halkun | 3rd shift W-Sun |
08:55.43 | halkun | it's a suck schedule |
08:55.48 | carrar | W-Sun? |
08:55.53 | halkun | wed-sun |
08:55.53 | carrar | Sun MicroSystems? |
08:55.56 | carrar | oh |
08:56.31 | *** join/#asterisk Wildy (~simba@mas4-gw.pleer.ru) |
08:57.23 | halkun | Ok, only the D-Link gives out DHCP |
08:57.39 | carrar | well you ahve it specified on the drawing that the AP is also |
08:57.49 | halkun | the AP will not accept a static setting. |
08:58.10 | halkun | I think that's my ineptitude though :P |
08:58.11 | carrar | AP can get it's IP's from DHCP |
08:58.14 | carrar | then |
08:58.18 | carrar | but thats crazy |
08:58.38 | carrar | Never seen a AP that doesn't have a static IP |
08:58.49 | carrar | I think it's user error |
08:59.01 | halkun | The AP has a static/ 192.168.0.4 |
08:59.33 | carrar | So move your DSL modem to Bridged mode, put your external IP address on the Asterisk box and make it a firewall/NAT server also |
08:59.38 | carrar | and a DHCP server |
08:59.44 | halkun | if you set you wireless up with a static however, the AP won't accept the connection. (It's more for my own note) |
09:00.01 | carrar | AP shouldn't be handing out IP's |
09:00.16 | halkun | I have no control over the DSL modem. That's from my ISP. I have a dynamic address from my ISP |
09:00.26 | carrar | Is the DSL modem in your house? |
09:00.31 | carrar | then you have control over it |
09:00.35 | halkun | yes. This is a home setup |
09:01.01 | carrar | You can do dynamic IP on the asterisk box from your provider too |
09:01.12 | carrar | a litle more nutty however |
09:01.35 | carrar | even more reason not to trust your DSL modem to protect your home network |
09:01.53 | halkun | No, I can't log into the DSL modem, I plug a phone line into it and the other side has PPPoE |
09:02.14 | carrar | What happens when you open a browser to 192.168.0.1? |
09:02.21 | halkun | You get my D-Link |
09:02.50 | carrar | so the Dlink has the external IP address? |
09:03.21 | halkun | yes, it's from my ISP. |
09:03.24 | carrar | again you should make yoru asterisk box your outside firewall/NAT/DHCP server |
09:03.38 | carrar | swap the dlink with your asterisk box |
09:04.12 | carrar | and put a mini switch on the backside of your asterisk box for interal connectivity |
09:04.35 | carrar | But I suspect thats too difficult for ya :) |
09:04.58 | halkun | I kind of see where you are going, but yes |
09:05.51 | halkun | I'll be doing some port forwarding so that the important bits of the * server can be seen by the outside world |
09:06.07 | carrar | see you wouldn't haev to do that my way |
09:06.15 | halkun | If course, I also have to deal with the fact my ISP also blocks all ports under 1024 |
09:06.23 | halkun | (except 22) |
09:06.30 | carrar | whats wrong with that |
09:06.47 | halkun | makes running servers on it a pain |
09:07.08 | carrar | not a aserisk server |
09:07.22 | halkun | I'm still fleshing out my understanding of Asterisk. I know the ports are pretty high there |
09:07.31 | ChannelZ | they is |
09:08.03 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
09:08.08 | halkun | This is my "Lab" set up so I can break things generally screw things up so I can learn |
09:08.25 | carrar | You also have to realize running a "server" with a dynamic IP makes no sense also |
09:08.43 | ChannelZ | dyndns |
09:08.47 | halkun | Well, I'llprobably get dynedns |
09:08.49 | carrar | So this is probably why your ISP offers a buisnes class dns service |
09:08.50 | halkun | yea |
09:08.59 | carrar | dsl |
09:09.06 | ChannelZ | No, that's to extract more money from you |
09:09.17 | halkun | I work from my ISP. I get basic home service for free |
09:09.23 | halkun | 6M DSL |
09:09.26 | carrar | well Freeinternet.com doesn't exist anymore |
09:09.33 | carrar | heh |
09:09.45 | halkun | I'm doing this on the insanely cheap |
09:10.05 | halkun | The idea is to get my droid to work on wifi so I don't have to by cell service |
09:10.26 | *** join/#asterisk HorizonXP (~xitij@69-165-140-215.dsl.teksavvy.com) |
09:10.36 | ChannelZ | well you may or may not have luck with that on free wifi hotspots |
09:10.47 | carrar | on your wireless network or other peoples wireles network? |
09:10.52 | halkun | Yes |
09:11.02 | carrar | Yes A or Yes B |
09:11.06 | halkun | I pretty much have Wifi wherever I go |
09:11.24 | carrar | So you're gonna wish your asterisk box wasn't behidn nat |
09:11.58 | halkun | My friends places, my work, and also I work for a comapny that does Wifi for hotels. I have the keys |
09:12.48 | halkun | I'm donna try and have as much of the * server available. I'll do everything short of moving into the DMZ |
09:12.51 | carrar | anyways, you got my opinion |
09:13.05 | carrar | don't come back if it doesn't work and you didn't take my suggestions :) |
09:13.32 | halkun | Oh, no, I have a second network card for the * server just in case |
09:14.00 | carrar | and use ChannelZ suggestion for dynamic DNS for your droid * server hostname |
09:14.02 | halkun | I'll have the server make the PPPoE connection and hand off to the dlink if worst comes to worst |
09:14.42 | carrar | well you might be able to use the switch portion of the dlink just for the switch part |
09:14.42 | halkun | Origally, I had the * server hooked up to 3102 |
09:14.49 | carrar | but I wouldn't do that |
09:14.55 | carrar | I'd dump it and get a real switch |
09:15.45 | carrar | 5 or 8 port switches are cheap |
09:15.50 | halkun | I found a 5 port in my car. I'll have to find the power plug. |
09:15.58 | carrar | see |
09:16.00 | carrar | free |
09:16.08 | halkun | It's why the laptop is wireless. I ran out of ports on the d-ling |
09:16.36 | carrar | use DHCPD 4.1.1 (http://www.isc.org/software/dhcp) |
09:16.40 | halkun | 4 port + an WAN port |
09:17.12 | carrar | switches don't have WAN ports |
09:17.13 | halkun | never ran a DHCP server before... scary! |
09:17.18 | halkun | no, my D-link |
09:17.30 | carrar | well ISC is the best |
09:17.32 | halkun | I've always had my routers handle it |
09:17.48 | carrar | plenty of example configs on google also when you need help |
09:17.51 | carrar | or jsut ask me |
09:18.23 | carrar | Routers suck for DHCP servers |
09:19.05 | carrar | You have the perfect box for a DHCP server |
09:20.27 | halkun | so, bet case is set DHCP on the * server, give it PPPoE so it can use the DSL modem and ditch the D-link for a nice 5 port switch |
09:20.58 | carrar | yup |
09:21.12 | halkun | that way I also have a direct link to the 3102 from * |
09:21.15 | halkun | It makes sense. |
09:21.38 | halkun | "I'll have to redraw the map again!" |
09:21.48 | carrar | well the 3102 plus into your internal switch |
09:21.52 | carrar | like everythign elswe does |
09:22.13 | carrar | which means everything internal has a "direct link" to it |
09:22.29 | halkun | Right, I'll give that a go then. Thanks for the advice :) |
09:22.30 | carrar | and by direct link I mean on the same logical network |
09:26.45 | carrar | Then once it's setup, get a IPv6 Tunnel form your Asterisk box to a IPv6 tunnel provider! |
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12:21.22 | beefpastry | Can anybody point me in the direction of how I might figure out how to get a polycom to display the callerid instead of the dialed parking extension (is this even possible)? |
12:42.33 | carrar | beefpastry, polycom is making a call |
12:42.44 | carrar | callerid would be the polycom |
12:43.09 | beefpastry | I am aware...was wondering if it could be overridden |
12:43.12 | carrar | there is no magical transfer of the parked call's caller id |
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12:44.37 | carrar | don't even think it's possible since the polycom initiated the call |
12:45.27 | carrar | unless you can send some sort of message to the phone outside of the parked call in some goofie mannor |
12:45.47 | carrar | which there isn't any support that I know of for that |
12:46.34 | carrar | could however write a app you dial |
12:46.45 | carrar | whch then sends that parked call to you as a new call |
12:46.53 | carrar | on your second line |
12:47.10 | carrar | then you would get the caller id of whatever your app sent |
12:47.27 | carrar | having the app extract that caller id out of the park call info |
12:48.08 | carrar | let me know when you have that written :) |
12:48.58 | carrar | or have it send it to you have you hang up that app you dialed |
12:49.06 | carrar | have=after |
12:49.28 | carrar | So I think it's doable |
12:49.35 | carrar | it a different way |
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13:05.47 | beefpastry | crap...ran a yum update to pull 1.6.0.27 and I got 1.6.2 |
13:05.55 | beefpastry | now seg fault |
13:06.04 | beefpastry | good times |
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13:15.05 | smooth_penguin | always hated rpms |
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13:38.46 | beefpastry | phew |
13:44.41 | riddlebox | man I was just reading on scenarios with findme/followme, who wants to sit on the phone waiting while the system rings 5 phones on after another! |
13:44.49 | riddlebox | I would hang up after 4-5 rings |
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15:50.10 | [sr] | howdy |
15:50.26 | [sr] | to install asterisknow in raid on IDE disks, there's a boot command, i forgot :( |
15:50.33 | [sr] | can you remind me? |
15:56.00 | [sr] | ok forget, wasnt needed :) |
16:00.57 | TJNII | #asterisknow |
16:01.00 | TJNII | ~asterisknow |
16:01.01 | infobot | methinks asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
16:04.32 | [sr] | im there |
16:04.33 | [sr] | :P |
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16:38.14 | omes | does anyone know about a howto/guide on how to set up an asterisk server that shares a sip phone number amongst multiple phones/user accounts? |
16:38.33 | omes | im completely new and have no idea where to start with asterisk |
16:38.47 | Kyosh | asteriskdocs.org |
16:38.52 | Kyosh | go read the asterisk book |
16:38.58 | Kyosh | great start |
16:39.27 | [TK]D-Fender | omes: Define "share" |
16:40.35 | omes | [TK]D-Fender: I have one sip account and multiple phones around my house, it would be nice if they could all use that line |
16:41.37 | omes | or i can set up multiple lines, but it would be nice if asterisk could do the routing from the phones to the sip accounts |
16:42.09 | omes | as the information there change quite frequently and i don't want to change the info on my phones that often, it's easier to do it all in one place |
16:42.18 | Kyosh | and they can, but you need to set up the asterisk box |
16:42.18 | Kyosh | or any ip pbx, but asterisk should be easiest |
16:42.18 | Kyosh | asteriskNow would be easy too |
16:42.22 | Kyosh | omes, you want something that is as easy as an out of the box solution, right? |
16:42.35 | omes | no |
16:42.43 | Kyosh | no? |
16:42.51 | omes | i want to know how to edit the config files to do that |
16:42.53 | Kyosh | so you want the challenge of the learning curve? |
16:42.58 | omes | indeed |
16:43.03 | Kyosh | either case you can |
16:43.13 | Kyosh | its just how fast to you want to be up and running |
16:43.24 | omes | doesnt matter if it takes a lot of time |
16:43.36 | omes | i have time, but i want to learn it the "proper" way ;) |
16:43.40 | [TK]D-Fender | omes: this is all basic dialplan. Time to hit the books |
16:44.05 | omes | ah i see, so the book Kyosh gave is good for that then? :) |
16:44.08 | [TK]D-Fender | omes: Here's some "inspiration" material for you to see what a simple setup might look like |
16:44.11 | [TK]D-Fender | ~jerjerguide |
16:44.11 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
16:44.15 | [TK]D-Fender | ^^^ |
16:44.41 | [TK]D-Fender | omes: Do not follow the installation procedure, but use this as a sample to help you understand the book as implemented in a small setup |
16:44.57 | omes | [TK]D-Fender: that link was perfect! |
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17:06.10 | omes | exten => _1NXXNXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN}) |
17:06.11 | *** join/#asterisk neurosys (~neurosys@adsl-8-187-44.mia.bellsouth.net) |
17:06.19 | omes | what does the underscore and X's mean here? |
17:06.31 | p3nguin | _ indicates a pattern match |
17:06.41 | omes | i mean N's |
17:06.42 | p3nguin | X represents numbers 0-9 |
17:06.44 | omes | the X'es i mean |
17:06.51 | omes | haha, the X'es i understand |
17:06.57 | [TK]D-Fender | omes: "_" means that what follows is a pattern and not just literal chars. X= any digit. N = 2-9 |
17:07.07 | p3nguin | N represents numbers 2-9 |
17:07.12 | [TK]D-Fender | omes: Time to go read chapter 5 <-------- |
17:07.18 | omes | ah, thanks :) |
17:07.38 | omes | [TK]D-Fender: i will definitely read the whole book! |
17:07.42 | p3nguin | and Z represents numbers 1-9, if I remember right. |
17:07.43 | omes | asterisk is great! |
17:10.27 | omes | exten => _XXXXXXXX,1,Dial(SIP/Phonzo/0047${EXTEN}) <-- so this would make a call to norway? |
17:11.00 | p3nguin | If the number starts with 0047 and needs another 8 numbers after that, probably. |
17:11.30 | omes | yeah, numbers in norway are 8 digits long and +47 sends them in the right direction ;) |
17:12.19 | omes | local, fetching registrations failed: 403 Forbidden |
17:12.20 | [TK]D-Fender | omes: Also depends on the format your provider requires |
17:12.25 | omes | seems i made a mistake somewhere |
17:12.39 | omes | [TK]D-Fender: i know 0047XXXXXXXX will work |
17:12.56 | omes | that's the error twinkle gave me |
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17:14.06 | omes | would one have to make a users.conf too? |
17:14.23 | p3nguin | Only if you wanted to use users.conf instead of the normal methods. |
17:14.43 | [TK]D-Fender | omes: your auth is bad.don't |
17:14.55 | [TK]D-Fender | ~users.conf |
17:14.55 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
17:16.08 | omes | well it still hates me :\ |
17:21.40 | omes | The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. |
17:21.45 | omes | hahaha, great quote there |
17:26.10 | omes | ERROR[9760]: pbx.c:3309 __ast_custom_function_register: Function MAILBOX_EXISTS already registered. <-- apparently that's where it kicks it |
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17:34.42 | [TK]D-Fender | omes: PASTEBIN "ls -la /usr/lib/asterisk/modules" |
17:34.44 | [TK]D-Fender | ~pb |
17:34.44 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
17:35.00 | omes | [TK]D-Fender: I had to use the default modules.conf |
17:35.17 | omes | then it worked, now i just need to get it running on a different ip so i can get my sip client up again |
17:35.57 | omes | again it forbids me access with the 403 error |
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17:36.48 | omes | http://pastie.org/951603 |
17:38.25 | [TK]D-Fender | omes: Twinkle is running on the same box as *? |
17:38.30 | omes | yes |
17:38.37 | omes | but i changed the ip for asterisk now |
17:38.49 | omes | so they're both up and running, back to the user authenticatoin problem |
17:39.01 | [TK]D-Fender | omes: pastebin your sip.conf entry for that device |
17:39.09 | omes | s/authenticatoin/authentication/ |
17:39.24 | omes | haha, nice bot :D |
17:39.55 | omes | http://pastie.org/951607 |
17:43.04 | [TK]D-Fender | omes: Try another client like Ekiga |
17:43.17 | omes | [TK]D-Fender: sure |
17:45.09 | omes | no sigar "Could not register" |
17:46.00 | omes | im gonna go eat, brb |
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17:58.13 | omes | there |
17:58.20 | omes | is there any way to debug the thing? |
17:58.36 | omes | so that i can see everything that's going on? like if its a password issue, username issue and so on |
18:03.54 | [TK]D-Fender | omes: use SIP DEBUG from * CLI, not an external dump. Try again, pastebin the attempt and your sip.conf |
18:04.08 | [TK]D-Fender | omealso your softphone should be running on a different port than 5060 |
18:06.39 | *** join/#asterisk anny__ (~chatzilla@80.79.159.226) |
18:06.47 | omes | haha sorry guys |
18:06.53 | omes | it *was* a twinkle problem after all |
18:07.02 | anny__ | hi all |
18:07.05 | omes | apparently twinkle binds to 0.0.0.0 too... |
18:07.17 | omes | Registered SIP '100' at 10.0.0.100 port 5061 |
18:07.19 | omes | voi voi |
18:07.24 | omes | im gonna try a call now wish me luck |
18:07.52 | anny__ | i'm new to astersik, can anyone tell me where can i find some documents to start reading |
18:08.12 | anny__ | possibly not as detailed as the official documents |
18:08.14 | pabelanger | ~bppl |
18:08.18 | pabelanger | ~book |
18:08.18 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
18:08.35 | anny__ | thx |
18:08.47 | omes | anny__: http://jeremy-mcnamara.com/asterisk/how-to-configure-asterisk/ <-- this really helped me too |
18:09.00 | anny__ | thx omes |
18:09.08 | omes | the book is not something you can do in an evening me thinks :P |
18:09.18 | anny__ | i know :) |
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18:09.28 | anny__ | just looking for a place to start |
18:09.41 | anny__ | but i've been browsing the official docs on the asterisk site |
18:09.59 | anny__ | i need something less detailed just to grasp the global picture |
18:19.06 | omes | http://pastie.org/951644 |
18:23.28 | omes | apparently that was a problem with my sip provider... |
18:23.43 | omes | is there some way to find out that didn't work with one of them and why it worked with another one? |
18:24.05 | omes | probably i should read the book and then ask the questions |
18:24.26 | omes | else #asterisk will be a concat of the pdf :P |
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18:27.40 | [TK]D-Fender | omes: pastebin the CLI SIP DEBUG (you eed to enable this) for your failed call |
18:28.56 | Nixy33 | Hi all, can anyone help a noob out here, got my asterisk setup working in our small office and trying to configure a remote extension. Using a Snom M3, have set the extention up and it's registering on the PBX. I can get my home number to ring (by dialing the prefix for the local analogue lines) but when I pickup there's no sound. If I try and use the voip prefix it all works fine. Also no core messages such as voicemail menu are audiable |
18:28.56 | Nixy33 | <PROTECTED> |
18:29.24 | omes | [TK]D-Fender: NICE |
18:29.32 | omes | [TK]D-Fender: you're an angel! it all works now anyways :) |
18:29.42 | omes | ill find out why the other provider borked later on |
18:31.19 | omes | just two last quick questions; how do i enable that, and how do i reload config files without restarting asterisk? |
18:32.01 | p3nguin | sip set debug |
18:32.03 | p3nguin | sip reload |
18:32.06 | p3nguin | dialplan reload |
18:32.10 | [TK]D-Fender | omes: Syntax varies by version. recently "sip set debug" |
18:32.18 | omes | No such command 'sip set debug' |
18:32.53 | omes | what about the extentions file, how do i reload that without restarting asterisk? |
18:33.00 | p3nguin | (1332.06) <p3nguin> dialplan reload |
18:33.05 | omes | ah, ok thanks :) |
18:43.16 | Nixy33 | any one got any ideas? |
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18:48.38 | pabelanger | Nixy33: Firewall or NAT issue |
18:49.42 | Nixy33 | thanks, so not a codec issue then? I'm baffled as to why the call when routed over VOIP(from the pbx) works but no local messages like "enter your voicemail password" work |
19:01.20 | [TK]D-Fender | nixpastebin a failed call with SIP DEBUG enabled |
19:01.22 | [TK]D-Fender | ~pb |
19:01.22 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
19:01.24 | [TK]D-Fender | ^^^^^ |
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19:23.10 | seanbright | anyone have any experience with lumenvox or vestec? |
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19:23.42 | seanbright | i have a requirement from a client that a customer be able to supply an arbitrary password to the system by speaking it |
19:24.07 | seanbright | but from the research i have done it looks like if the word isn't part of the supplied grammar, it's not getting recognized |
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20:12.41 | biscuit314 | Hi all! I'm hoping someone can help me with my AGI python script problem... |
20:13.07 | biscuit314 | I have a simple python script properly chmod'ed in the correct folder: /var/lib/asterisk/agi-bin/ |
20:13.31 | biscuit314 | my dialplan includes: exten => 789,n,AGI(agitest.py, 'test1', 'test2') |
20:13.41 | biscuit314 | The script itself (agitest.py) is as follows: |
20:14.08 | biscuit314 | import sys |
20:14.08 | biscuit314 | f = open('/home/michael/agidump.txt', 'w') |
20:14.08 | biscuit314 | for arg in sys.argv: |
20:14.09 | biscuit314 | <PROTECTED> |
20:14.09 | biscuit314 | f.close() |
20:15.03 | biscuit314 | I can run the script from terminal - no problems. But when I dial the extension I get... |
20:15.05 | biscuit314 | <PROTECTED> |
20:15.05 | biscuit314 | <PROTECTED> |
20:15.37 | biscuit314 | I have successfully run PERL scripts, but no luck at all with python. |
20:15.43 | biscuit314 | What am I doing wrong? |
20:16.20 | seanbright | not putting a #!/usr/bin/python as the first line? |
20:16.48 | biscuit314 | slaps forehead... I'll try that... |
20:17.05 | seanbright | and it should be executable |
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20:17.15 | seanbright | and when you paste stuf like that in here, use a pastebin |
20:17.16 | seanbright | ~pb |
20:17.16 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
20:19.32 | biscuit314 | That did the trick.... thank you so much! (and my apologies for pasting 3+ lines... newbie mistake...) |
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20:39.39 | shader | what's the bare minimum of configuration required to get asterisk to start? |
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20:41.30 | xheliox | Hey! It's really ambiguous day, everyone join in! |
20:42.21 | seanbright | shader: a cpu |
20:42.32 | xheliox | some sort of storage device. |
20:44.40 | shader | lol |
20:45.27 | shader | I cleared out the config folder, because I wasn't sure if any of the generated sample config files was interfering with my test extensions.conf and sip.conf, but then of course asterisk wouldn't start |
20:45.49 | shader | so now I'm using the samples again, and I wish I knew which files I didn't actually need |
20:46.03 | shader | and which I have to fill in myself |
20:51.36 | [TK]D-Fender | shader: modules.conf. ANy module you load may require its config file or seize |
20:52.17 | [TK]D-Fender | shader: Beginners should install the samples and just trash the 2 files you said you already did yourself. |
20:52.26 | [TK]D-Fender | shader: so start with the samples for the rest |
20:56.20 | shader | ok |
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21:09.01 | krenel | hello |
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21:10.23 | wcselby | ~sipnat |
21:10.23 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
21:10.54 | krenel | would it be possible to use a separate asterisk box for voicemail only and still get working MWI on the first box? |
21:13.03 | xuser | ~itsp |
21:13.03 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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. |
21:13.14 | xuser | ~itsplist-us |
21:13.14 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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21:53.09 | MiserySoft | Just bought a TDM400 clone card on eBay with 2 FXO + 2 FXS for $80 .. very impressed. |
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22:30.42 | Bartockbatz | anyone familiar with the SIP provider Axialys? |
22:32.00 | Bartockbatz | for some damn reason, I cannot get my Asterisk server to register |
22:33.50 | Bartockbatz | I can get my Sipgate trunk to work just fine - something I should know about non -NANPA numbers? Does it make a difference? |
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22:38.09 | rare1980_ | hi all ...... on asterisk CLI i am getting this error msg |
22:38.10 | rare1980_ | ast_carefulwrite: write() returned error: Broken pipe |
22:38.19 | rare1980_ | any idea what this msg means? |
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22:49.23 | omes | hahaha, asterisk is great fun |
22:49.31 | omes | im seding faxes over my VoIP line now... |
22:49.52 | omes | if they're longer than just a line of text they tend to fail though ^^ |
22:50.28 | omes | Bartockbatz: try to change the useragent |
22:50.41 | omes | my sip provider blocked asterisk, so i had to change it.... |
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22:53.10 | BarthezZ | yeah omes, FoIP is just a pain in the ass which tends to fail :p |
22:53.57 | omes | i suppose FoIP would be good (T.38) :P |
22:54.21 | omes | this is Fax over VoIP :P |
22:54.21 | TJNII | Why would a provider block asterisk? |
22:54.21 | TJNII | I'd tell them to FOAD. |
22:54.21 | omes | TJNII: i ask myself the same question :S |
22:54.24 | BarthezZ | yeah true omes :p, still sucks ^^ |
22:54.51 | omes | indeed :P |
22:55.36 | BarthezZ | faxes should just be banned |
22:55.42 | BarthezZ | use freaking email people! |
22:55.56 | omes | well, you could tunnel your emails over fax |
22:56.11 | omes | Email-to-fax and Fax-to-email on the other end |
22:56.27 | omes | prays, please dont die this time :( |
22:56.39 | omes | my quality setting is set to 10, that doesnt help much lol |
22:56.39 | BarthezZ | or just leave the fax part out, email to email :) |
22:56.41 | TJNII | Analog faxes should be banned. Put a friggin NIC in them already. Its 2010! |
22:57.01 | omes | yay, it works! |
22:57.19 | omes | HAHAHA |
22:57.26 | omes | the quality is so bad i just have to laugh :P |
22:57.39 | BarthezZ | quality of the fax? :p |
22:57.43 | omes | yeah :P |
22:58.09 | BarthezZ | what app do you use to send/receive them? |
22:58.25 | omes | ill have some show and tell soon |
22:58.28 | omes | sendfax |
22:58.48 | omes | a client for hylafax, with iaxmodem as the modem and asterisk as the phoneline |
22:58.59 | BarthezZ | hehe oke |
22:59.22 | omes | that is, asterisk as my operator |
22:59.31 | omes | my WLAN is my phone line, at least parts of the way |
22:59.38 | omes | i guess that doesnt really improve the quality :P |
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23:08.47 | omes | http://i40.tinypic.com/11vma6w.jpg |
23:08.51 | omes | here's my show and tell |
23:08.56 | omes | Fax over VoIP |
23:13.06 | *** join/#asterisk BarthezZ (~bart@ipd50a21c9.speed.planet.nl) |
23:13.21 | BarthezZ | omg there was a glitch in the (power) matrix :p |
23:14.23 | omes | indeed :P |
23:14.27 | omes | oh noez! :( |
23:14.32 | omes | 05/09/10 02 07 48 < omes> http://i40.tinypic.com/11vma6w.jpg |
23:14.32 | omes | 05/09/10 02 07 52 < omes> here's my show and tell |
23:14.32 | omes | 05/09/10 02 07 57 < omes> Fax over VoIP |
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23:14.54 | BarthezZ | lol, fail :p |
23:14.54 | omes | i find it hillarious i even got to send an iamge over my wlan.... |
23:15.00 | omes | it shouldnt be possible... |
23:15.17 | BarthezZ | yeah, totally beats tcp/ip with ftp/http/<about any ip protocol> ;p |
23:15.17 | omes | actually its great success! |
23:15.39 | omes | well... considering what could go wrong im amazed it went through... |
23:15.57 | omes | SIP/Phonzo-00000001 answered IAX2/fax-3638 |
23:15.59 | BarthezZ | true :p |
23:16.05 | omes | waiting for it to finish then im going to bed :P |
23:16.25 | omes | Hungup 'IAX2/fax-3638' yay |
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23:17.54 | omes | i think im gonna write a howto on how i did this |
23:18.01 | omes | it might even be phun :D |
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23:42.13 | Slugs_ | hi |
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