| 00:01.05 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 00:02.50 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 00:02.55 | bbourdage | When I go to VoiceMail(ext,u), the persons unavailable message starts, plays about 3 seconds stops, and then plays the please leave a message after the tone ?, I have triple checked and the unavailable message is much longer |
| 00:27.32 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 00:31.12 | *** join/#asterisk chasing`Sol (~cS@197.132.19.174) |
| 00:38.37 | woleium | I have a problem with unreachable polycom peers. 'sip show peers' lists their IP, and I can ping them from the server, but they appear unreachable. I have one handset on a remote subnet that works fine, and swapping configs between phones I can make any extension work on the one handset that works (i.e. it's not the config) any ideas? |
| 00:41.02 | *** join/#asterisk RickCogley (~anonymous@EM114-48-160-57.pool.e-mobile.ne.jp) |
| 00:41.33 | *** join/#asterisk Te3-BloodyIron (~no@S01060004e21ea5b9.cg.shawcable.net) |
| 00:43.47 | Te3-BloodyIron | hi so i have an asterisk system connecting to a trunk (on another system we use). when the asterisk system registers it uses the proper realm, however when i try to make a call, the invite doesn't use the correct realm |
| 00:43.59 | Te3-BloodyIron | instead it uses the IP of the first asterisk system, but i want it to use the realm as defined in the trunk |
| 00:44.09 | Te3-BloodyIron | which would be the hostname of the target system |
| 00:51.05 | woleium | it was permissions in sip_custom.conf |
| 00:51.09 | woleium | localnet |
| 00:51.12 | woleium | d'oh |
| 00:55.21 | p3nguin | We could have never guessed that, since that is not an asterisk file. |
| 00:56.29 | WIMPy | enoch: So it's 2 FXS, not 2 FXO you're after? Makes more sense. If you want it real cheap, use a single BRI card and a two port ATA. Or a multi BRI and a terminal adaptor. |
| 01:00.36 | bbourdage | When I go to VoiceMail(ext,u), the persons unavailable message starts, plays about 3 seconds stops, and then plays the please leave a message after the tone ?, I have triple checked and the unavailable message is much longer |
| 01:08.08 | *** join/#asterisk celord (~cesar@celord.ice.co.cr) |
| 01:08.33 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 01:09.16 | asteriskmonkey | anyone have experience with the digium fax stuff? I keep getting receivefax_exec: error initializing channel ... in t38mode even though I have ulaw audio fallback set on |
| 01:10.45 | asteriskmonkey | also get messages like error reading frame while generating CED tone on.. |
| 01:16.23 | ruben23 | guys anyone can help, my asterisk CLI are being populated by this error message ----> http://pastebin.com/xMvWdRcW |
| 01:17.33 | WIMPy | ruben23: Tell 3025 to check their password. |
| 01:21.09 | ruben23 | WIMPy: its suppost to be on local only, but the one accessing is public Ip seems it is a hack..? |
| 01:21.35 | WIMPy | That's the way the internet works. |
| 01:24.02 | ruben23 | WIMPy: any solution to solve this warning, should i delete the extensions..? |
| 01:24.03 | p3nguin | Good ol' internets. |
| 01:24.20 | p3nguin | Extensions have nothing to do with your device trying to register with a bad password. |
| 01:24.29 | p3nguin | Not a damn thing. |
| 01:24.40 | WIMPy | I'd go and read some Asterisk secutity stuff to find out why the guessed a correct username. |
| 01:25.00 | p3nguin | Oh, oh, oh... I know!!! |
| 01:25.13 | p3nguin | Didn't use a good enough device name! |
| 01:27.40 | *** join/#asterisk celord (~cesar@celord.ice.co.cr) |
| 01:31.10 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 01:32.13 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:32.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:35.42 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 01:59.41 | *** join/#asterisk RickCogley (~anonymous@pc2.cook-japan-3pl-unet.ocn.ne.jp) |
| 02:02.33 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 02:16.12 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-190-50.hsd1.va.comcast.net) |
| 02:41.28 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 02:44.36 | *** join/#asterisk revolve (~3@cpc11-bagu10-2-0-cust741.1-3.cable.virginmedia.com) |
| 02:50.23 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 03:12.55 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:29.05 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 03:33.08 | *** join/#asterisk bmg505 (~leon@196-209-84-63.dynamic.isadsl.co.za) |
| 03:36.05 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 03:58.53 | *** join/#asterisk Steel_Reign (~Steel_Rei@72-28-219-021-dhcp.mia.fl.atlanticbb.net) |
| 04:00.26 | Steel_Reign | hey all. I got a question for everyone. what's the best way to setup extensions? should I give full numbers? xxx-xxx-xxxx or just extensions? xxxx |
| 04:01.56 | bbourdage | Hello Steel, I am new to Asterisk , but if you did 10 digit extensions, then everyone at a station phone would have dial that every time ?. If this is a big install base 5,000 users, maybe. How many users |
| 04:02.45 | p3nguin | steel_reign: Full numbers *ARE* extensions. Extensions are whatever you configure them to be. |
| 04:03.03 | *** join/#asterisk slingr (santas@will.one.day.hack-the-pla.net) |
| 04:03.40 | Steel_Reign | well isn't that what SIP Alias is for? so I could just put the end of the number there and if that is dialed then it would be routed to that person |
| 04:04.12 | Steel_Reign | @p3nguin. i know sorry wrong use of terminology |
| 04:05.26 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 04:06.21 | Steel_Reign | anyone got an answer? |
| 04:07.01 | ChannelZ | I don't really understand the question |
| 04:07.27 | ChannelZ | As p3nguin said extensions can be whatever you want them to be. |
| 04:08.01 | ChannelZ | I personally don't like calling my devices numbers just to make the dialplan easier, but plenty of people seem to. |
| 04:08.31 | p3nguin | There's no such thing as "sip alias" in asterisk. |
| 04:08.41 | ChrisInSydney | Steel_Reign: Are you referring to your sip.conf handsets or the internal dialling. eg extn 201 calls 202 and transfers a call ? |
| 04:09.06 | p3nguin | sip.conf has nothing to do with extension 201 nor 202. |
| 04:09.16 | p3nguin | Extensions are configured in extensions.conf. |
| 04:09.23 | ChrisInSydney | ..assuming you are using SIP handsets |
| 04:09.44 | p3nguin | ~devices |
| 04:09.45 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 04:09.46 | ChrisInSydney | True |
| 04:10.50 | p3nguin | If you want to configure extension 201 to call a SIP phone by the name of 000011112222, that's done with a Dial() command. |
| 04:11.25 | p3nguin | exten => 201,1,Dial(SIP/000011112222,30) |
| 04:11.31 | p3nguin | This is extension 201. |
| 04:11.44 | p3nguin | ~book |
| 04:11.44 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:11.47 | p3nguin | Start reading. |
| 04:13.08 | ChrisInSydney | p3nguin: trouble is, Asterisk (and telephony for that matter) for the newbie is a bit like a George Clinton / Parlaiment / Funkadelic gig. You have to experience it and tehn it becomes ovbvious |
| 04:13.30 | ChrisInSydney | :D |
| 04:14.14 | revolve | ChrisInSydney: you sir are a legend |
| 04:14.54 | ChrisInSydney | revolve: Only in my lunch break |
| 04:15.33 | revolve | wouldnt happen to know a good sip broker for someone who'll only ever be rarely making calls across the planet from the uk wouldja? |
| 04:16.44 | p3nguin | ~itsplist-uk |
| 04:16.44 | infobot | methinks 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. |
| 04:16.44 | ChrisInSydney | revolve: UK to Aus ? |
| 04:17.02 | revolve | ChrisInSydney: to south africa first and probably to the usa a fair bit |
| 04:17.20 | revolve | infobot: !!! thankyou |
| 04:17.41 | revolve | ah |
| 04:17.52 | revolve | only one i've not seen before is gradwell. tyvm nonetheless |
| 04:17.54 | p3nguin | If I had the infrastructure or the money to lease it from someone else, I'd have an ITSP in the UK. |
| 04:18.51 | ChrisInSydney | The lag is a bit of a killer to SA to use a local ITSP over there |
| 04:19.39 | p3nguin | I think you'll need an ITSP that is local to you and let them terminate over landlines (or however th[D[D[D[D[Dey see fit). |
| 04:19.48 | p3nguin | hmm? |
| 04:19.54 | p3nguin | Let me try that again... |
| 04:20.02 | p3nguin | I think you'll need an ITSP that is local to you and let them terminate over landlines (or however they see fit). |
| 04:20.52 | ChrisInSydney | I had an Unkle in South Africa. He used to say: "Just because I am from South Africa and just because I am white, people accuse me of being a racist, well I'm not, because everybody knows that racisim is a crime and everybody knows that crime is for blecks" |
| 04:21.03 | revolve | rifk |
| 04:21.13 | revolve | aunt is in SA but she's a weird jesusfreak |
| 04:21.56 | ChrisInSydney | "wierd jesusfreak": is there any oher kind |
| 04:21.57 | ChrisInSydney | ?? |
| 04:22.08 | ChrisInSydney | oh yeah, scary jesusfreak |
| 04:22.34 | revolve | lawl |
| 04:22.53 | revolve | well she used to really put out in her 20s according to my gran, so weird as in hypocritical I guess |
| 04:23.12 | revolve | morally reproving and full of shit. |
| 04:23.13 | *** join/#asterisk RickCogley (~anonymous@EM1-113-168-95.pool.e-mobile.ne.jp) |
| 04:23.51 | ChrisInSydney | My uncle was an agnostic, or as he put it, a God fearing Athiest |
| 04:24.06 | revolve | lol'd |
| 04:24.50 | revolve | relativist here |
| 04:25.01 | ChrisInSydney | He also used to say that if the Jews is Israel and Muslims in Palestine and Lebanon could be more Christan towards eachother, then that could be the start of peace in the middle east |
| 04:25.19 | revolve | hummus. |
| 04:25.22 | revolve | peas in the middle east |
| 04:25.30 | ChrisInSydney | :D |
| 04:25.33 | revolve | :D |
| 04:25.44 | ChrisInSydney | I dont know why I am laughing..... |
| 04:25.45 | ChrisInSydney | but I am |
| 04:25.51 | revolve | \o/ |
| 04:26.17 | revolve | http://www.youtube.com/watch?v=xzEJvHE_rEg |
| 04:27.36 | ChrisInSydney | Now thats how they need to teach about contexts, devices, extensions and dial plans |
| 04:28.10 | revolve | hahaha |
| 04:28.31 | revolve | yes. that would... man. been writing some lyrics lately and yeah, someone should do it |
| 04:28.47 | revolve | it's how i learned about atrazine the other day lol |
| 04:28.58 | ChrisInSydney | xD |
| 04:28.58 | revolve | ChrisInSydney: http://www.youtube.com/watch?v=3MxrH4lN0-A |
| 04:29.33 | revolve | he's a scientist who discovered that it was turning male frogs into females who had ovaries, eggs and gave birth |
| 04:30.22 | ChrisInSydney | Off topic, but...(means I dont care)...did he take lessons from Tony Iommi, or the other way around, |
| 04:30.33 | revolve | no idea who that is sorry |
| 04:30.35 | ChrisInSydney | and tripple humbuckers in a strat |
| 04:30.45 | ChrisInSydney | lead guitarist for black sabbath |
| 04:30.46 | revolve | oh, black sabbath |
| 04:30.47 | revolve | yeah |
| 04:30.56 | revolve | the funkadelic guitarist? dunno |
| 04:31.01 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 04:31.20 | ChrisInSydney | I was referring to his technique, choice of notes |
| 04:31.38 | *** join/#asterisk teratoma (~teratoma@i.dont.get.mad.i.get.stabby.net) |
| 04:31.54 | revolve | do you play guitar? |
| 04:32.12 | ChrisInSydney | He has a similar feel |
| 04:32.17 | revolve | yeah |
| 04:32.44 | revolve | it seems like a really natural arrangement that has probably manifested through tonnes of guitarists all seperate and slightly distinct from eachother |
| 04:32.57 | revolve | felt like something i'd perceived before but i know i've not heard it before |
| 04:35.00 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 04:35.08 | ChrisInSydney | Some of the licks are quite like T I. Boxed runs....BTW love the stuff around at 8:00 with the two of them going off |
| 04:35.15 | revolve | amen |
| 04:35.42 | ChrisInSydney | there goes the Hendrix at 8:50 |
| 04:36.21 | ChrisInSydney | revolve: You have made my day. I am grinning from ear to ear. |
| 04:36.30 | *** join/#asterisk ccesario_ (~ccesario@187.17.166.162) |
| 04:36.48 | ChrisInSydney | must go to a meeting now. 4:00pm at a pub |
| 04:37.02 | ChrisInSydney | c yaz |
| 04:37.09 | slingr | cya |
| 04:37.51 | revolve | .o/ glad i could help |
| 04:39.38 | *** join/#asterisk damjan (~damjan@unaffiliated/damjan) |
| 04:43.36 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 04:46.48 | *** join/#asterisk nobodyshome (~root@bas10-kitchener06-1176001622.dsl.bell.ca) |
| 04:46.53 | nobodyshome | soup all!? |
| 04:47.02 | dijib | soup |
| 04:47.09 | p3nguin | soup! |
| 04:47.13 | p3nguin | No soup for you! |
| 04:47.14 | dijib | SOUP! |
| 04:47.24 | dijib | How is everyone? |
| 04:47.30 | p3nguin | Still IRCing as root, I see. |
| 04:47.38 | dijib | how in the? |
| 04:47.53 | dijib | how do you even know that? |
| 04:48.13 | dijib | Asterisk is still going strong |
| 04:48.17 | p3nguin | CTCP USERINFO reply from dijib: root |
| 04:48.33 | dijib | i dont even knw |
| 04:48.47 | dijib | hows 2012 treating you p3nguin |
| 04:49.00 | dijib | SeRi still around? |
| 04:49.04 | p3nguin | Seems to be standard issue. |
| 04:49.13 | p3nguin | He isn't around too often. |
| 04:49.21 | dijib | thanks for the vote of confidence |
| 04:49.39 | dijib | if only i could remember his sip uri i would bother him |
| 04:50.13 | dijib | so i got some interesting audio monitored the other day, so im hoping it helps in my battle will keep you posted p3 |
| 04:50.16 | *** join/#asterisk RickCogley_ (~anonymous@EM1-113-168-95.pool.e-mobile.ne.jp) |
| 04:51.45 | p3nguin | I bet you run asterisk as root, too. |
| 04:51.50 | dijib | nope. |
| 04:51.59 | slingr | lol |
| 04:52.32 | dijib | ive got an issue with my shit shaper though, every couple of days y firewall needs resetting. |
| 04:52.36 | dijib | any ideas |
| 04:52.45 | p3nguin | He who IRCs as root will run anything as root. |
| 04:53.23 | slingr | heh |
| 04:53.46 | slingr | hey p3nguin, do you know if there is any other places to get freepbx on irc other than this network. i'm having trouble and i never get answers there |
| 04:53.56 | dijib | crap i think y1659 asterisk 20 0 35844 15m 5624 S 0.7 3.1 9:02.51 asterisk |
| 04:54.00 | dijib | 1659 asterisk 20 0 35844 15m 5624 S 0.7 3.1 9:02.51 asterisk |
| 04:54.08 | p3nguin | I wouldn't have any clue about FreePBX crap. |
| 04:54.25 | dijib | which is worse freepbx or asterisknow? |
| 04:54.40 | p3nguin | That's like comparing leather to a recliner. |
| 04:54.55 | dijib | see i have no clue |
| 04:55.01 | p3nguin | AsteriskNOW is a Linux/Asterisk distribution. |
| 04:55.08 | dijib | ahh ok |
| 04:55.22 | dijib | did they move away from trunk? |
| 04:55.36 | p3nguin | It offers a choice of FreePBX, the Digium/Asterisk GUI, or no GUI at all. |
| 04:55.45 | p3nguin | Trunk what? |
| 04:57.06 | slingr | <p3nguin> That's like comparing leather to a recliner. < this is going in my quote book |
| 04:58.46 | dijib | oh i didnt even know that the two were gui's |
| 04:58.56 | dijib | nevermind im wasting everyone time and money as usual |
| 04:59.09 | p3nguin | You're using up all my bits. |
| 04:59.29 | p3nguin | I was going to listen to music with that bandwidth. |
| 05:09.14 | *** join/#asterisk xertres (~xertres@h-136-1-238.a176.priv.bahnhof.se) |
| 05:12.12 | [TK]D-Fender | slingr: Big reason not to expect to have gotten very far : you haven't shown anything at all for people to help you with.. |
| 05:14.39 | dijib | http://www.reuters.com/article/2012/02/14/russia-submarine-nuclear-idUSL5E8DE7U920120214 |
| 05:14.42 | dijib | nice |
| 05:17.58 | slingr | aye aye [TK]D-Fender, i'll see what else I can dig up. |
| 05:22.16 | *** join/#asterisk gladier (~gladier@pedobear.molested.me) |
| 05:23.30 | gladier | hey guys - what would make asterisk not sent a 407 - auth required to a particular peer |
| 05:23.41 | gladier | other than the obvious insecure=invite |
| 05:24.24 | [TK]D-Fender | gladier: that is the point of it... |
| 05:25.21 | *** join/#asterisk Tyrael1 (~Ryan@c-50-129-214-142.hsd1.in.comcast.net) |
| 05:26.21 | gladier | no i understand that. what i dont understand is why asterisk isn't sending a 407 to this peer when i dont have any insecure in my config |
| 05:26.53 | dijib | http://bambuser.com/v/2368801 |
| 05:26.57 | dijib | anybody else seen this? |
| 05:27.06 | dijib | somewhere in syria |
| 05:27.19 | dijib | live |
| 05:27.34 | revolve | shit |
| 05:27.37 | revolve | thanks dijib |
| 05:30.50 | [TK]D-Fender | dijib: My heart goes out to the Syrian people.... I hope they can gather enough forces to stand up for themselves and fight to take their country back inthe hands of the people. |
| 05:31.12 | revolve | dijib: got any more? |
| 05:32.08 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 05:32.23 | *** join/#asterisk gladier (~gladier@pedobear.molested.me) |
| 05:33.18 | gladier | ok so now my irc client has finished having a spak attach... anyone have any reason that asterisk wouldn't send a 407 auth required to one peer when there is no insecure=blah in the config? |
| 05:34.04 | *** join/#asterisk Micc (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 05:34.17 | p3nguin | No secret. |
| 05:34.34 | Micc | whats the syntax for including a a res_config_curl in extensions.conf I have to add a switch = realtime/?? |
| 05:38.08 | *** join/#asterisk damjan (~damjan@unaffiliated/damjan) |
| 05:43.03 | Micc | can I use res_config_curl to put my whole extensions.conf on an http server? |
| 05:43.42 | dijib | revolve: nope |
| 05:46.24 | revolve | k |
| 05:51.21 | *** join/#asterisk bmg505 (~leon@196-209-84-63.dynamic.isadsl.co.za) |
| 05:53.30 | dijib | revolve: where are you from? |
| 05:54.46 | revolve | dijib: why? |
| 05:56.49 | revolve | dijib |
| 05:58.35 | p3nguin | You must be new to socialization. |
| 05:59.33 | revolve | who? |
| 05:59.39 | revolve | this is IRC |
| 05:59.41 | carrar | who's on first |
| 05:59.45 | revolve | what's on second |
| 05:59.56 | p3nguin | I don't know! |
| 06:06.06 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-oagbnlabadxohbai) |
| 06:06.52 | dijib | home run |
| 06:07.09 | revolve | dijib: from the uk |
| 06:07.11 | revolve | why? |
| 06:07.28 | revolve | well |
| 06:07.33 | revolve | nevermind |
| 06:07.35 | revolve | ignore that |
| 06:07.37 | revolve | where are you from? |
| 06:09.29 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 06:12.39 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 06:12.52 | dijib | canada |
| 06:14.21 | slingr | whereaboots? |
| 06:16.14 | carrar | Oh |
| 06:16.16 | carrar | North USA |
| 06:20.12 | [TK]D-Fender | carrar: We're bigger, and we're on top. If this were prison you'd be our BITCH :p |
| 06:20.25 | carrar | heh |
| 06:20.31 | slingr | haha |
| 06:20.40 | carrar | but in reality, you know you are really out bitch! |
| 06:20.44 | carrar | our |
| 06:20.46 | slingr | canucks outnumber the yanks in this convo 3-1 :D |
| 06:20.57 | carrar | everyone is our Bitch |
| 06:21.12 | carrar | till China takes over the world |
| 06:22.20 | carrar | Lets see if the pres can do even worse |
| 06:22.29 | carrar | err |
| 06:22.36 | carrar | Lets see if the NEXT president can do even worse |
| 06:22.55 | carrar | it's a trend I believe |
| 06:24.26 | slingr | next president? in 2016? |
| 06:25.23 | [TK]D-Fender | RP 2012! |
| 06:26.20 | ChannelZ | heh - Ron Paul will be in a group home by then |
| 06:28.10 | Maliuta | President of what? McDonalds? |
| 06:28.33 | Maliuta | The only presidents we have here are of the corporate kind. |
| 06:48.01 | *** join/#asterisk ziz212 (~chatzilla@203.115.2.22) |
| 06:52.09 | *** join/#asterisk _gm (73ba8fec@gateway/web/freenode/ip.115.186.143.236) |
| 06:56.07 | ziz212 | Dear All, I am writing a dial plan for asterisk and I need to check user entered dtmf number is a valid extension in created in the elastix. I have searched this in the google but i didn't got any information for this. Pls some one help for this. |
| 06:57.12 | kaldemar | ziz212: core show function DIALPLAN_EXISTS |
| 06:58.21 | *** join/#asterisk RickCogley (~anonymous@114.179.8.185) |
| 06:59.14 | kaldemar | ziz212: you could use it like Goto(${IF($[${DIALPLAN_EXISTS(yourcontext,${GIVENEXTEN})}]?yourcontext,${GIVENEXTEN}:invalid)}) |
| 06:59.34 | kaldemar | or something like that. |
| 07:11.57 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 07:12.02 | v0lZy | Hello |
| 07:12.10 | v0lZy | hi ChrisInSydney |
| 07:12.24 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:12.26 | ziz212 | @kaldemar Thanks for the comment. But I cant understand the commands. Pls be kind enough to explain little bit. |
| 07:13.26 | kaldemar | ziz212: to see documentation for the commands in asterisk CLI, use commands "core show application Goto", "core show function IF", "core show function DIALPLAN_EXISTS" |
| 07:14.33 | kaldemar | GIVENEXTEN is just a variable, assuming you already know how to get user input to a variable. if not, use application Read. |
| 07:14.39 | ziz212 | Let say my Read function read the dtmf to a variable like My_rxnum, then in here what your doing is check $My_rxnum inside the dial plan ? isnt it? |
| 07:14.59 | kaldemar | you could do it all with ReadExten too if it fits your purposes. |
| 07:15.46 | ziz212 | isnt it? |
| 07:16.56 | kaldemar | the command i have checks if the user-entered extension exists in a context called "yourcontext", and goes to the extension if it does exist. if it doesn't, go to label "invalid". |
| 07:17.09 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:17.11 | schmidts | good morning |
| 07:17.19 | ziz212 | thanks for your generous help |
| 07:17.23 | kaldemar | read the application and function docs, they will explain the commands in more detail. |
| 07:18.21 | ziz212 | in that case what should be the "yourconext"? I am using elastix as my asterisk server |
| 07:19.42 | kaldemar | ziz212: you should know since it is your dialplan. |
| 07:21.26 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:23.45 | ziz212 | @kaldemar: ok , how could i get the list of extensions created in the system or what will be the context that have all created extensions? Sorry for bothering you. I am really having a problem in this? |
| 07:25.23 | ziz212 | My target is to read the dtmf that entered by user and check the with some x (x could be file or some thing that i dont know) weather it is a created extension with my elastrix box. |
| 07:25.28 | kaldemar | ziz212: "dialplan show" in CLI. |
| 07:25.49 | kaldemar | ziz212: i don't know how elastix builds its dialplan since i don't use it. |
| 07:26.43 | ziz212 | how astersik hold created extensions ? |
| 07:27.27 | kaldemar | ziz212: as far as i know elastix uses freepbx, so folks at #elastix or #freepbx know better what the dialplan looks like. |
| 07:27.28 | *** join/#asterisk xertres (~xertres@mgt01.telecomputing.se) |
| 07:28.18 | ziz212 | is it in a dial plan then I can use your entered DIALPLAN_EXISTS |
| 07:28.31 | kaldemar | ziz212: what you need to check is a context, as in "yourcontext" in the example that i gave earlier. dialplan consists of contexts that hold extensions. |
| 07:28.51 | kaldemar | yes, those were all dialplan commands. |
| 07:35.18 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:37.02 | ziz212 | Can i write my dial plan and put to pastbin? |
| 07:37.24 | ziz212 | @kaldemar: Can i write my dial plan and put to pastbin? |
| 07:38.12 | kaldemar | you better continue in #elastix or #freepbx, they know the internals better. |
| 07:43.11 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:51.59 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 07:52.02 | *** join/#asterisk the_tux (~suraj@49.248.152.176) |
| 07:52.04 | the_tux | hi all |
| 07:52.40 | the_tux | need one quick help in understanding the meaning of this "[directories](!)" what this means? |
| 07:52.46 | the_tux | its in asterisk.conf |
| 07:53.13 | the_tux | I am confused about (!) symbol at the end.. is it kind of not to use this ? |
| 07:53.30 | ziz212 | ok thanks |
| 07:54.02 | WIMPy | the_tux: It's a template. Later used with [something](template) |
| 07:55.08 | the_tux | WIMPy, so its just template and is it playing any role in current asterisk setup or its just for reference for creating / configuring the stuffs? |
| 07:56.36 | WIMPy | I don't understand that question. You can check for the occurance of its use as I just wrote. |
| 07:58.06 | kaldemar | the_tux: (!) disables a context or makes it a template that another context can inherit. |
| 07:58.15 | the_tux | ok thanks WIMPy got the answer... I was worried if that means not to use :) |
| 07:58.21 | kaldemar | oh, i was echoing. |
| 07:59.01 | the_tux | kaldemar, nice to see you after long time :) |
| 07:59.19 | WIMPy | Never thought of maing a section a template to disable it. Interesting idea. |
| 08:00.33 | *** join/#asterisk Nasga (~Nasga@199.79.125.78.rev.sfr.net) |
| 08:01.13 | the_tux | so kaldemar it means the section under [directories](!) can be used as template but is not playing any role at this moment in the configuration. All the stuffs in that context are ideal unless I am removing (!) from "[directories](!)"... (was that confusing? :( ) |
| 08:02.14 | WIMPy | No, you include the settings of the template by adding it to another section. |
| 08:02.42 | v0lZy | hi WIMPY, hi kaldemar |
| 08:02.43 | WIMPy | [template](!) and [somethingelse](template) |
| 08:02.55 | WIMPy | That includes template in to somethingelse. |
| 08:08.24 | the_tux | cool !! thanks WIMPy for describing in simple :) |
| 08:08.38 | *** join/#asterisk anny__ (~chatzilla@77.42.241.114) |
| 08:09.36 | anny__ | how can we see what are the installed and working codecs in asterisk, i tried (core show codecs) but a message was displayed at the top stating that this list is an information and does not reflect the current configuration |
| 08:11.32 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
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| 08:17.14 | Micc | why does realtime curl add /static?file= to the url? |
| 08:21.39 | WIMPy | anny__: try 'core show translation' |
| 08:22.13 | v0lZy | guys |
| 08:22.24 | v0lZy | has anyone ever used gigaset DECT's ? |
| 08:22.31 | v0lZy | specifically A510IP ? |
| 08:22.42 | anny__ | WIMPy: i tried it, it show the translation time between codecs but no video codec appears in the list |
| 08:22.51 | v0lZy | I'm having this weird issue where i cant access it if im on a different subnet |
| 08:23.01 | v0lZy | I mean, i cant access the web interface. |
| 08:23.17 | *** join/#asterisk Wolfeyes (~Wolfeyes@41-133-97-245.dsl.mweb.co.za) |
| 08:23.54 | WIMPy | anny__: I've never bothered about video, so I could be wrong, but as far as I know no video codecs are supported, it's all just pass through. |
| 08:24.09 | *** part/#asterisk Wolfeyes (~Wolfeyes@41-133-97-245.dsl.mweb.co.za) |
| 08:24.25 | WIMPy | v0lZy: Sounds like a normal networking issue to me. |
| 08:24.32 | WIMPy | i.e. routing |
| 08:24.41 | v0lZy | its not though |
| 08:24.45 | v0lZy | I can ping it and everything |
| 08:24.49 | v0lZy | I just cant open the web interface. |
| 08:24.58 | v0lZy | i can access the pbx fine |
| 08:25.01 | WIMPy | What happens? |
| 08:25.02 | anny__ | WIMPy: yes maybe you're right it's passthrough |
| 08:25.11 | v0lZy | the connection to the web interface simply times out |
| 08:25.20 | v0lZy | for example, my pbx is on 192.168.1.2 |
| 08:25.31 | v0lZy | and im on 192.168.2.150 |
| 08:26.16 | v0lZy | and my DECT is on 192.168.1.3 |
| 08:26.25 | v0lZy | i can access 192.168.1.2 just fine (http or https) |
| 08:26.37 | v0lZy | i cant access 192.168.1.3 |
| 08:26.51 | v0lZy | (it only has http anyway) |
| 08:26.58 | v0lZy | i can ping it just fine |
| 08:26.58 | WIMPy | But you can ping .3? |
| 08:27.04 | v0lZy | yeah |
| 08:27.58 | anny__ | WIMPy: how can we know why video negotiation has failed? |
| 08:28.08 | WIMPy | Either it's some strange secutity "feature", or a compatibility issue. |
| 08:28.27 | WIMPy | I've seen some devices not coping with ECN enabled hosts. |
| 08:29.23 | WIMPy | anny__: i have to pass that one. |
| 08:30.01 | anny__ | WIMPy: thx anyway |
| 08:31.05 | v0lZy | hm |
| 08:31.07 | v0lZy | w8 |
| 08:31.09 | v0lZy | hold on |
| 08:31.11 | v0lZy | now i assigned it a static ip |
| 08:31.14 | v0lZy | instead of dhcp |
| 08:31.18 | v0lZy | and i cant ping it... |
| 08:31.37 | v0lZy | not even from the router |
| 08:31.40 | v0lZy | which is super strange |
| 08:31.43 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:32.45 | v0lZy | though i think im getting to the bottom of it... |
| 08:35.00 | v0lZy | hm |
| 08:35.02 | v0lZy | guess |
| 08:35.02 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:35.03 | v0lZy | not |
| 08:35.09 | v0lZy | my router is set to 192.168.1.72/24 |
| 08:35.22 | v0lZy | the dect is supposed to be 192.168.1.3 |
| 08:35.30 | v0lZy | the ippbx is 192.168.1.2 |
| 08:36.53 | v0lZy | no idea whats going on here. |
| 08:37.12 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 08:37.51 | v0lZy | WIMPy: Im trying to ping 192.168.1.3 from 192.168.1.72 |
| 08:38.59 | Micc | I was hoping res_config_curl let me just put the extensions.conf or whatever.conf on another server. but apparently its just another way to access a database with a cgi. |
| 08:41.31 | v0lZy | ok, can ping it now |
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| 08:51.38 | *** join/#asterisk prasanthprabalan (~root@218.248.24.19) |
| 08:51.48 | prasanthprabalan | Hi all |
| 08:52.01 | v0lZy | But i cant access the GUi |
| 08:52.22 | prasanthprabalan | My asterisk setup is showing wrong time |
| 08:53.51 | prasanthprabalan | STRFTIME(${EPOCH},India,"%Y-%m-%d_%H:%M:%S") |
| 08:54.02 | prasanthprabalan | this command is showing wrong time |
| 08:54.13 | prasanthprabalan | Not matching with my system time |
| 08:54.34 | wdoekes2 | did you tzconfig after starting asterisk? |
| 08:55.07 | prasanthprabalan | no |
| 08:55.19 | prasanthprabalan | i havent |
| 08:56.32 | prasanthprabalan | My system time is 2.26 PM and asterisk is showing 8.56 AM |
| 08:57.10 | prasanthprabalan | If i am changing the system time to one hour asterisk time is also changing to exact one hour |
| 09:00.25 | wdoekes2 | I don't know if it parses India correctly. what happens if you put Europe/Amsterdam there? it should show 10:00 |
| 09:00.48 | zkn | what's the difference between Set(CALLERID
) and Set(CALLERPRES
.) ? |
| 09:01.22 | zkn | what exactly is CALLERPRES used for? |
| 09:01.47 | kaldemar | zkn: CALLERPRES only sets CLID presentation, CALLERID does that and other things aswell. CALLERPRES is deprecated. |
| 09:01.55 | WIMPy | zkn: Both have been deprecaded for a long time. Try 'core show function CALLERID'. |
| 09:01.59 | kaldemar | zkn: core show function CALLERPRES, core show function CALLERID |
| 09:02.13 | wdoekes2 | prasanthprabalan: (or Asia/Rangoon) |
| 09:02.34 | WIMPy | Does that still exist? |
| 09:02.51 | WIMPy | Oh, I didn;t think it did. |
| 09:03.55 | kaldemar | prasanthprabalan: India is probably not a valid zone in your system. |
| 09:04.40 | zkn | WIMPy: if both are deprecated, what is the proper new way to handle them ? |
| 09:04.48 | wdoekes2 | prasanthprabalan: Asia/Kolkata is probably the one you want |
| 09:05.07 | WIMPy | zkn: the CALLERID function. |
| 09:05.10 | kaldemar | STRFTIME will just silently return UTC if the zone is invalid. |
| 09:05.10 | prasanthprabalan | Yes thank u.. |
| 09:05.36 | prasanthprabalan | @wdoekes2: Asia/Kolkata is the correct one.. |
| 09:06.51 | zkn | WIMPy: OK, CALLERPRES seemed confusing, so I'll continue using CALLERID |
| 09:07.09 | zkn | the function, not application |
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| 11:23.11 | v0lZy | hi guys |
| 11:23.21 | v0lZy | kaldemar, WIMPy , ChrisInSydney |
| 11:23.30 | v0lZy | i have a question regarding security |
| 11:23.48 | WIMPy | Don't connect to the internet. |
| 11:23.50 | v0lZy | I've just been talking to my ISP, and we're talking completely different languages. |
| 11:24.11 | v0lZy | They usually speak in DMZ's and stuff |
| 11:24.21 | v0lZy | anyway... |
| 11:24.37 | ChrisInSydney | key |
| 11:24.41 | ChrisInSydney | hey |
| 11:24.56 | v0lZy | I have my pbx BEHIND my router. |
| 11:25.02 | ChrisInSydney | thats good |
| 11:25.07 | v0lZy | now, I dont know how askozia does it |
| 11:25.21 | v0lZy | but, it has a checkbox you tick to tell it that its behind NAT and it all works ok |
| 11:25.27 | ChrisInSydney | do you have 5060 forwarded ? |
| 11:25.54 | v0lZy | no, i dont |
| 11:26.04 | ChrisInSydney | excellent. Correct answer :-) |
| 11:26.14 | ChrisInSydney | There is 90% of your security |
| 11:26.17 | kaldemar | ~sipnat |
| 11:26.17 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 11:26.33 | v0lZy | My issue is |
| 11:26.36 | v0lZy | that i talked to my ISP |
| 11:26.46 | v0lZy | and they want to "secure" the connection |
| 11:26.52 | ChrisInSydney | ahhh |
| 11:27.00 | v0lZy | and we talked about it and i was generally pro their "connection security" |
| 11:27.07 | v0lZy | we were about to implement it now, by them giving me another IP |
| 11:27.14 | v0lZy | but i only have 1 fiber going into my router |
| 11:27.15 | kaldemar | v0lZy: what has your ISP to do with your VoIP connections? |
| 11:27.36 | ChrisInSydney | http://xkcd.com/806/ |
| 11:27.39 | v0lZy | so i thought.. .ok, its gonna be a VPN thing... and they go "no, we dont do VPN" ... so i go.. ok, so we can do a VLAN then and they go "we dont route that way" |
| 11:27.41 | kaldemar | v0lZy: are they also your ITSP? |
| 11:28.10 | v0lZy | and then they come up with an idea of me defining a VirtualIP and doing 1:1 NAT from a public IP to a private IP on which my PBX is located |
| 11:28.23 | v0lZy | and now im like... wtf do I need that? |
| 11:28.36 | v0lZy | yeah |
| 11:28.49 | v0lZy | they provide us with both a sip trunk and our internet conenction |
| 11:28.51 | ChrisInSydney | You shouldn't need to do this if you register your connection. If they are simply forwarding SIP to an IP, then..... |
| 11:29.05 | WIMPy | If you hadn;t done it yourself, I had asked that question. |
| 11:29.07 | ChrisInSydney | do you register => |
| 11:29.42 | v0lZy | well |
| 11:29.49 | v0lZy | they are supposed to give us a sip trunk |
| 11:29.55 | v0lZy | 10 parallel calls etc |
| 11:29.58 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 11:30.55 | v0lZy | so we've been running circles |
| 11:31.34 | ChrisInSydney | v0lZy: do you need to register or do they just forward SIP to an IP address ? |
| 11:32.35 | v0lZy | dont know |
| 11:32.39 | v0lZy | havent gotten to that stage |
| 11:32.42 | v0lZy | but i think register |
| 11:32.46 | v0lZy | username and password |
| 11:32.51 | v0lZy | at least thats how it was on the test account |
| 11:32.55 | v0lZy | but that wasnt a trunk. |
| 11:33.59 | kaldemar | it's basically the same whether someone calls it a trunk or not. |
| 11:34.06 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 11:34.37 | v0lZy | thats what i figure |
| 11:34.52 | v0lZy | I mean... geez, waht the hell am i doing in this company |
| 11:34.56 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 11:35.21 | v0lZy | its extra rare for me not to have to educate our providers techs... :D |
| 11:35.26 | v0lZy | and im not even proficient in this stuff |
| 11:35.45 | v0lZy | half-assed country with half-assed people working in half assed companies providing half assed services |
| 11:35.56 | v0lZy | thats like |
| 11:36.05 | v0lZy | 1/8 assed all together. |
| 11:36.22 | ChrisInSydney | Like Queen was 1/8 gay |
| 11:36.41 | v0lZy | ... slovenia, where if you want something done right, you have to do it yourself... and reclaim 7/8 ass. |
| 11:37.00 | ChrisInSydney | Brian, Roger and John were straight. Freddie was Bi, theregore they were 1/8 gay |
| 11:37.17 | WIMPy | If it makes you feel better: It's the same in Germany. |
| 11:37.34 | v0lZy | u sleep with men you're gay in my book, even if u bed a girl here and then :D |
| 11:37.44 | ChrisInSydney | Its probably the same here, but no one really cares, the weather is too nice |
| 11:39.22 | v0lZy | thing is, this time around |
| 11:39.25 | v0lZy | my ISP's pretty good |
| 11:39.34 | v0lZy | the guys know their stuff pretty well, service is ok |
| 11:39.54 | v0lZy | (thats why we switched over to them) |
| 11:40.09 | v0lZy | but its just internal confusion apparently.... some people working on VoIP, others on networking |
| 11:40.49 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 11:41.32 | ChrisInSydney | OK. Ask them if you can have a registration rather than a straight IP trunk, that way you can keep your port 5060 behind NAT |
| 11:41.53 | v0lZy | and the voip people are used to putting their voip stuff on 1 ip |
| 11:42.11 | ChrisInSydney | thats ok so long as you can register |
| 11:42.33 | WIMPy | Where's the relation there? |
| 11:42.35 | v0lZy | Do i understand this correctly: option 1, i login to the "trunk" with username/password |
| 11:42.47 | v0lZy | option 2, they just route everything to my IP |
| 11:42.56 | v0lZy | option 1, = i can be behind nat |
| 11:43.09 | WIMPy | yes |
| 11:43.13 | v0lZy | option 2, i have to bridge to the end of their pipe... |
| 11:43.27 | WIMPy | bridge? |
| 11:43.38 | v0lZy | ... well.. route |
| 11:43.39 | WIMPy | You may have to forward a port. |
| 11:44.15 | WIMPy | Possibly not when you enable qualify, but that depends on their setup. |
| 11:44.17 | v0lZy | I understand where they are coming from... they dont want a username/password thing alas someone could crack into it... but in that case, why not just restrict access to my IP!??! |
| 11:44.27 | WIMPy | Are calls all comming from the same ip/port? |
| 11:44.37 | v0lZy | Dont know |
| 11:44.58 | v0lZy | havent received info on their trunk yet... |
| 11:45.03 | ChrisInSydney | do you have a lot of DIDs ? |
| 11:45.15 | v0lZy | whats a DID? |
| 11:45.21 | ChrisInSydney | Direct In Dial |
| 11:45.26 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-zsgkqfcxkvqzhzxp) |
| 11:45.33 | ChrisInSydney | where you are given a block of numbers |
| 11:45.36 | v0lZy | uhm.. yeah |
| 11:45.41 | v0lZy | multiple DID numbers. |
| 11:45.46 | ChrisInSydney | how many ? |
| 11:45.55 | v0lZy | ah, they call that DDI blocks... |
| 11:46.06 | v0lZy | well, i have ... 60 numbers |
| 11:46.21 | v0lZy | as far as the provider is concerned, they should all be reachable from the outside. |
| 11:46.44 | v0lZy | in reality, i tend to block two or 3 |
| 11:46.57 | v0lZy | but yeah, 90% of them should be diretly dilable. |
| 11:47.01 | v0lZy | in. |
| 11:47.17 | ChrisInSydney | K. They may (think) that they have to create a separate logn for each number |
| 11:47.54 | ChrisInSydney | thats why they want to put you on an IP route |
| 11:48.46 | ChrisInSydney | so you would port forward to their SIP server and firewall 5060 to drop all packets except ones comming from that IP |
| 11:49.05 | ChrisInSydney | if you can do that on the router, you would |
| 11:49.10 | ChrisInSydney | easiest |
| 11:49.20 | ChrisInSydney | Otherwise IP tables |
| 11:49.43 | v0lZy | me port forward to their sip server and drop all incoming 5060 packets that are not coming from their server? |
| 11:49.54 | v0lZy | port forward to their sip server... as in outbound NAT? |
| 11:50.15 | ChrisInSydney | correct |
| 11:50.30 | ChrisInSydney | you will probably also need to do the same with RTP |
| 11:50.30 | v0lZy | or a static route? |
| 11:50.49 | ChrisInSydney | static route is only if you have a public IP subnet |
| 11:51.27 | v0lZy | hm |
| 11:51.30 | ChrisInSydney | you could run dd-wrt with Milkfish / OpenSER |
| 11:51.38 | v0lZy | well i have automatic outbound nat rules |
| 11:52.06 | v0lZy | im runing pfsense |
| 11:52.11 | ChrisInSydney | cool |
| 11:52.16 | v0lZy | but the way i see it |
| 11:52.21 | ChrisInSydney | on another box |
| 11:52.31 | ChrisInSydney | ? |
| 11:53.02 | v0lZy | .... me doing port forwarding to them ... id go on my LAN interface, block all outgoing traffic except traffic directed at their public ip |
| 11:53.28 | v0lZy | and open up 5060 for their server on my WAN interface and NAT it to my PBX. |
| 11:53.30 | ChrisInSydney | outgoing is OK, incomming is more important |
| 11:53.40 | ChrisInSydney | incoming |
| 11:54.12 | v0lZy | have to run |
| 11:54.15 | v0lZy | thanks for the help |
| 11:54.18 | ChrisInSydney | cool |
| 11:54.21 | ChrisInSydney | good luck |
| 11:54.21 | v0lZy | ill be back in 30 or so |
| 11:54.36 | v0lZy | have to go home, get ready for the doctors appointment then go to the doctor at 15:00 (its 13.00 now) |
| 11:54.41 | ChrisInSydney | I'll probably be in bed. 11:00pm here |
| 11:54.42 | v0lZy | thanks |
| 11:55.05 | v0lZy | take care, talk soon1 |
| 11:55.07 | v0lZy | bye |
| 11:55.10 | v0lZy | thanks again |
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| 12:48.10 | v0lZy | back |
| 12:48.27 | v0lZy | ChrisInSydney, WIMPy: the trunk is without registration |
| 12:49.48 | ChrisInSydney | port forward your SIP and RTP ports. Limit SIP inbound to your ITSP |
| 12:52.51 | v0lZy | what are the sip and rtp ports? |
| 12:52.54 | v0lZy | what number? |
| 12:53.07 | ChrisInSydney | have a look at sip.conf and rtp.conf |
| 12:53.12 | *** join/#asterisk topro (~quassel@host-62-245-142-50.customer.m-online.net) |
| 12:53.24 | ChrisInSydney | 5060 for SIP. |
| 12:53.39 | ChrisInSydney | RTP will be a range |
| 12:54.16 | v0lZy | ah, excellent |
| 12:55.34 | WIMPy | Or if your router is smart enough, you can just forget about it. |
| 12:56.11 | ChrisInSydney | pfsense |
| 12:56.43 | WIMPy | That's just a frontend, isn't it? |
| 12:57.10 | WIMPy | Distro, ok. |
| 12:57.50 | ChrisInSydney | not too sure. I think ita a package based on BSD, or is that m0n0wall |
| 12:58.05 | v0lZy | pfsense is based on monowall. |
| 12:58.15 | WIMPy | pfsense is a fork of m0n0wall. |
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| 13:15.27 | leifmadsen | I never port forward |
| 13:15.41 | leifmadsen | WIMPy: kinda... but pfsense has more features. I had to change from m0n0wall to pfsense locally |
| 13:15.59 | ollii | openbsd + pf.conf + pfctl |
| 13:16.03 | leifmadsen | actually reads the full scrollback now ;) |
| 13:16.03 | ollii | <3 |
| 13:16.17 | leifmadsen | managing your own base OS == </3 |
| 13:16.41 | WIMPy | leifmadsen: That's just what I read on their homepage. |
| 13:16.46 | leifmadsen | WIMPy: :) |
| 13:17.10 | leifmadsen | I do like pfsense better than m0n0wall -- it may not be a requirement in your network, but I needed upnp functionality |
| 13:17.46 | WIMPy | I haven't used anythign BSD for ages. |
| 13:18.00 | WIMPy | 11 years, I guess. |
| 13:18.04 | v0lZy | pfsense is amazing |
| 13:19.09 | v0lZy | i havent yet came accross anything like pfsense |
| 13:19.23 | v0lZy | it has a ton of features |
| 13:19.30 | v0lZy | and its fairly easy to use. |
| 13:19.41 | v0lZy | and what it does it does well |
| 13:19.49 | v0lZy | stable, dependable |
| 13:20.03 | WIMPy | Ok, if you don't have to forward rtp ports, it might be ok :-) |
| 13:20.19 | leifmadsen | works good locally on my soekris net4801 |
| 13:22.30 | v0lZy | rtp are udp right? |
| 13:22.39 | WIMPy | yes |
| 13:22.46 | WIMPy | always |
| 13:23.58 | v0lZy | dont see why there'd be a problem with rtp then |
| 13:24.19 | WIMPy | What kind of problem? |
| 13:24.29 | v0lZy | WIMPyOk, if you don't have to forward rtp ports, it might be ok |
| 13:24.40 | v0lZy | was refering to that, i dont know what eaxactly you ment |
| 13:24.51 | WIMPy | If it can find out by itself. |
| 13:25.04 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:25.07 | WIMPy | I don't have any ports opened for rtp. |
| 13:25.29 | v0lZy | i'm still not on the bullseye here |
| 13:25.38 | v0lZy | u mean the issue with NAT and sip headers? |
| 13:25.58 | WIMPy | Not important if you only talk to your IPSP, but if you have peers in changing locations, that's a very good thing (to me at least). |
| 13:26.29 | v0lZy | vpn? |
| 13:26.46 | WIMPy | nope |
| 13:26.59 | v0lZy | my ippbx has an option |
| 13:27.05 | v0lZy | to input public ip address |
| 13:27.06 | WIMPy | The magic of connection tracking. |
| 13:27.19 | v0lZy | and it rewrites all the sip headers.. so no issue with pfsense in that case |
| 13:27.21 | v0lZy | ah... |
| 13:27.28 | v0lZy | pfsense has something like 'sticky connections' |
| 13:28.05 | WIMPy | That doesn't sound like opening ports on the fly. |
| 13:28.08 | v0lZy | and aoutomatic outbound nat etc |
| 13:28.24 | v0lZy | but opening ports... like port knocking? |
| 13:28.48 | WIMPy | No, like sniffing SIP traffic and finding out what's negotiated there. |
| 13:29.03 | v0lZy | ah, no, it doesnt do that as far as i know |
| 13:29.05 | v0lZy | but |
| 13:29.18 | v0lZy | it has an option to download packages |
| 13:29.29 | v0lZy | so im sure it can be integrated if noone thought of it yet |
| 13:30.10 | WIMPy | Not having used it for ages, I have no idea, what BSD offers in that area. |
| 13:30.29 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:4dca:32db:2926:5b83) |
| 13:32.45 | v0lZy | i have to get ready for my doctors appointment |
| 13:33.41 | v0lZy | brb |
| 13:34.35 | WIMPy | Take care. These guys can be dangerous. |
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| 13:48.49 | v0lZy | im not going yet |
| 13:48.51 | v0lZy | just getting ready |
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| 13:52.04 | WIMPy | wonders ifa broken pattern match in Loopback is the reason why it has become almos impossible to interctively dial a number. |
| 13:55.35 | jkroon | hi guys, how can I include a literal $ in a string? |
| 13:55.52 | WIMPy | Have you tried \$ ? |
| 13:55.58 | jkroon | WIMPy, yes |
| 13:56.12 | jkroon | eg, Gosub(foo,1,1(this$is$a${}literal$string)) |
| 13:56.12 | WIMPy | BTW: I updated the cabling thing. |
| 13:56.35 | kaldemar | jkroon: which version of asterisk? |
| 13:56.41 | jkroon | 1.8.8.2 |
| 13:57.44 | jkroon | \${foo} results in \bar (assuming foo=bar), whereas I want the literal string ${foo} |
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| 13:59.51 | anny__ | how can we add theora (video codec) support to asterisk 1.6 or 1.4 |
| 14:00.42 | pigpen | not that this answers your question, but that is to be the beauty of 10.x |
| 14:01.45 | pigpen | I have been spending quite a bit of time with 10.1.x lately, mainly for the faxing and t.38 support |
| 14:01.50 | jkroon | kaldemar, ideas/suggestions? |
| 14:02.46 | pigpen | jkroon, you may want to post in paste bin exactly what you are doing?this way, if the literal is not possible, there may be a work around. |
| 14:02.51 | kaldemar | jkroon: adding \'s used to help on some versions. strings may behave differently in different apps/places in dialplan. |
| 14:03.15 | kaldemar | jkroon: are you using it in a Gosub? |
| 14:03.30 | [TK]D-Fender | kaldemar, He's showing as part of an actual priority label |
| 14:03.49 | [TK]D-Fender | kaldemar, Not a variable target to jump to, but the label itself |
| 14:03.50 | kaldemar | [TK]D-Fender: just making sure that's where it's really used. |
| 14:03.51 | WIMPy | BTW: How do you write control chars in the dialplan? Die \x11 or \x0111 work? |
| 14:03.53 | jkroon | kaldemar, Gosub(callback-setup,s,1(uls-testqueue,\$[\${QUEUE_MEMBER(uls-testqueue,ready)}>0])) |
| 14:04.33 | jkroon | basically I EVAL() that second parameter at a later stage |
| 14:04.46 | jkroon | but it goes out to a .call file first. |
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| 14:06.17 | kaldemar | jkroon: one way would be to decode/encode the argument. |
| 14:06.33 | jkroon | kaldemar, that is a very good idea! thanks. |
| 14:07.17 | kaldemar | jkroon: func BASE64* |
| 14:09.02 | WIMPy | While searchin for somethign else, I just found a reference to $\{var\} to escape a variable. |
| 14:09.27 | jkroon | kaldemar, actually URIDECODE will be simpler (still readable) |
| 14:09.37 | jkroon | just replace $ with %24 |
| 14:09.47 | pigpen | kaldemar, hmm?I have never used that?I got to google that... |
| 14:10.12 | jkroon | kaldemar, brilliant idea thanks! |
| 14:12.44 | pigpen | so by using the BASE64 function, it encodes the string and does not "enumerate" it (such as a variable or such) in the dial plan? |
| 14:13.51 | WIMPy | So is there a way to write control characters? Or would that require something like URIDECODE? |
| 14:14.24 | WIMPy | And is there an answe outside the source? |
| 14:14.52 | kaldemar | this is awesome: exten => foo,1,Gosub(sub,s,1(\$\{CHANNEL\})) and in [sub] exten => s,1,Noop(${REPLACE(ARG1,\\\\)}) |
| 14:15.08 | kaldemar | 4 \'s makes a single literal "\". |
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| 14:20.00 | Lantizia | Lo, does anyone know of a nice passive billing system (i.e. by passive I mean doesn't need any changes to dialplan, looks at what to bill via AMI maybe?)... a2billing is causing all kinds of headaches where call forwards and such no longer works |
| 14:20.33 | Lantizia | something that uses the "account code" setting of the peers would be handy |
| 14:21.10 | kaldemar | Lantizia: making your own parser for CDR is quite simple. |
| 14:21.12 | WIMPy | oocalc :-) |
| 14:21.45 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 14:21.46 | Lantizia | kaldemar, nah I need something the non-linuxy engineers can use with a nice webgui like a2billing has - something that can make phone bills in pdf too |
| 14:22.00 | Lantizia | the only one I've found that might do this is asterBilling |
| 14:22.34 | kaldemar | Lantizia: exactly. making a parser and a web page for output is not that complex a task. |
| 14:23.32 | Lantizia | so your answer is ... no i don't know of one |
| 14:24.21 | kaldemar | Lantizia: never seen a freely distributed simple solution like that, yes. |
| 14:24.33 | kaldemar | Lantizia: don't expect to find one either. |
| 14:29.50 | pigpen | Lantizia, I agree with kaldemar. I knew zero php?and I wrote an interface that does what I need it to do for billing. |
| 14:29.57 | pigpen | it is not terribly hard. |
| 14:30.28 | pigpen | so this would be a pro-php'y thing. |
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| 14:38.42 | *** join/#asterisk bodagetta (~Michael@173-24-237-126.client.mchsi.com) |
| 14:39.07 | bodagetta | hello |
| 14:41.43 | WIMPy | Holy shit! |
| 14:41.55 | WIMPy | Are switches used on every priority? |
| 14:43.03 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 14:43.41 | WIMPy | Not that this explains my initial issue, but the dundi queries continue after the dialplan has been hit. |
| 14:43.53 | WIMPy | That doesn't seem to make sense. |
| 14:44.00 | WIMPy | Is that a bug or a feature? |
| 14:44.15 | bodagetta | I'm having some weird registration problems. I have three Polycom 331's in a remote location behind NAT. They were all registered fine until an internet outage.. Now, two will register but the third will not. Then the third phone will register and one of the previously two that were registered will drop off |
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| 14:45.13 | pigpen | bodagetta, I take it your sip server is on the other side of the NAT? |
| 14:46.07 | bodagetta | My SIP server is in the DMZ of my router |
| 14:46.20 | bodagetta | (I' m aware of the security issues - just trying to get it working for now) |
| 14:47.15 | bodagetta | I have two other Polycom 321's at a different location behind the same network configuration and they are registered fine. So two remote locations, one with 3, 331's. One with 2, 321's |
| 14:47.28 | pigpen | I take it that the sip server is also a production web server, hence the DMZ, or was the port free and you thought it would be cool to use it? |
| 14:48.04 | bodagetta | The SIP server is PIAF purple running on an old PC |
| 14:49.09 | pigpen | You may want to direct this issue to PIAF. |
| 14:49.13 | *** join/#asterisk jonlewisorg (~jlewis@solo.atlantic.net) |
| 14:49.31 | pigpen | this channel is largely asterisk only, native. not a package install. |
| 14:49.47 | pigpen | in short though, I think your source udp ports is your issue. |
| 14:50.13 | pigpen | space them out: 5060, 5061, 5062 on each phone. |
| 14:50.27 | pigpen | NAT in general is not a good thing when it comes to VOIP. |
| 14:50.34 | jonlewisorg | don't suppose there's |
| 14:50.45 | jonlewisorg | <PROTECTED> |
| 14:50.56 | pigpen | if the asterisk box just an asterisk box, get it on the net, don't stick it on a dmz. that is just a waste of time. |
| 14:51.20 | bodagetta | ok, so if I set the source udp to 5061 on a phone it will still connect to the asterisk box correctly? |
| 14:51.26 | jonlewisorg | I've got a number of servers being hit by what I assume is a new sip scanner, taking advantage of http://downloads.asterisk.org/pub/security/AST-2008-003.html and the suggested workaround does not appear to work |
| 14:51.45 | pigpen | bodagetta, sure, you are setting the source port, not the destination. |
| 14:51.45 | bodagetta | currently all the phones are set to 5060 |
| 14:51.49 | pigpen | right. |
| 14:52.18 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 14:52.19 | pigpen | it may or may not work. Your firewall may also not support this type of thing well. |
| 14:52.21 | bodagetta | so the destination is actually specified in the message.. now i feel stupid |
| 14:52.32 | pigpen | but once again, it is silly to have an asterisk only box on a dmz. |
| 14:52.40 | *** join/#asterisk DynamicFail (~Jinxed---@147.177.57.6) |
| 14:52.47 | pigpen | why on a DMZ? |
| 14:52.48 | DynamicFail | Can asterisk register sccp phones? |
| 14:52.49 | bodagetta | i have other computers behind the router |
| 14:52.56 | pigpen | and? |
| 14:53.33 | pigpen | I prefer a dedicated network for my phones, however, there are many instances where that is not possible (cabling and such). |
| 14:53.46 | pigpen | stick it on your main network and have less issues. |
| 14:53.54 | bodagetta | i figured my options were either set port forwarding on the router to the asterisk box, or put it in a DMZ |
| 14:54.15 | pigpen | If you said, " I have 4000 phones?.." that would be different. |
| 14:54.27 | pigpen | port forwarding? If you have it on the same network? |
| 14:55.06 | pigpen | You are making this much more difficult than it needs to be. |
| 14:55.12 | bodagetta | no, i have a cable modem that connects to a router, and then the router is connected to other computers as well as the asterisk box |
| 14:55.16 | pigpen | Also, ditch PIAF and use the real deal. |
| 14:55.48 | bodagetta | thanks for your help, i'll try changing the source udp ports |
| 14:55.55 | pigpen | Ok, do this. Get your phones on the sam net?you will be happier. |
| 14:56.25 | bodagetta | Oh I wish I could.. I need to set up site to site VPN for that? |
| 14:57.07 | pigpen | Do this: draw your network. Stick it on a paste bin somewhere. I have yet to see a need for a vpn. |
| 14:57.24 | [TK]D-Fender | jonlewisorg, That is a 4 year old advisory and you haven't told us what you're running or shown any backup.... |
| 14:58.24 | bodagetta | i physically have an office in Atlanta and an office in Huntsville with cable modem connections in each for internet access |
| 14:58.54 | pigpen | ?and you want them to talk to each other, LD free. |
| 14:59.04 | pigpen | What firewalls are you using? |
| 14:59.14 | pigpen | And do you have static IP's? |
| 14:59.22 | bodagetta | i want 1 asterisk server |
| 14:59.37 | bodagetta | no firewall other than the router, no static IP but I could get that |
| 15:00.08 | pigpen | Huntsville Georgia or Texas? |
| 15:00.17 | bodagetta | Alabama, home of digium |
| 15:00.31 | pigpen | ah, yeah, forgot. |
| 15:01.17 | bodagetta | i went to college with Mark Spencer, he was computer engineering and I was electrical hence my lack of network intelligence.. I should take him to lunch and pick his brain |
| 15:01.30 | jonlewisorg | [TK]D-Fender: I know its an old advisory, and that the 1.0 code is "dead"...but we unfortunately have some systems running 1.0.11.1 that never upgraded because we had unresolved audio issues with 1.2.x when we tried upgrading long ago |
| 15:01.53 | [TK]D-Fender | jonlewisorg, 1.0 isn't just dead.... it was cremated. |
| 15:02.03 | pigpen | bodagetta, sounds like a good idea. But you need real firewalls, and ditch the router. |
| 15:02.08 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:02.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:02.25 | pigpen | bodagetta, vpn it up, stick an Asterisk box at one location, or host it somewhere. |
| 15:02.41 | pigpen | All traffic to and from the pbx and phones should have NO NAT! |
| 15:02.42 | [TK]D-Fender | jonlewisorg, time to upgrade. (actually, that time was at least 4 years ago) |
| 15:02.52 | bodagetta | any recommendations on a firewall? |
| 15:02.54 | jonlewisorg | but the issue is, the workaround of setting sip.conf's context in [general] to a non-existent one or to one that issues Hangup on all calls, isn't working |
| 15:03.20 | [TK]D-Fender | jonlewisorg, pastebin the sip debug and your configs so we can at least see it... |
| 15:03.21 | [TK]D-Fender | ~pb |
| 15:03.21 | 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.us , 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. |
| 15:03.23 | [TK]D-Fender | ^^^ |
| 15:03.28 | jonlewisorg | looking at the fix for this in the 1.2.26->1.2.27 patch, it was a one liner |
| 15:03.31 | pigpen | bodagetta, I am bias. We use the product we developed that runs on linux. FWIPSEC. |
| 15:03.52 | pigpen | bodagetta, if you want more info, pm me. |
| 15:04.01 | [TK]D-Fender | jonlewisorg, Those versions are all ancient and have more security holes than a #9 spnoge |
| 15:04.05 | [TK]D-Fender | sponge* |
| 15:04.07 | jonlewisorg | so I'm guessing I can fix this in 1.0.11.1 and put off the pain a few more years |
| 15:04.40 | WIMPy | And using 1.0 is no pain??? |
| 15:04.43 | jonlewisorg | there are other sip or iax holes in 1.0.x? |
| 15:04.44 | pigpen | bodagetta, otherwise, good luck. We went to this for a reason. Oh, we also do tons of VPN's & VOIP. |
| 15:05.20 | pigpen | misses asterisk 1.0?.no T.38 to clutter my mind... |
| 15:05.28 | jonlewisorg | WIMPy: these servers are obviously old...were setup long ago, and work fine...as long as the script kiddies don't hit them with sip scanners trying to call jamaica and st. lucia |
| 15:05.30 | bodagetta | thanks for the info, I will check it out |
| 15:05.45 | bodagetta | going to try changing the source ports |
| 15:05.50 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:05.58 | jonlewisorg | if not for roaming users with soft phones, I'd just firewall off sip access...but I can't do that |
| 15:06.20 | WIMPy | jonlewisorg: I don't know 1.0. I only started with 1.4, but I know that I wouldn't want to use that any more. |
| 15:07.09 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 15:07.24 | pigpen | Anybody know if 10.x has issues with fax detect? |
| 15:08.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:08.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:10.40 | jonlewisorg | yeah...this fix looks Trivial |
| 15:11.23 | jonlewisorg | just remove from check_user_full these two lines if (ast_strlen_zero(of)) return 0; |
| 15:13.48 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 15:19.25 | DynamicFail | [TK]D-Fender, What is asterisk's ability to work with sccp/cisco equipment |
| 15:25.25 | *** join/#asterisk syadnom (~syadnom@216.187.136.98) |
| 15:26.09 | syadnom | hi all. having a strange issue with my system. I have a sangoma B600DE card in and I am getting about a 3 second delay between inbound call and the system ringing. |
| 15:26.35 | *** join/#asterisk sustav (~alfa@nat/digium/x-xgkpinjfvevmzzsi) |
| 15:26.38 | ollii | #sangoma |
| 15:26.49 | syadnom | oh, didnt realize there was a chan |
| 15:26.59 | ollii | good support in there :) |
| 15:27.02 | syadnom | thanks |
| 15:28.01 | [TK]D-Fender | DynamicFail, I don't use SCCP or Cisco at all... |
| 15:29.40 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:31.45 | DynamicFail | Does anyone know what legal options asterisk is able to with the regards of being able to set up a trunk between a cisco call manager and asterisk system and the ability for asterisk to register sccp phones directlyl |
| 15:33.32 | jonlewisorg | DynamicFail: that's 2 questions...I know you can do sip trunking between cisco cm and asterisk 1.0...as I've done it with a customer's cm |
| 15:35.40 | [TK]D-Fender | DynamicFail, * only speaks SCCP to PHONES and has no legal issues. your PHONES are another matter, and that's up to you and Cisco. |
| 15:35.55 | [TK]D-Fender | DynamicFail, * does not talk SCCP to CCM, you'll need to use something else. |
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| 15:57.28 | d_preston215 | What is the most common cause of MixMonitor recording sync issues? |
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| 16:41.16 | DynamicFail | [TK]D-Fender, but with asterisk 1.6 by deafult I should be able to register sccp phones ? |
| 16:41.38 | [TK]D-Fender | * 1.6 supports SCCP phones |
| 16:42.24 | DynamicFail | nice!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 16:42.34 | DynamicFail | one key jamed there |
| 16:45.38 | beek | I thought that perhaps exclamation points were on sale. |
| 16:49.57 | jaytee | never saw exclamation points on sale, just semi-colons. A while back colons were on sale but they had polyps. |
| 16:51.46 | *** join/#asterisk Eren (~eren@unaffiliated/eren) |
| 16:51.47 | [TK]D-Fender | Don't worry.. they grow on you... |
| 16:52.07 | Eren | anyone here that has an experience with amateur radio? |
| 16:52.35 | Eren | I'm trying to build a inter-linked emergency-communication infrastructure using internet and analog radios |
| 16:54.00 | Eren | I initially used svxlink, however, there is a single point of failure. I want the system to be autonomous, meaning that all the remote nodes should act as a repeater, and when other nodes connect to them, it should send the voice to connected nodes, and accept the voice from these nodes as well |
| 16:55.02 | WIMPy | Are you talkin phone calls or some sort of broadcast? |
| 16:56.00 | Eren | WIMPy: some sort of broadcast, I'm not interested in phone calls actually right now |
| 16:56.29 | Eren | think locally, when I talk with my radio, the listening node should repeat it (half-duplex) |
| 16:57.01 | Eren | additionally, other nodes should be able to connect to that node, and my voice should be gone to connected nodes |
| 16:57.15 | WIMPy | How do you make a flood fill type ting work via an analog link? |
| 16:57.35 | Eren | and as you may guess, these nodes should repeat them, so that all the radios could be connected together |
| 16:57.43 | Eren | WIMPy: hm? |
| 16:58.44 | WIMPy | If you send the message and several nodes receive them and then all of those repeat the message, you'll have some "collision", I guess. |
| 16:59.50 | WIMPy | And then if the next ones repeat the message, the ones having repeated it before would receive it again, wouldn;t they? |
| 17:00.04 | Eren | no |
| 17:00.21 | Eren | think that we have a different cities away from them named A, B, C |
| 17:00.35 | Eren | we set up a repeater system on the hills of A, B, and C |
| 17:00.54 | Eren | when I push a button on my radio in A, it will repeat |
| 17:01.00 | WIMPy | ptp then? |
| 17:01.25 | Eren | additionally, if B is connected to A, the hamradio operators in city B will also hear me |
| 17:02.07 | Eren | is it possible? I've looked at app_rpt, but as I don't have experience with asterisk, I wanted to ask :) |
| 17:02.33 | leifmadsen | Eren: no one here is going to have experience with app_rpt -- what you're talking about is a very specific situation |
| 17:02.37 | WIMPy | I see a basic network design issue first. |
| 17:02.41 | leifmadsen | I don't know anyone other than the other who actually uses it |
| 17:02.54 | leifmadsen | so you might want to find the community that actively deploys app_rpt for answers in that realm |
| 17:03.05 | Eren | leifmadsen: yeah right |
| 17:03.38 | Eren | WIMPy: not like, it's kind of a amateur-radio specific, I couldn't express my situation clearly |
| 17:04.05 | WIMPy | Yes, but radio is usually broadcast. |
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| 17:42.36 | *** join/#asterisk nitram (nitram@korben.fhloston.org) |
| 17:43.31 | nitram | hi, just a quick question. how do i aggregate multiple extensions in one ael line? 123|124 => { NoOP();} does not work. |
| 17:46.20 | pigpen | Since I know we all enjoy faxing so much, I thought I would share my latest issue. |
| 17:46.47 | pigpen | asterisk 10.1.0, res_fax_digium, analog line (verizon), via Digium 4 port card, fxo. |
| 17:46.58 | pigpen | See: http://pastebin.com/2aa428QB |
| 17:47.27 | pigpen | I can pass a page or two inbound, from two sources (one is asterisk box with iaxmodem/hylafax, other is efax) |
| 17:47.45 | [TK]D-Fender | nitram, You don't |
| 17:48.10 | [TK]D-Fender | nitram, If a pattern is application you can use that instead. That specific example is doabl |
| 17:48.12 | [TK]D-Fender | e |
| 17:48.31 | pigpen | with little to no issue. However, if I send a fax inbound, it hits 5 - 9 pages and gets a T1_TIMEOUT, T2_TIMEOUT or LOSS_OF_HS_SYNC error. |
| 17:48.52 | pigpen | however in the noted paste bin, to me, it seems that it is ending clean. |
| 17:49.02 | pigpen | however I am new to the output, hence the post. |
| 17:49.03 | nitram | [TK]D-Fender: it is a list of random extensions without pattern |
| 17:49.20 | nitram | damn |
| 17:50.14 | [TK]D-Fender | nitram, Goto() <- |
| 17:50.41 | [TK]D-Fender | or Gosub, or Macro, or .... |
| 17:51.06 | pigpen | Naturally, just as I post, I find out I need to leave. Any ideas, pls post. Tks. |
| 17:55.21 | *** join/#asterisk ke-esc (~ke-esc@155.229.209.170) |
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| 17:56.45 | ke-esc | Good afternoon all.. trying to set up realtime, for now just with sippeers.. I've got the database created, asterisk can connect to the database, etc., etc., but my softphone cannot loggon. I added everything from sip.conf for the user entry into the database table, but not dice... any non-obvious things i might be missing? |
| 17:59.19 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 17:59.56 | [TK]D-Fender | ke-esc, show us the configs. |
| 17:59.57 | [TK]D-Fender | ~pb |
| 17:59.57 | 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.us , 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. |
| 17:59.58 | [TK]D-Fender | ^^^ |
| 18:04.10 | ke-esc | [TK]D-Fender, not sure exactly what you'll need- I've added enough to show the realtime config, and the values being read from the db.. http://pastecode.com/p2 |
| 18:05.45 | [TK]D-Fender | ks3, res_config <- |
| 18:08.29 | ke-esc | [TK]D-Fender, don't have a res_config..just the res_odbc |
| 18:08.39 | [TK]D-Fender | You need it |
| 18:08.46 | [TK]D-Fender | Go read the sample configs. |
| 18:10.08 | ke-esc | Even when using ODBC? the only res_configs in my /usr/src/asterisk/configs are res_config_mysql.conf.sample and res_config_sqlite.conf.sample |
| 18:11.07 | [TK]D-Fender | extconfig rather.... |
| 18:11.07 | leifmadsen | ke-esc: when using odbc those files are not used at all |
| 18:11.16 | leifmadsen | extconfig.conf, res_odbc.conf |
| 18:11.37 | leifmadsen | there is not res_config file |
| 18:11.40 | leifmadsen | s/not/not |
| 18:11.42 | leifmadsen | bah |
| 18:11.44 | leifmadsen | s/not/no |
| 18:12.09 | *** join/#asterisk xertres (~xertres@h-136-1-238.a176.priv.bahnhof.se) |
| 18:12.10 | ke-esc | ahh okay, extconfig was in my pastebin... just two lines right now : [settings] \n sippeers=>odbc,asterisk,sippeers |
| 18:12.40 | leifmadsen | ok, so sippeers are configured to be loaded from ODBC using the sippeers table |
| 18:13.16 | leifmadsen | so you need to configure res_odbc.conf |
| 18:13.25 | [TK]D-Fender | leifmadsen, http://pastecode.com/p2 |
| 18:13.26 | leifmadsen | which in turn needs the /etc/odbc.ini and /etc/odbcinst.ini files configured |
| 18:13.29 | [TK]D-Fender | leifmadsen, Seems to have.. |
| 18:13.36 | leifmadsen | [TK]D-Fender: I don't click on links |
| 18:13.42 | [TK]D-Fender | ke-esc, Go PB the SIP debug of a reg attempt in the meantime. |
| 18:13.57 | [TK]D-Fender | leifmadsen, ok, hand-retype it then :p |
| 18:14.15 | leifmadsen | if you don't have rtcachefriends=yes in sip.conf you also won't see the registrations loaded into memory |
| 18:16.46 | ke-esc | [TK]D-Fender, added to pastebin |
| 18:16.53 | ke-esc | leifmadsen, i'll add that and try |
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| 18:29.56 | ke-esc | Bah, this is frustrating... One would think that so long as it can read the same values that were in sip.conf it would work just fine, but apparently one would be wrong |
| 18:32.38 | *** join/#asterisk enoch (~arturo@unaffiliated/enoch) |
| 18:32.41 | enoch | hi all |
| 18:32.46 | enoch | can asterisk handle faxes? |
| 18:32.47 | enoch | how? |
| 18:34.06 | leifmadsen | very carefully |
| 18:34.26 | *** join/#asterisk rhamnett (~rick@5e095f41.bb.sky.com) |
| 18:34.48 | enoch | but how it works? a software? an hardware fax? |
| 18:35.44 | leifmadsen | using T.38 |
| 18:35.50 | [TK]D-Fender | Or not. |
| 18:39.18 | rrittgarn | Anybody here have experience with provisioning Cisco SPA 500 series phones? |
| 18:48.33 | ke-esc | Got it! I had thought since one of the fields was dynamic (yes/no), that I should set that to yes and didn't set host=dynamic |
| 18:50.59 | *** join/#asterisk libryder (~Adium@209.33.214.243) |
| 18:51.17 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 18:51.20 | libryder | is it expected behavior to receive dtmf tones unreliably over VOIP> |
| 18:51.50 | libryder | there is latency on dtmf tones and sometimes the tone doesn't even make it through |
| 19:00.53 | ke-esc | So I noticed in the asterisk docs that they suggest using names for SIP peers instead of just extension numbers.. The trouble with that though is I can't just do something like 'exten => _1XX,1,Dial(SIP/${EXTEN})'.... what are others doing so they don't have to put individual rules in for each extension |
| 19:01.21 | *** join/#asterisk imcdona (imcdona@2001:470:e92e:1:c83a:d8a3:2804:8faf) |
| 19:02.11 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 19:02.12 | [TK]D-Fender | ke-esc, Do do separate extens for each. That is the way it's supposed to be |
| 19:03.36 | p3nguin | ~devices |
| 19:03.37 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 19:03.43 | p3nguin | ke-esc: See above. |
| 19:03.48 | libryder | ~dtmf |
| 19:03.48 | infobot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
| 19:04.05 | *** part/#asterisk libryder (~Adium@209.33.214.243) |
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| 19:06.26 | ke-esc | What is the rationale for that? I can understand abstracting people/extensions from devices, but typically a user logs in with an extension and password.. why abstract the person from the extension? |
| 19:06.27 | p3nguin | exten => 762,1,Dial(SIP/000011112222,36); this is extension 762, which dials a SIP device by the name of 000011112222. |
| 19:07.33 | p3nguin | The people are real people. |
| 19:07.57 | p3nguin | Step 1) Abstract an extension number for a person. |
| 19:08.15 | p3nguin | Step 2) Configure any arbitrary device for use on the system. |
| 19:08.18 | [TK]D-Fender | ke-esc, He's commenting on your mixed up usage of "extension" vs "device" |
| 19:08.30 | *** join/#asterisk afink (~afink@207.106.66.194) |
| 19:08.57 | p3nguin | Step 3) Associate a person's extension number with a device in the dial plan. |
| 19:10.03 | [TK]D-Fender | ke-esc, and no, they log in with a USERNAME and PASSWORD. |
| 19:10.15 | p3nguin | A "user" which is a person, typically does not have to log in. |
| 19:10.31 | p3nguin | Except in cases where they do. |
| 19:10.53 | imcdona | Anyone having issues with Bria and Asterisk? In particular the SIP dialog that Bria is having with Asterisk is leaving thousands of " Rx: PUBLISH" type sip channels open. Eventually Asterisk stops accepting sIP connections all together and has to be restarted. |
| 19:10.56 | [TK]D-Fender | ke-esc, Any association to a dialplan extension is purely coincidental . |
| 19:11.05 | p3nguin | Phones do the authentication with asterisk. The "user" (which is a person) only uses the phone. |
| 19:11.53 | ke-esc | ok ok, i see it =) |
| 19:14.50 | p3nguin | On the back of most hard phones is a tag displaying the MAC address of the phone. That is the perfect "username" for the device, which is used to configure the peer in sip.conf. That "username" is unique to that phone and isn't likely to ever be duplicated, lost, or forgotten. |
| 19:15.44 | p3nguin | Passwords should be fairly randomized -- it makes it harder to guess if someone can guess the MAC address "username." |
| 19:16.11 | p3nguin | I like to use apg to create long, totally random passwords for my devices. |
| 19:17.08 | ke-esc | okay.. so "devices" are configured in sip.conf, but within the section for each device is specified the callerid and context (which defines what the device has access too... local/ld/intl etc), shouldn't these be specified elsewhere then? |
| 19:18.09 | p3nguin | The context parameter is that of the context in extensions.conf where calls from the device will be sent. |
| 19:18.24 | p3nguin | I prefer to use a context of 'phones' for all of my phones. It just makes sense. |
| 19:19.13 | *** join/#asterisk dxd828 (~dxd828@88-109-118-127.dynamic.dsl.as9105.com) |
| 19:19.37 | ke-esc | How do you specifiy who is or is not allowed to make international calls for example? |
| 19:20.10 | p3nguin | Within the 'phones' context in extensions.conf I include all other contexts that phones should be able to access, such as 'internal' and 'outgoing-calls'. |
| 19:21.34 | ke-esc | Right, but say real person "Bob" should be able to make intl calls, but real user "Joe" shouldn't... I don't care what phone has access to what, I care when user Bob sits at desk 5 and keys in his credentials, he can access what he needs. |
| 19:22.27 | ke-esc | Right now I do that by specifying different contexts for different devices, but this breaks the abstraction |
| 19:23.26 | ke-esc | ( not trying to be difficult, just trying to wrap my head around this ideal setup_) |
| 19:24.03 | [TK]D-Fender | ke-esc, Contexts are the abstraction you're supposed to do |
| 19:24.04 | p3nguin | Well, I was describing static phones and users rather than hot-desking... |
| 19:24.24 | p3nguin | But it can still be worked out. |
| 19:24.35 | [TK]D-Fender | ke-esc, And there are several ways you can use contexts with hot-desking devices. |
| 19:26.12 | ke-esc | Okay, one last q before I put this down for the day :) What is the regexten parameter in sip.conf for? It seems now that it shouldn't be there, or that I'm just using it inappropriately |
| 19:27.32 | *** part/#asterisk dxd828 (~dxd828@88-109-118-127.dynamic.dsl.as9105.com) |
| 19:31.11 | [TK]D-Fender | keit does just what the docs say. It makes "exten => WHATYOUPUT,1,NoOp" in the dialplan (as loaded) dynamic when the device is registered |
| 19:31.43 | *** join/#asterisk zkn (~zkn@82.131.68.58.cable.starman.ee) |
| 19:32.12 | zkn | what causes this alert: chan_sip.c:28467 reload_config: No valid transports available, falling back to 'udp'. ? |
| 19:32.33 | [TK]D-Fender | zkn, Not having listed UDP or TCP exclusively or explicitly |
| 19:34.53 | *** join/#asterisk Russ (~russ@conference/linaro-connect/x-kavcsgjgulefppyv) |
| 19:35.06 | zkn | and how would I need to do that? |
| 19:36.02 | zkn | udpbindaddr ? |
| 19:36.07 | zkn | tcpbindaddr ? |
| 19:37.28 | [TK]D-Fender | Something like that, don't recall the actual parm names for these offhand. The samples list them... |
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| 19:46.37 | *** join/#asterisk fofware (~fabian@host63.190-31-63.telecom.net.ar) |
| 19:51.13 | wdoekes2 | zkn: transport=udp |
| 19:52.08 | zkn | wdoekes2: interesting, I have by now disabled tcp support, and added transport=udp everywhere and still get this alert when reloading chan_sip.so |
| 19:52.30 | wdoekes2 | added to [general]? |
| 19:53.45 | zkn | nope, to user template |
| 19:54.19 | zkn | got it |
| 19:57.31 | zkn | now when I enable tcp, I start getting ximit Broken pipe errors |
| 19:58.46 | zkn | and all peers become UNREACHABLE |
| 20:00.36 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 20:02.58 | zkn | ok, simple module reload did not work, had to unload and the load to get things working |
| 20:04.24 | imcdona | I've got a SIP debug that I have a question about: http://pastebin.com/wR6fgFpw In this particular dialog you can see that asterisk replies with a 489 (Bad Event) but the channel stays open and never closes eventually causing the sip channel driver to stop accepting connections after thousands of thse. A SIP show channels shows the cahnnel active with the sIp timer inactive. |
| 20:05.03 | imcdona | does this mean the channel will live forever? As far as I can tell the cahnnels stay open and causes sip to stop responding |
| 20:07.27 | *** join/#asterisk mahables (~Jason@c-76-101-2-235.hsd1.fl.comcast.net) |
| 20:07.45 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 20:12.03 | mahables | Hi guys, I'm having some difficulty with distributed device state using AIS on Asterisk 1.8.9.2. From the CLI, "ais clm show members" shows both nodes on both servers, but when changing a cutom device state (from either node) the device state is not updated on the other node. Can anyone help? |
| 20:14.17 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 20:14.51 | *** join/#asterisk xertres_ (~xertres@h-136-1-238.a176.priv.bahnhof.se) |
| 20:22.55 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 20:27.10 | zkn | it's maybe kind of trivial to most of you here but does anyone know how to make vim editor actually use the syntax highlighting files that come with asterisk? there actually some versions already included with vim but nevertheless I get no syntax highlighting.. on the other hand on some linux distros it works automatically... |
| 20:28.35 | leifmadsen | zkn: like here? http://vimdoc.sourceforge.net/htmldoc/syntax.html |
| 20:30.02 | leifmadsen | I type "vim load syntax file" into google and got a bunch of hits |
| 20:30.03 | wdoekes2 | zkn: put 'set modeline' in your ~/.vimrc and then put '; set syn=asterisk:' at the top of your config file? |
| 20:30.09 | ChrisInSydney | zkn: From a windoze PC i use crimson (now emerald) editor and its FTP. |
| 20:30.18 | zkn | yes yes
dammit
vim |
| 20:30.26 | zkn | i just type vi |
| 20:30.29 | ChrisInSydney | I can DCC you a syntax file (good for 1.4) |
| 20:30.54 | zkn | on some distros it automatically launches vim even though i use vi command |
| 20:33.13 | leifmadsen | zkn: look at your .bashrc file then for aliases |
| 20:33.15 | zkn | all I had to do i use "alias vi=vim" |
| 20:33.24 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 20:33.38 | zkn | great stuff! |
| 20:40.33 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:40.39 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
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| 20:51.53 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 20:52.54 | Neptu | hej how i can run a extension.py in python?? |
| 20:53.09 | saxa | python extension.py |
| 20:53.15 | Neptu | where do i need to specify which extension will be consider |
| 20:53.29 | saxa | i don't know whats in extension.py |
| 20:53.46 | saxa | probably there is something inside the file explaining how to use it |
| 20:54.23 | saxa | where did you find this extension.py ? |
| 20:54.37 | Neptu | i mean u have extension.lua |
| 20:54.40 | Neptu | etc |
| 20:54.54 | saxa | Neptu: i hav eno idea |
| 20:55.12 | saxa | if you show me extension.py i can maybe help you something. |
| 21:01.34 | [TK]D-Fender | Neptu, Thre is no direct interface for python. |
| 21:02.11 | Neptu | [TK]D-Fender: so i need to code in lua then if i wan to build something more complex? |
| 21:02.24 | [TK]D-Fender | Neptu, Or use AGI's |
| 21:02.48 | Neptu | can i use perl? |
| 21:02.53 | mahables | Hi guys, I'm having some difficulty with distributed device state using AIS on Asterisk 1.8.9.2. From the CLI, "ais clm show members" shows both nodes on both servers, but when changing a cutom device state (from either node) the device state is not updated on the other node. Can anyone help? |
| 21:04.28 | Neptu | [TK]D-Fender: i was willing to connect to a database to retrieve numbers to be dialed and build a preplanned call plan thats why i need something more "fency" |
| 21:05.31 | [TK]D-Fender | Neptu, So far all of that logic might have a reason to happen outside of AGI and the dialplan altogether. |
| 21:06.32 | Neptu | [TK]D-Fender: do not get your point there... but i might need to do some stuff |
| 21:08.31 | [TK]D-Fender | Neptu, Depends what will trigger a number to be looked up, and where you are in the call process. |
| 21:10.06 | Neptu | yep yep agree |
| 21:10.39 | Neptu | anyway could be nice to have python to be able to make de coding... my knowlege of lua is next no 0 |
| 21:11.59 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 21:14.09 | [TK]D-Fender | Neptu, Same with pretty much everybody. I'm betting its usage is very very low in the grand scheme of things. |
| 21:14.32 | Neptu | [TK]D-Fender: feels like it... |
| 21:14.38 | *** join/#asterisk jblack (~jblack@pool-71-173-0-61.sctnpa.east.verizon.net) |
| 21:14.50 | [TK]D-Fender | Neptu, standard dialplan does most of what everyone needs, AGI tends to cover the rest. |
| 21:16.25 | Neptu | [TK]D-Fender: i have an small project, i want to build a callcenter and for that i need to define stuff on database like list of calls to be performed and dump the recordings of the conversations to be available in the web later on... so AGI will not do.. need heavy interaction with the db |
| 21:17.10 | [TK]D-Fender | Neptu, It is very unclear what this interaction is, and when it needs to take place. |
| 21:17.19 | [TK]D-Fender | Neptu, this may all be completly outside of * |
| 21:17.23 | ChrisInSydney | Neptu: How many simultaneous recording sessions ?? |
| 21:17.43 | ChrisInSydney | database is more straightforward. Recording can be a bitch |
| 21:17.46 | jblack | Agi works great for that. I've done it. But you can look at the DB() |
| 21:18.14 | Neptu | ChrisInSydney: at this point i need around 10 |
| 21:18.34 | ChrisInSydney | 10 is achievable. 50 is a different case |
| 21:18.50 | Neptu | ChrisInSydney: at this point i want to make it work with one server might happend that i need a couple of boxes... that i already know. |
| 21:18.52 | [TK]D-Fender | lol... |
| 21:19.06 | [TK]D-Fender | 50 = not acheivable? That is a sad diagnosis |
| 21:19.51 | ChrisInSydney | [TK]D-Fender: 50 is achievable, but you need to consider more investment in the hardware |
| 21:19.52 | Neptu | anyway im just getting familiar with the system im trying to get a dialplan to work from scratch and seems asterisk comes with a lot there by default |
| 21:20.01 | Neptu | dialplan show gives me a lot |
| 21:20.10 | ChrisInSydney | didnt say it wasnt achievable ;p |
| 21:20.18 | Neptu | hehehe |
| 21:20.28 | [TK]D-Fender | ChrisInSydney, Hardly... |
| 21:20.51 | [TK]D-Fender | Neptu, those are sample configs and shouldn't be used |
| 21:20.52 | ChrisInSydney | ok, perhaps i didnt express myself correctly |
| 21:21.01 | [TK]D-Fender | ChrisInSydney, Or you're doing it wrong :) |
| 21:21.10 | Neptu | ChrisInSydney: i have a crazy idea ive been reading in the book u can "cluster" some boxes and redirect stuff i was just wondering if rasberry pi can candle that preasure and scale good on this |
| 21:21.47 | Neptu | so you can get cheap scale on your operation using cheap hw |
| 21:22.06 | [TK]D-Fender | Neptu, You want to run a call center off a dongle? That would probably get called things like "tragic" |
| 21:22.32 | [TK]D-Fender | Neptu, And * doesn't scale by clustering little jusnk toys like that |
| 21:22.58 | Neptu | i mean each of those must handle atleast 4 users and cost 39 dolars |
| 21:23.09 | Neptu | ok |
| 21:23.18 | Neptu | just ideas of a newbie |
| 21:24.41 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:24.41 | ChrisInSydney | [TK]D-Fender: Just getting a coffee. Help my brain. Its morning here |
| 21:24.59 | [TK]D-Fender | ChrisInSydney, Caffeine guuuuuuuuuuuddddddd |
| 21:25.13 | ChrisInSydney | Neptu: Have a look at openWRT / DD-WRT and you can get Linux on a router. |
| 21:26.01 | ChrisInSydney | a bit more than $30, but for $90 here wholesale gets you a Netgear WNR3500L with a 450 MIPS CPU, 8 ROM 64 RAM and USB |
| 21:26.32 | ChrisInSydney | Gotta grind those beans |
| 21:26.38 | ChrisInSydney | brb |
| 21:27.07 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 21:29.00 | Neptu | ChrisInSydney: yep i saw that some friends are using that as multipurpouse box anyway im courious the new arm mali 608 has gpu comping and that should speed a lot the coded calculation... will be interesting to see it running on this and the price is not over 50$ per board... |
| 21:30.15 | [TK]D-Fender | Neptu, That thing should handle a few users if you minimize the load on it. |
| 21:30.44 | [TK]D-Fender | Neptu, I'd say it'd be useful for minimal SIP proxy use for remote clients behind NAT, etc |
| 21:31.00 | ChrisInSydney | I am a big fan of atom chips. Cheap boards, low power |
| 21:31.04 | Neptu | that is definetly good stuff :D |
| 21:31.17 | Neptu | ChrisInSydney: thumbs up!! |
| 21:31.28 | Neptu | what is here incls/swits/igpats |
| 21:31.31 | ChrisInSydney | that ways its x86/ |
| 21:31.39 | ChrisInSydney | easy to compile |
| 21:31.39 | Neptu | this is including a lot of stuff in my dialplan |
| 21:31.59 | Neptu | ChrisInSydney: that can be argue but i kind of agree |
| 21:31.59 | [TK]D-Fender | Neptu, If that's the sample, TRASH IT. |
| 21:32.10 | [TK]D-Fender | And on that not... checkout time, heading home... BBIAB |
| 21:32.39 | Neptu | [TK]D-Fender: that comes by default is just my diaplan is bigger than the bible and i did not put nothing myself... |
| 21:32.40 | ChrisInSydney | if you are worried about scaling, I have a few 30 - 40 users running on atom 1.6 hypertthread with a gig of ram |
| 21:33.08 | Neptu | ChrisInSydney: that is quite ok... the book says the performace is quite low |
| 21:33.22 | Neptu | i was expecting a lot less than that |
| 21:33.59 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-230.dsl.net.pk) |
| 21:34.01 | ChrisInSydney | nahh. Pushed out 300 handsets on a early Celeron D with 1/2 G ram |
| 21:34.08 | ChrisInSydney | in apinch |
| 21:34.11 | ChrisInSydney | a pinch |
| 21:34.25 | Neptu | ChrisInSydney: that sounds great |
| 21:34.31 | ChrisInSydney | not 300 calls though |
| 21:34.37 | ChrisInSydney | maybe 30 max |
| 21:34.41 | Neptu | ChrisInSydney: i will try my stuff this week see how much i can get from it |
| 21:35.01 | Neptu | i will be happy now with 10 persons in a conference room |
| 21:35.10 | ChrisInSydney | mostly calls to the astDB not an SQL |
| 21:35.32 | ChrisInSydney | get a linode and screw with that. See what happens |
| 21:35.46 | ChrisInSydney | You can also get some Amazon EC2 stuff for 12 momnths free |
| 21:38.19 | Neptu | ChrisInSydney: mmmmmmm im trying now openstack |
| 21:38.34 | mechbangirc | I have asterisk 1.8.7 on a box which performs great under light load, Under heavy load (like 50-60 concurrent calls) Asterisk returns error message 603 decline busy/congestion. I searched a little 603 is generated from my server not from upstream carrier, caller is also running asterisk. Any idea? |
| 21:38.34 | Neptu | some friends have that is a EC2 opensource alternative |
| 21:38.44 | Neptu | anyway lets get down to config this |
| 21:39.00 | mechbangirc | I know already it is not channel capacity issue. |
| 21:39.13 | Neptu | http://pastebin.com/TZv8DLgE ->> do you see why this is including all kind of garbage to the dialplan??? i do not see nothing |
| 21:41.31 | ChrisInSydney | I'll pour that codffee and have a squizz |
| 21:41.48 | ChrisInSydney | mechbangirc: Whats your CPU load like ?? |
| 21:41.59 | ChrisInSydney | are you doing transcoding ?? |
| 21:42.04 | ChrisInSydney | exp to 729 |
| 21:42.05 | mechbangirc | it is under 50% |
| 21:42.13 | mechbangirc | G729 all the way |
| 21:42.20 | ChrisInSydney | in and out |
| 21:42.24 | mechbangirc | yes |
| 21:42.25 | ChrisInSydney | SIP ?? |
| 21:42.29 | mechbangirc | yes SIP |
| 21:42.59 | ChrisInSydney | voice prompts all that |
| 21:43.02 | ChrisInSydney | (have to ask) |
| 21:43.26 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:43.27 | mechbangirc | nothing, basically selling traffic as wholeseller, dont even answer the call |
| 21:44.41 | ChrisInSydney | going to pur a coffee.....and drink it.....need my brain to stop squeeking.... |
| 21:44.44 | ChrisInSydney | brb |
| 21:45.38 | *** part/#asterisk jblack (~jblack@pool-71-173-0-61.sctnpa.east.verizon.net) |
| 21:45.55 | Neptu | lets see the basics if i modify extensions.config and i do dialplan reload from the cli the extension then the new options in extension.conf must be loaded right? |
| 21:46.03 | ChrisInSydney | That stopped the rocker cover noise |
| 21:47.04 | mechbangirc | 603 decline is for bad configuration right? |
| 21:48.18 | ChrisInSydney | mechbangirc, Neoptu: Got a paid support call to attend to |
| 21:48.21 | ChrisInSydney | wil be back |
| 21:49.35 | mechbangirc | nice! |
| 21:51.12 | *** join/#asterisk dxd828 (~dxd828@88-109-118-127.dynamic.dsl.as9105.com) |
| 21:51.15 | Neptu | ChrisInSydney: see ya |
| 21:51.16 | *** part/#asterisk dxd828 (~dxd828@88-109-118-127.dynamic.dsl.as9105.com) |
| 21:59.24 | *** join/#asterisk ipconfeng (~ipconfeng@pool-98-114-57-252.phlapa.fios.verizon.net) |
| 21:59.46 | ChrisInSydney | mechbangirc: Look up Kamailio. It may help |
| 22:00.11 | ChrisInSydney | Neptu: Are you using AEL ?? |
| 22:00.26 | ipconfeng | can anyone recommend a good way to detect and alert and fraud? |
| 22:01.16 | Neptu | ChrisInSydney: nop seems asterisk was not reloading the dialplan correctly... the files where not there and it keeped on reloading the same stuff... rebooted the box and now its fine |
| 22:01.23 | mechbangirc | ChrisInSydney: 60 concurrent calls is not something asterisk can not handle, there must be some problem |
| 22:02.12 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 22:02.27 | ChrisInSydney | There was a problem there. Got dropped |
| 22:02.30 | ChrisInSydney | :-/ |
| 22:02.33 | ChrisInSydney | another call |
| 22:02.45 | ChrisInSydney | bbr. 60 calls should be OK |
| 22:03.16 | ChrisInSydney | Mind you Kamailio is probably the correct hammer for your application |
| 22:04.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:05.52 | ChrisInSydney | Neptu: Strange. Someo sort of caching |
| 22:06.04 | Neptu | ChrisInSydney: yep |
| 22:07.25 | ChrisInSydney | mechbangirc: What billing are you using. Does it use AGI ? |
| 22:07.47 | Neptu | [Feb 15 22:04:10] WARNING[16630]: chan_sip.c:29234 reload_config: No valid transports available, falling back to 'udp'. |
| 22:07.50 | Neptu | this is weird |
| 22:08.28 | mechbangirc | ChrisInSydney: yes it uses AGI, I use A2billing |
| 22:08.30 | ChrisInSydney | Neptu: 1.8 ? |
| 22:08.38 | Neptu | latest |
| 22:08.40 | Neptu | 10 |
| 22:08.43 | ChrisInSydney | Asterisk Forums View topic - SIP 603 Declined after AGI execution |
| 22:08.48 | ChrisInSydney | ahh |
| 22:09.03 | mechbangirc | thanks lemme check |
| 22:09.03 | ChrisInSydney | Neptu: I dont know if I am properly qualified. |
| 22:09.20 | Neptu | ChrisInSydney: ;) do not worry be happy |
| 22:09.24 | Neptu | gonna start trying |
| 22:10.04 | ChrisInSydney | Just jumped from 1.4 to 1.8. Still reworking the database and dial plans before we deploy across all our clients |
| 22:10.56 | ChrisInSydney | mechbangirc: asterisk2billing.org View topic - SIP response 603 and FAILOVER . |
| 22:11.14 | ChrisInSydney | Mind you they are single posts, no responses |
| 22:11.21 | ChrisInSydney | but you are not alone |
| 22:11.41 | *** join/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 22:11.46 | mechbangirc | I read those, I am not using failover |
| 22:12.26 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 22:17.59 | zkn | where/how could I get a list of all native column names for a CDR table? |
| 22:19.28 | saxa | No translator path exists for channel type DAHDI (native 0x4c (ulaw|alaw|slin)) to 0x400 (ilbc) |
| 22:19.35 | saxa | what exactly this means ? |
| 22:19.41 | saxa | a codec problem ? |
| 22:20.08 | [TK]D-Fender | yes |
| 22:20.31 | saxa | I'm trying to find out why my SIP phones calls but no audio whne I get the call answered |
| 22:21.03 | saxa | but is no codec compilled in * or is it a problem in my phone ? |
| 22:21.20 | saxa | ok let me try to use only ulaw |
| 22:22.18 | ChrisInSydney | Neptu: You can stop loading the modules you dont want., such as AEL. Should be in modules.conf |
| 22:23.35 | ChrisInSydney | mechbangirc: YOu are stretchng my knowledge. Where does it get to in the dial plan or does it just dump with a 603 ? |
| 22:24.06 | saxa | ok using only ulaw places the call and the other side rings, but I can not hear anything from DAHDI |
| 22:24.33 | saxa | lets try gsm |
| 22:24.37 | mechbangirc | ChrisInSydney: interestingly it is happening to 1709 prefixed numbers. I am double checking if I have a route. LOL can it be... |
| 22:25.11 | Neptu | ChrisInSydney: is working now is the dam bad quality of the samples :P |
| 22:26.13 | saxa | [Feb 15 23:23:59] NOTICE[2279]: chan_sip.c:9093 process_sdp: No compatible codecs, not accepting this offer! |
| 22:26.56 | ChrisInSydney | Neptu: Then start with a hello world dial plan and build from there |
| 22:27.30 | Neptu | ChrisInSydney: already on it so far is working nicely |
| 22:28.08 | Neptu | ChrisInSydney: how i can configure the codec with the better sound quality?? gsm or ulaw? |
| 22:28.22 | mechbangirc | I do have routes, why to only 1709, my system just wont forward these to carrier. |
| 22:28.29 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:28.30 | ChrisInSydney | g722 |
| 22:28.31 | ChrisInSydney | :D |
| 22:29.03 | Neptu | lets si if 3cx supports that |
| 22:29.04 | ChrisInSydney | from bad to best gsm, 729, 711 (ulaw/alaw) g722 |
| 22:29.28 | ChrisInSydney | 729 you need a license to do transcoding |
| 22:29.55 | ChrisInSydney | I just use alaw as I am in Aus. ulaw is standard across the states |
| 22:30.07 | Neptu | ok |
| 22:30.14 | ChrisInSydney | 722 is sweet if your handsets support it |
| 22:30.27 | ChrisInSydney | you have a SIP handset |
| 22:30.29 | Neptu | since all my devices are android phones or softphones i can go for g722 |
| 22:30.35 | ChrisInSydney | cool |
| 22:30.41 | *** join/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 22:30.59 | Neptu | allow=g722 disallow=all |
| 22:31.04 | Neptu | hehehe lets see |
| 22:31.17 | mechbangirc | disallow=all then allow=g722 |
| 22:31.18 | [TK]D-Fender | And put a transcoding load on your system for every call that isn't between those devices |
| 22:31.46 | Neptu | [TK]D-Fender: then im lost how i do that? |
| 22:32.24 | ChrisInSydney | Neptu: Most carriers dont do 722. You will be transcoding fro all of your calls |
| 22:32.31 | Neptu | ahhh big shit the andorid 3cx only suports ulaw speex or gsm |
| 22:32.36 | ChrisInSydney | outbound. But it sounds nice |
| 22:32.47 | ChrisInSydney | ulaw it is |
| 22:32.50 | ChrisInSydney | Media5 ?? |
| 22:32.54 | Neptu | ChrisInSydney: that is for the output most of my calls are internal calls not outside |
| 22:33.30 | Neptu | let me check android market see what supports g722 |
| 22:33.32 | ChrisInSydney | mechbangirc: So its only to one carrier, or to one area code that you get the 603s |
| 22:33.42 | ChrisInSydney | Media5, Bria |
| 22:34.10 | Neptu | ai ai |
| 22:34.22 | Neptu | media5-fone Medias corporation installing |
| 22:34.49 | ChrisInSydney | They work well on iPhone (sorry for swearing) |
| 22:35.09 | Neptu | then the problem are the samples... mmmm transcoding will be needed |
| 22:35.20 | Neptu | ChrisInSydney: be happy :D |
| 22:35.52 | ChrisInSydney | Has anyone here seen the South Park " Huma Cent-iPad" episode ? |
| 22:36.01 | *** join/#asterisk chasing`Sol (~cS@197.132.87.255) |
| 22:36.04 | ChrisInSydney | human |
| 22:36.28 | Neptu | hahaha i just got tired of southpark long time ago |
| 22:36.34 | Neptu | i saw kenny die to many times |
| 22:36.40 | Neptu | :D |
| 22:36.56 | ChrisInSydney | You need to watch it if you have an Apple product or use iTunes |
| 22:37.04 | ChrisInSydney | (sorry for swaring again) |
| 22:37.36 | Neptu | hahaha i have an ibook with linux installed do not know if cualifies me or not... |
| 22:39.37 | *** join/#asterisk dxd828 (~dxd828@88-109-118-127.dynamic.dsl.as9105.com) |
| 22:39.40 | Neptu | ah media5 is not for free u need to pay to get g722 |
| 22:39.40 | *** part/#asterisk dxd828 (~dxd828@88-109-118-127.dynamic.dsl.as9105.com) |
| 22:39.46 | ChrisInSydney | yup |
| 22:39.58 | *** join/#asterisk GVolkmann (~administr@rrcs-24-213-178-60.nyc.biz.rr.com) |
| 22:42.30 | Neptu | checking alternatives |
| 22:42.53 | Neptu | its a pitty you can not code the dialplan in python... |
| 22:43.07 | GVolkmann | exit |
| 22:43.09 | GVolkmann | quit |
| 22:43.35 | Neptu | looks at GVolkmann |
| 22:43.56 | saxa | how can I check which codec is supported by dahdi ? |
| 22:44.08 | *** join/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 22:44.48 | saxa | core show codecs shows all the codecs I have, correct ? |
| 22:46.20 | ChrisInSydney | Neptu: YOu could via AGI to an extent. You could move over to YATE, I think that they have a API set for Python |
| 22:47.12 | Neptu | i will check that |
| 22:47.37 | Neptu | just disapointed no softphone free fro android with g722 |
| 22:48.30 | ChrisInSydney | AGI should give you everything you need |
| 22:48.40 | ChrisInSydney | another call |
| 22:48.41 | ChrisInSydney | brb |
| 22:48.50 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 22:48.57 | navaismo | ~book |
| 22:48.57 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:49.32 | zkn | hmm
i've added a new column (holdtime) to my car table in mysql, i use adaptive_odbc, when I answer a call in queue app I send the call with gosub to another context where i have written "Set(CDR(holdtime)=${QEHOLDTIME})", but after this the logging to my backend has stopped completely, what could be wrong? |
| 22:50.49 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 22:54.29 | Neptu | ChrisInSydney: thanks for the help, gonna get a shower and try to sleep 11:54 pm here |
| 22:54.31 | Neptu | ;) |
| 22:54.46 | Neptu | see ya |
| 22:55.03 | p3nguin | saxa: No, it does not show you what codecs you have available. |
| 22:55.50 | p3nguin | saxa: If you want to see what codecs you have available, you can check the translation table: core show translation |
| 23:04.21 | saxa | oh p3nguin thx, let me see |
| 23:05.01 | *** join/#asterisk Russ (~russ@conference/linaro-connect/x-ydeszzfkcwdvulzv) |
| 23:08.28 | saxa | http://pastebin.com/N2e5zgVs, ok I don't understand why it won't call from one box to another via iax. I get no authority found, but one box is registered to another. I have both contexts set up, it was working and it does not work anymore. |
| 23:08.47 | saxa | http://pastebin.com/N2e5zgVs |
| 23:08.57 | saxa | huh |
| 23:09.20 | saxa | I know you guys soon will kick me out of there :) |
| 23:10.31 | p3nguin | The first paste is an invalid ID. |
| 23:10.55 | p3nguin | Oh, I see what happened. |
| 23:10.56 | saxa | yes, its the same as the second, I put a comma without space |
| 23:11.09 | saxa | really ? |
| 23:11.14 | p3nguin | The URL included the trailing comma. |
| 23:11.21 | saxa | yeah |
| 23:11.22 | p3nguin | http://pastebin.com/N2e5zgVs, = invalid paste ID. |
| 23:11.28 | saxa | lol |
| 23:11.51 | saxa | thats why I pasted it again :D |
| 23:14.09 | p3nguin | Do you have access to both systems? |
| 23:14.15 | saxa | yes |
| 23:14.28 | saxa | i do not see any incoming call on the called CLI |
| 23:14.35 | saxa | when I try to call |
| 23:15.01 | p3nguin | I'd like to see the peer entries for each of the systems which appears on the other system. |
| 23:15.18 | saxa | ok let me paste both |
| 23:15.41 | p3nguin | If you are using register statements, be sure to include them. |
| 23:16.56 | saxa | http://pastebin.com/zubA3rj1 |
| 23:17.20 | saxa | let me paste the iax.conf on both sides |
| 23:17.24 | *** join/#asterisk volga629 (~slava@76-10-130-18.dsl.teksavvy.com) |
| 23:17.53 | volga629 | flipityflop: Hi, I got fonebridge working |
| 23:18.18 | volga629 | Waiting when PRI will be connected by ISP |
| 23:18.28 | zkn | how could I log queue hold time to CDR ? |
| 23:19.30 | saxa | http://pastebin.com/pZbHC6kW |
| 23:19.44 | saxa | ^^^ here are the iax.conf of both sides |
| 23:24.08 | zkn | or let's be more specific: how could I extract the variables in Queue app and have their values logged to cdr_adaptive_odbc ? |
| 23:24.38 | saxa | I have read voip-info.org that this kind of error No authority found can also be because it can't find the context |
| 23:24.58 | saxa | but both contexts exist |
| 23:31.25 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 23:33.27 | p3nguin | saxa: If those are your real user/pass in there, I'd recommend changing it right away. |
| 23:33.38 | saxa | yes |
| 23:33.45 | saxa | I will change them now |
| 23:34.02 | saxa | but I want to figure out why it doesnt call |
| 23:35.01 | p3nguin | Are those two hosts using static IP addresses? |
| 23:37.54 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mnfpllaavqidvtym) |
| 23:38.13 | saxa | only one |
| 23:38.23 | p3nguin | Which one is static? |
| 23:38.27 | saxa | the rcitaly |
| 23:38.35 | saxa | 217 |
| 23:39.04 | saxa | should I remove host=dynamic in it ? |
| 23:39.05 | p3nguin | Then rcitaly does not need to register to braserv, so I'll edit that accordingly in the pastebin so you can adjust it soon. |
| 23:39.42 | saxa | ok |
| 23:41.04 | saxa | but the book, on page 112 and 113 has in the examples both with static ip and is using host=dynamic |
| 23:41.28 | [TK]D-Fender | saxa: your usernames are all wrong |
| 23:41.31 | saxa | both, osaka and toronto have host=dynamic, |
| 23:41.34 | [TK]D-Fender | saxa: they don't match |
| 23:41.43 | saxa | but why it registers then ? |
| 23:42.15 | [TK]D-Fender | because the REGISTER has the right user on the other side, but the PEERS do not |
| 23:42.35 | saxa | let me recheck, |
| 23:43.00 | [TK]D-Fender | You also have each side registering... this is not needed |
| 23:44.35 | saxa | ok I don't understand then why, I was following this example in the book, and replaced the usernames |
| 23:45.02 | [TK]D-Fender | saxa: You really aren't looking at the peers |
| 23:45.20 | [TK]D-Fender | And somebody has to be fixed over there |
| 23:45.23 | saxa | but when only one server was registered it was working , but the rcitaly then was spawning lots of rejected messages |
| 23:45.46 | [TK]D-Fender | saxa: your peers do not match,. |
| 23:45.46 | saxa | is confused :) |
| 23:45.49 | p3nguin | Perhaps this example configuration can help you: http://pastebin.com/vPzpwysV |
| 23:45.52 | [TK]D-Fender | the usernames are different |
| 23:45.56 | [TK]D-Fender | mismatch = reject |
| 23:46.02 | [TK]D-Fender | it is as obvious as it looks |
| 23:46.08 | saxa | exactly |
| 23:46.27 | [TK]D-Fender | in fact .... the secrets don't even match |
| 23:46.35 | [TK]D-Fender | what is that mess you've shown us |
| 23:46.37 | [TK]D-Fender | ? |
| 23:48.13 | p3nguin | saxa: Try this: http://pastebin.com/sF40ed5P |
| 23:48.26 | p3nguin | Use it exactly as I have entered it. |
| 23:49.00 | [TK]D-Fender | Still messed up |
| 23:49.09 | p3nguin | What do you think is wrong with it? |
| 23:49.20 | [TK]D-Fender | nope, hold on that |
| 23:50.06 | p3nguin | I think it's right. |
| 23:50.26 | [TK]D-Fender | permit & deny are worthless currently |
| 23:50.31 | p3nguin | We've accounted for the user name on the opposite host and we use a matching password. |
| 23:50.49 | p3nguin | I didn't edit his ACLs. |
| 23:50.59 | [TK]D-Fender | The rest seems OK |
| 23:51.01 | p3nguin | Just made the rest of it so it should work. |
| 23:52.18 | p3nguin | saxa: Don't leave me hanging. |
| 23:53.37 | saxa | http://pastebin.com/7tipgVae |
| 23:53.46 | saxa | thats how i modified it |
| 23:53.55 | saxa | but i still get the no authority |
| 23:54.02 | p3nguin | Use the one I GAVE YOU. |
| 23:54.13 | saxa | oh i didnt see |
| 23:54.20 | p3nguin | (1748.13) <p3nguin> saxa: Try this: http://pastebin.com/sF40ed5P |
| 23:54.26 | p3nguin | ^ important |
| 23:54.34 | saxa | yep |
| 23:54.37 | saxa | i saw it now |
| 23:54.52 | p3nguin | (1748.26) <p3nguin> Use it exactly as I have entered it. |
| 23:55.02 | p3nguin | ^ also important |
| 23:58.33 | *** join/#asterisk Russ (~russ@conference/linaro-connect/x-axdfjtwtqybzpdvy) |
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