| 00:00.30 | navaismo | mattwj2002, yeah there are many blogs talking about that |
| 00:00.43 | mattwj2002 | yeah I'll dig around |
| 00:01.03 | mattwj2002 | navaismo: I have rediscovered sip recently |
| 00:01.33 | mattwj2002 | not to push a particular sip provider but I forgot how cool it is |
| 00:01.35 | mattwj2002 | :) |
| 00:01.51 | mattwj2002 | I am using callwithus dang voip is inexpensive |
| 00:01.52 | mattwj2002 | :) |
| 00:04.01 | mattwj2002 | navaismo: last question (I promise) |
| 00:04.18 | mattwj2002 | is a rpi fast enough to create a conference bridge? |
| 00:04.21 | navaismo | ask |
| 00:04.44 | navaismo | in my enviroment RPI only suppirt three calls at decent quality |
| 00:04.58 | navaismo | of course it depends on what else you are running |
| 00:05.04 | mattwj2002 | of course |
| 00:05.05 | navaismo | in my case apache and mysql |
| 00:05.30 | mattwj2002 | I am assuming you only want one codec no transcoding in other words |
| 00:05.31 | navaismo | but if you stay in plan asterisk i guess you can handle maximum 5 calls |
| 00:05.36 | navaismo | yes |
| 00:06.03 | mattwj2002 | thank you! |
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| 03:06.29 | Shawn0228 | ok. so i have a weird one. I want to take my google voice number, have it set up as an inbound route, ring my extension in my office, as well as ring my pots cell phone. i have figured out that part. the hard part is if there is any way to pass along the caller id from the inbound google voice, to the outbound google voice number that I will be using to connect the call through my cell phone. |
| 03:06.59 | Shawn0228 | anyone have any ideas? |
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| 03:28.16 | [TK]D-Fender | <PROTECTED> |
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| 03:43.19 | Shawn0228 | [TK]D-Fender: ive already googled it, thank you for that though. i might be explaining this a little unclear. I have two google voice numbers, the one i currently have tied to my cell phone, and the one currently tied to my asterisk box. i want to tie both of them to the asterisk box, with the one from the cell phone coming in, and the one from the home phone going out. if i were to get a call from the cell phone, asterisk would |
| 03:43.19 | Shawn0228 | <PROTECTED> |
| 03:43.19 | Shawn0228 | <PROTECTED> |
| 03:44.26 | [TK]D-Fender | If you followed what all those link suggestions showed you'd see the common theme : You can't. |
| 03:45.49 | Shawn0228 | much better response than a google link. thanks :D. just trying to see if someone here, through common ingenuity/hackery figured it out. |
| 03:46.23 | Shawn0228 | im wondering if i can figure out how to pass along the caller id as a toast message to my android before it connects the call. some programming is definitely involved... |
| 03:46.26 | [TK]D-Fender | No hack. They don't care what you send. |
| 03:46.48 | [TK]D-Fender | If you want to set outbound callerid you'll have to use a service that allows it |
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| 06:33.47 | iulhk | wanted to make a call center setup, having pstn digital line (RJ 45), can anybody recomend upto 8 channels supported card for inbound call center setup ? |
| 06:39.30 | [TK]D-Fender | What kind of line EXACTLY is it? |
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| 06:51.31 | iulhk | <[TK]D-Fender> fiber optic line, RJ 45 |
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| 06:51.55 | [TK]D-Fender | that is a connector type. that says nothing about the signalling on it |
| 06:52.18 | eirirs | there are no such thing as a RJ45 fiber optic |
| 06:52.41 | [TK]D-Fender | Rj45 = copper |
| 06:53.19 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 06:53.27 | iulhk | yes, |
| 06:53.33 | iulhk | let me little explain |
| 06:54.34 | enoch | hi all, im building a remote pbx that will have 3 aix lines and will do some automatic calls. Will it run smooth on a Atom 230 ( 2x1.6ghz ) with 1gb of ram?? |
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| 06:54.49 | enoch | sorry iax* |
| 06:56.11 | [TK]D-Fender | enoch: My watch can handle that... and it's ANALOG |
| 06:56.20 | iulhk | i have done call center setup by using DIDs, first time i will be doing this, want to make inbound call center setup, office located in dubai, they told me they have upto 8 channels supported RJ45 connector in their office, i need router for this ? |
| 06:56.45 | [TK]D-Fender | iulhk: that tells us NOTHING about the signalling on that line. |
| 06:57.17 | [TK]D-Fender | iulhk: You need real details on exactly what that is. |
| 06:57.19 | enoch | lol |
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| 07:04.05 | iulhk | <[TK]D-Fender> help me to understand, i have done testing with spa3102 but its analog adapter, by using spa 3102 as its 1 fxo port and 1 channel supported card, i am to able to take calls directly in my sip, no need for any zaptel, according to my understanding its PSTN line with having RJ45 connector, supported upto 8 channels, and origantor provider can increase channel from the back-end |
| 07:04.05 | iulhk | as i heard, your valuable information will be more encourging |
| 07:04.45 | [TK]D-Fender | iulhk: You keep talking "RJ45". that is NOT useful |
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| 07:05.19 | [TK]D-Fender | iulhk: You seem to have no understandaing of the SIGNALLING on that line. Go call your provider and get that answer. We cannot help you until you do |
| 07:10.39 | [TK]D-Fender | me heads to bed.... |
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| 09:21.11 | mattwj2002 | hi guys |
| 09:21.12 | mattwj2002 | :) |
| 09:21.25 | mattwj2002 | meetme is that still in asterisk 11? |
| 09:22.49 | mattwj2002 | or should I be using ConfBridge? |
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| 09:35.26 | boratynskikamil | Hello. |
| 09:37.06 | mattwj2002 | hi boratynskikamil do you know anything about confbridge? |
| 09:37.07 | mattwj2002 | :) |
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| 09:37.51 | Psi|4ward | Hi there, updated from 1.6 to 1.8 but getting chan_sip.c:20354 handle_response_invite: Failed to authenticate on INVITE |
| 09:38.08 | Psi|4ward | is there anything wrong with Dial(SIP/${DENG}@pbx&SIP/${GIGASET}@pbx,10); |
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| 09:45.48 | boratynskikamil | Question. Assume I have 5 SIP accounts. Is it good to create extension like that: http://wklej.org/id/1282713/ |
| 09:53.42 | mattwj2002 | Asterisk! |
| 09:53.54 | mattwj2002 | I have a conference bridge :) |
| 09:55.12 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:d5c5:773c:d278:a69b) |
| 10:01.17 | boratynskikamil | What's wrong in my extensions? |
| 10:02.01 | mattwj2002 | boratynskikamil: I would love to help |
| 10:02.07 | mattwj2002 | but that is way beyond me |
| 10:02.08 | mattwj2002 | :) |
| 10:02.27 | mirela666 | boratynskikamil: If you want HelpDesk to ring them all |
| 10:03.04 | boratynskikamil | mirela666: Yes, I wold like to. |
| 10:03.08 | mirela666 | boratynskikamil: you need to do same => 1,Dial(SIP/HD-01&SIP/HD-02&...) |
| 10:03.31 | boratynskikamil | And isn't it possible to make it different extension? |
| 10:03.41 | mirela666 | boratynskikamil: his way you are overridding extension 1 5 times in that helpdesk context |
| 10:04.27 | mirela666 | if you want to ring them one by one, first one leave as 1 and edit rest as n |
| 10:04.27 | boratynskikamil | Ouch... |
| 10:04.40 | mirela666 | * will count them as 2,3,4,5 steps |
| 10:04.57 | boratynskikamil | So they will not get connection at same time, right? |
| 10:05.30 | mirela666 | first on will answer the call and rest will stp ringing |
| 10:07.24 | boratynskikamil | mirela666: Got it. Or mayby is it possible to make it better look like? I mean Dial(SIP/01& \newline ? |
| 10:07.52 | mirela666 | no, but you can use Queue |
| 10:08.00 | mirela666 | core show application Queue |
| 10:08.45 | mirela666 | that is more elegant and have a lot of fine tuning :D |
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| 10:12.41 | mattwj2002 | sweeet |
| 10:12.54 | mattwj2002 | I have a confrence bridge with music! |
| 10:12.55 | mattwj2002 | :) |
| 10:13.51 | mattwj2002 | music on hold |
| 10:13.52 | mattwj2002 | :) |
| 10:16.19 | whizzi | nice :) |
| 10:16.33 | mattwj2002 | in all fairness |
| 10:16.43 | mattwj2002 | I had to use a lot of examples online |
| 10:16.48 | mattwj2002 | but it works! |
| 10:17.34 | whizzi | I should fix that as well on my phone system. I just noticed I'm still using MeetMe under Asterisk 11 and I don't have a Dahdi device |
| 10:18.04 | mattwj2002 | whizzi: I have this bridge running on an raspberry pi |
| 10:18.11 | mattwj2002 | a $35 conference bridge! |
| 10:18.13 | mattwj2002 | :) |
| 10:18.16 | whizzi | :D |
| 10:18.32 | mattwj2002 | indeed |
| 10:18.58 | mattwj2002 | no external numbers |
| 10:19.08 | mattwj2002 | you have to be on my home network at the moment |
| 10:19.12 | mattwj2002 | but it is cool |
| 10:19.13 | mattwj2002 | :) |
| 10:19.38 | *** join/#asterisk imox (~imox@p57A97199.dip0.t-ipconnect.de) |
| 10:19.52 | whizzi | I'm running a full blown Asterisk PBX with call-recording and using Googletts.agi to pronounce voices on an Atom 1.4Ghz machine :P |
| 10:20.05 | mattwj2002 | nice Atom processors! |
| 10:20.24 | mattwj2002 | aka almost a raspberry pi but I also run windows if necessary |
| 10:20.28 | mattwj2002 | :P |
| 10:20.38 | mattwj2002 | not quite but you know what I mean |
| 10:20.52 | mattwj2002 | what is googletts.agi? |
| 10:21.25 | whizzi | text2speech for asterisk using Google Translate |
| 10:21.33 | whizzi | http://zaf.github.io/asterisk-googletts/ |
| 10:21.47 | mattwj2002 | oh neat |
| 10:21.50 | whizzi | it's awesome |
| 10:21.59 | mattwj2002 | does it sound good? |
| 10:22.22 | whizzi | English pronunciation is fair, Dutch is kinda odd (sounds like an old Dutch queen) |
| 10:22.35 | mattwj2002 | lol |
| 10:22.42 | mattwj2002 | I don't know Dutch! |
| 10:22.42 | whizzi | Russian is fair as well, so is German |
| 10:22.43 | mattwj2002 | :P |
| 10:22.53 | whizzi | and those are the languages I tested |
| 10:22.58 | whizzi | and understand most of the time |
| 10:23.05 | mattwj2002 | or Russian or German (well one sentence and one word of German) |
| 10:23.08 | mattwj2002 | :) |
| 10:23.44 | mattwj2002 | it is something like spec in Duetch |
| 10:23.48 | mattwj2002 | and nine |
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| 10:23.58 | mattwj2002 | I don't even know how to spell it |
| 10:23.59 | mattwj2002 | :P |
| 10:24.03 | whizzi | "nein" :P |
| 10:24.17 | mattwj2002 | no whatever |
| 10:24.28 | mattwj2002 | nein = no? |
| 10:25.10 | mattwj2002 | oh well |
| 10:25.10 | mattwj2002 | :) |
| 10:25.18 | whizzi | yes |
| 10:26.14 | mattwj2002 | whizzi: I am American |
| 10:26.20 | mattwj2002 | all I know is English! |
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| 10:26.35 | mattwj2002 | I took two years of Spanish but I suck so bad at it |
| 10:26.41 | mattwj2002 | :) |
| 10:27.27 | *** part/#asterisk whizzi (~whizzi@82-171-3-8.ip.telfort.nl) |
| 10:27.40 | *** join/#asterisk whizzi (~whizzi@82-171-3-8.ip.telfort.nl) |
| 10:27.50 | whizzi | has to stop pressed CMD-W :P |
| 10:28.04 | mattwj2002 | cmd w? |
| 10:28.17 | mattwj2002 | oh quit on a mac? |
| 10:28.25 | mattwj2002 | is that like Alt F4? |
| 10:28.59 | whizzi | alt-f4, that's on windows right? :P |
| 10:29.11 | whizzi | cmd-w = close window |
| 10:29.37 | boratynskikamil | Question. |
| 10:29.56 | boratynskikamil | Why, when I add line: callcounter=yes in SIP.conf, I cannot execute command pri show spans? |
| 10:29.59 | boratynskikamil | In example? |
| 10:30.09 | boratynskikamil | And, in general, DAHDI devices config losts... |
| 10:30.53 | whizzi | probably winkey-w or something on windows-based |
| 10:31.39 | mattwj2002 | right |
| 10:31.39 | mattwj2002 | same thing then |
| 10:31.39 | mattwj2002 | :) |
| 10:32.35 | mattwj2002 | brb |
| 10:54.55 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 10:55.22 | mattwj2002 | why couldn't mp3 support be built into asterisk? |
| 10:55.24 | mattwj2002 | :P |
| 10:55.39 | mattwj2002 | I am guessing it is the whole license thing |
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| 11:40.08 | WIMPy | I guess you can buy switchvox instead and the license is included. |
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| 12:16.36 | WIMPy | Is DUNDi supported by anything else than Asterisk? |
| 12:23.38 | *** join/#asterisk af_ (~af@static-82-85-142-162.clienti.tiscali.it) |
| 12:26.07 | bulkorok | DUNDi is a trademark of Digium, Inc. It was created by Mark Spencer. |
| 12:26.12 | bulkorok | so I guess not |
| 12:26.24 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 12:26.29 | v0lZy | Hi |
| 12:26.33 | v0lZy | I have a quick question |
| 12:26.49 | v0lZy | I have a trunk from my ISP which requires me to register |
| 12:26.53 | v0lZy | I can register and receive calls |
| 12:27.30 | v0lZy | Placing calls fails and the ISP says i need to send registration data with every call |
| 12:27.46 | v0lZy | Where in sip.conf or dial command or somewhere do I set this? |
| 12:27.46 | WIMPy | Same story for IAX, but that is supported elsewhere. |
| 12:27.51 | v0lZy | Hi WIMPy ! |
| 12:29.12 | WIMPy | In the peer. |
| 12:29.35 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 12:30.02 | v0lZy | WIMPy: In the peer. ... thats for me? |
| 12:30.29 | WIMPy | jepp |
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| 12:30.54 | v0lZy | wheres the 'peer' :d |
| 12:31.05 | MaliutaLap | I could use an example of that - it may be the cause for one of my ITSP issues |
| 12:31.15 | v0lZy | you mean type=peer on the Sip account for the phone, or type=peer on the sip trunk ? |
| 12:31.16 | MaliutaLap | can't make a fecking call via them |
| 12:31.41 | WIMPy | v0lZy: The credentials on that peer. |
| 12:31.57 | MaliutaLap | goes "ahhhhh" |
| 12:32.01 | v0lZy | WIMPy: what are the options ... auth= and secret= or something? |
| 12:32.07 | WIMPy | You probably have to use the defaultuser=. |
| 12:32.12 | v0lZy | i jsut put the credentials from the sip trunk on the peer? |
| 12:32.18 | v0lZy | ah .. with a . |
| 12:32.20 | WIMPy | . o O (Is that the current name of that parameter?) |
| 12:32.20 | v0lZy | ok |
| 12:32.42 | MaliutaLap | WIMPy: depends on what version you're using :) |
| 12:32.43 | WIMPy | What about the "."? |
| 12:33.19 | MaliutaLap | WIMPy: it forgot to have a / ? ;) |
| 12:33.27 | WIMPy | I know. But I always struggle to get the right names for the sip.conf stuff. |
| 12:34.09 | MaliutaLap | WIMPy: I think there's a list of paramater names somewhere ... I just use google to find them - and I read changelogs alot (I'm twisted like that) |
| 12:34.29 | WIMPy | Yes, the sip.conf.sample, off course. |
| 12:34.47 | MaliutaLap | WIMPy: didn't think that was exactly up to date |
| 12:34.58 | WIMPy | Not even that? |
| 12:35.00 | v0lZy | WIMPy: do i juse put defaultuser=myuser and thats it? or do i have to use the secret and username as well? |
| 12:35.27 | WIMPy | That is the username, and yes, you have to supply the secret as well. |
| 12:35.41 | WIMPy | )$^%*)"^ |
| 12:36.00 | WIMPy | How do I make Asterisk not think its IP is 127.0.0.1? |
| 12:36.03 | v0lZy | hm |
| 12:36.23 | MaliutaLap | v0lZy: check the voip-info entry on defaultusername - it will tell you what was deprecated in what versions |
| 12:37.27 | MaliutaLap | WIMPy: you mean that's not your servers address? and I've been hacking at it all day! ;) |
| 12:37.32 | bulkorok | WIMPy: udpbindaddr/tcpbindaddr |
| 12:38.11 | MaliutaLap | goes to bed before he gets too silly |
| 12:38.20 | wdoekes | v0lZy: probably fromuser= along with proper credentials in secret= |
| 12:39.22 | WIMPy | It is a valid address :-) but not the one I'd like it to annonce via DUNDi. |
| 12:40.36 | *** join/#asterisk mavrix (~kvirc@109.66.0.140) |
| 12:44.48 | boratynskikamil | Question. Is it possible to add queue member as pattern? |
| 12:48.05 | mavrix | I've installed Asterisk on a remote VPS that uses a shared IP |
| 12:48.31 | mavrix | ports 24000-2000 are forwarded to VPS |
| 12:48.44 | mavrix | sorry 24000-25000 |
| 12:49.17 | WIMPy | I've added res_stun_monitor, which is working, but it still uses 127.0.0.1 :-( |
| 12:49.38 | mavrix | in sip.conf [general] section I've set "bindport = 24001" - but still can't call SIP URI on my system on that port |
| 12:49.59 | mavrix | I guess I need to do additional settings about NAT sip.conf.. |
| 12:50.11 | mavrix | can someone instruct me what to do? |
| 12:51.00 | WIMPy | mavrix: You need additional configuration to get voice going. But the call itself should work. |
| 12:51.54 | mavrix | but call dowsn't go through at all. Nothing appears at log |
| 12:52.21 | mavrix | wonder what can be the reason |
| 12:52.29 | WIMPy | Then it probably didn't get to your server. |
| 12:53.27 | mavrix | the VPS provider gave me ports 24000-25000 |
| 12:53.37 | WIMPy | UDP or TCP? |
| 12:53.39 | mavrix | and at least for tcp web server it's working |
| 12:53.55 | WIMPy | That won't get you anywhere. |
| 12:54.22 | mavrix | you mean udp not enabled by default? |
| 12:54.41 | WIMPy | I'm not your vps provider. |
| 12:54.59 | mavrix | is ther any command I can check this from terminal? |
| 12:55.27 | WIMPy | You can only check that on the host, obviousely. |
| 12:56.06 | mavrix | I see |
| 12:56.22 | mavrix | will contact provider |
| 12:56.38 | mavrix | jsut want to make sure about the problem |
| 12:57.13 | mavrix | whether "bindport = 24001" only is enough for calls work |
| 12:58.48 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 12:58.52 | WIMPy | You also need to specify RTP ports in rtp.conf that are reachable. |
| 12:59.07 | WIMPy | Otherwise you get calls, but no audio. |
| 13:00.32 | mavrix | yea I'll do this later thanks |
| 13:00.46 | mavrix | after UDP works |
| 13:00.59 | WIMPy | It's all UDP |
| 13:01.06 | mavrix | sure |
| 13:01.23 | mavrix | bu the way when I call my DID numbers everything ok |
| 13:01.35 | mavrix | DIDs registered with asterisk |
| 13:05.51 | *** join/#asterisk traph (~traph@unaffiliated/traph) |
| 13:06.56 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 13:07.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:18.36 | v0lZy | WIMPy rocks |
| 13:18.40 | v0lZy | as do the rest here :) |
| 13:18.44 | v0lZy | thanks, I got it working! |
| 13:18.58 | boratynskikamil | Asterisk 1.8.25 |
| 13:19.16 | WIMPy | wonders if the 127.0.0.1 issue is caused by the fact that this host has IPv6 enabled? |
| 13:19.20 | boratynskikamil | exten => _HD-0X,hint,SIP/HD-0${EXTEN} - is it acceptable form for creating hints such as HD-01? |
| 13:19.37 | *** join/#asterisk zoid_ (~awainer@181.29.129.12) |
| 13:19.42 | WIMPy | Almost |
| 13:20.06 | WIMPy | ${EXTEN} is the whole extension, not just the wildcard part of it. |
| 13:21.15 | boratynskikamil | exten => _HD-0X,hint,SIP/${EXTEN} - is it acceptable form for creating hints such as HD-01? |
| 13:21.18 | boratynskikamil | Like that? |
| 13:21.35 | WIMPy | That should work. |
| 13:21.52 | boratynskikamil | Still nope. |
| 13:22.19 | boratynskikamil | <PROTECTED> |
| 13:22.25 | boratynskikamil | Something like that. |
| 13:22.43 | boratynskikamil | On core show hints |
| 13:22.49 | boratynskikamil | After dialplan reload |
| 13:23.05 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 13:23.42 | *** join/#asterisk ccesario (~ccesario@177.35.100.6) |
| 13:23.52 | WIMPy | I have no idea what that sould look like. I heve never tried pattens. |
| 13:24.35 | boratynskikamil | So is it good practice to define it manually? |
| 13:25.42 | WIMPy | Patterns should work. I just don't have extensions where they would work. |
| 13:26.59 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 13:27.35 | boratynskikamil | WIMPy: Like that: http://wklej.org/id/1282833/ it works properly. :-) |
| 13:31.48 | michael_work | sup guys and girls |
| 13:32.00 | michael_work | anyone ever used HANGUPCAUSE on 11+ ? |
| 13:32.03 | [TK]D-Fender | boratynskikamil: patterns work.... after you SUBSCRIBE to them |
| 13:32.27 | michael_work | i tried it with hanguphandpler push and in h regualr - no result |
| 13:32.33 | michael_work | do not get any datya :( |
| 13:32.39 | michael_work | keys are empty |
| 13:32.55 | boratynskikamil | [TK]D-Fender: subscibecontext? |
| 13:34.04 | [TK]D-Fender | boratynskikamil: No, and actual subscription |
| 13:34.07 | [TK]D-Fender | an* |
| 13:34.34 | boratynskikamil | [TK]D-Fender: subscription? Wiki link? |
| 13:36.24 | mirela666 | learning by doing |
| 13:36.44 | [TK]D-Fender | [08:32][TK]D-Fenderboratynskikamil: patterns work.... after you SUBSCRIBE to them <----------------- |
| 13:36.50 | michael_work | i don;'t see any point of regex in hints btw |
| 13:38.21 | [TK]D-Fender | michael_work: for large system with sloppy admins |
| 13:38.33 | [TK]D-Fender | michael_work: I can't see any real need... it just makes people lazy |
| 13:39.05 | WIMPy | Lazy is good |
| 13:39.39 | mirela666 | agree |
| 13:39.48 | mirela666 | :) |
| 13:40.01 | boratynskikamil | WIMPy: I am very lazy. :-) |
| 13:40.46 | michael_work | no i mean how it would work |
| 13:41.10 | michael_work | oh |
| 13:41.18 | michael_work | you use numeric extensions :) |
| 13:41.28 | michael_work | did not think about it :) |
| 13:41.56 | michael_work | never mind |
| 13:41.58 | michael_work | my bad |
| 13:42.20 | michael_work | it would not suit our use so i tried to thought how it can be used from our point of view :P |
| 13:42.41 | michael_work | anyway, does one here tried to use HANGUPCAUSE? |
| 13:42.50 | michael_work | or even SIP_CAUSE on 11+ |
| 13:43.00 | michael_work | it seems that it always return nothing |
| 13:43.04 | michael_work | returns* |
| 13:43.39 | *** join/#asterisk plm (~neo@189.2.146.50) |
| 13:43.43 | plm | Hi all |
| 13:44.02 | WIMPy | Normal HANGUPCAUSE certainly works. Not using SIP specific stuff. |
| 13:48.03 | *** join/#asterisk Papierkorb (~merettig@obelix.netzquadrat.de) |
| 13:48.50 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 13:50.52 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 13:59.50 | WIMPy | Ok, it looks like it had nothing to do with ipv6. |
| 14:00.19 | WIMPy | So does anyone have any idea why Asterisk would use 127.0.0.1? |
| 14:00.41 | WIMPy | I haven't had that issue before and this box isn't much different from the others. |
| 14:01.03 | WIMPy | The only thing google comes up with is that I'm not the first to experience that issue. |
| 14:01.28 | *** join/#asterisk Neozonz|Disc (~arajakul@unaffiliated/neozonz) |
| 14:01.52 | plm | Do you recommend some embedded board where I can add thats cards (analogue lines)? http://www.digium.com/en/products/telephony-cards/analog |
| 14:01.59 | Neozonz|Disc | what are the user/pass when you are authenticating against asterisk via ldap? |
| 14:05.34 | *** join/#asterisk aurs (~aurs@46.249.240.119) |
| 14:07.55 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:08.02 | michael_work | WIMPy, i need to know the sip code i got on Dial |
| 14:08.07 | Katty | hugs carrar |
| 14:08.16 | Katty | carrar: it's not so much monitoring software, as automation software. |
| 14:08.26 | [TK]D-Fender | plm: depends what you need |
| 14:08.30 | Katty | carrar: tho at this point, i'm not really sure what i want to automate. |
| 14:09.06 | michael_work | HANGUPCAUSE without parameters would return me the ast code which can be mapped to different SIP statuses |
| 14:09.09 | Katty | carrar: doing /something/ if the output of dahdi show status doesn't show what i want it to show |
| 14:09.31 | Katty | good morning, fender bender. |
| 14:10.53 | [TK]D-Fender | Katty: Mew. |
| 14:10.58 | plm | [TK]D-Fender: I want to do asterisk central (voip accounts + analogue lines to receive calls) |
| 14:11.20 | [TK]D-Fender | plm: How many lines? |
| 14:11.36 | plm | [TK]D-Fender: 2-20 lines |
| 14:11.49 | plm | [TK]D-Fender: depends of the client |
| 14:11.54 | [TK]D-Fender | plm: that is a very wide range and makes scaling hard |
| 14:12.06 | Katty | [TK]D-Fender: how're you this lovely morning, dear? |
| 14:12.10 | [TK]D-Fender | plm: If you're referring to MULTIPLE different clients it'll depend |
| 14:12.15 | [TK]D-Fender | Katty: Alive-ish |
| 14:12.41 | MaliutaLap | Katty! |
| 14:12.53 | plm | [TK]D-Fender: yes. ok. But for now think in just six lines |
| 14:13.09 | Katty | [TK]D-Fender: sounds like a good start |
| 14:13.14 | plm | [TK]D-Fender: or four lines |
| 14:13.15 | Katty | MaliutaLap: ohai! :> |
| 14:13.17 | Katty | hugs MaliutaLap |
| 14:13.26 | [TK]D-Fender | plm: I'm a fan of Sangoma's products. For 1-2 lines I'd go with their USBFXO device. for larger I'd go with B600d's |
| 14:13.38 | [TK]D-Fender | (3-4). |
| 14:13.58 | plm | [TK]D-Fender: and about digium cards? http://www.digium.com/en/products/telephony-cards/analog |
| 14:15.44 | [TK]D-Fender | plm: They'd be my second choice. There is no "third choice" IMO for internal cards. |
| 14:15.50 | *** join/#asterisk imox (~imox@p57A97199.dip0.t-ipconnect.de) |
| 14:16.36 | *** part/#asterisk Papierkorb (~merettig@obelix.netzquadrat.de) |
| 14:16.54 | plm | [TK]D-Fender: you are not fan of sangoma, but they are you first choice? |
| 14:19.22 | WIMPy | wonders where that "not" came from. |
| 14:19.57 | Katty | thinks WIMPy means knot. |
| 14:20.37 | WIMPy | ties a knot in to Kattys hair |
| 14:20.49 | Katty | :< |
| 14:21.06 | Katty | refills WIMPy's caffeine. |
| 14:21.13 | [TK]D-Fender | #readitagainsam |
| 14:21.27 | WIMPy | No, only one knot. That looked liek two :-) |
| 14:21.39 | WIMPy | can't work on caffeine. |
| 14:21.43 | MaliutaLap | I put the Digium at the top of the list |
| 14:21.44 | MaliutaLap | mine is still ticking 7 years on |
| 14:21.44 | MaliutaLap | WIMPy: I think it's ironic ... as in "Men do not like sex" ;) |
| 14:22.07 | Katty | WIMPy: does it cause anxiety? |
| 14:22.25 | WIMPy | No. Just a lack of concentration. |
| 14:22.36 | Neozonz|Disc | [TK]D-Fender, any good hints towards debugging 403 error on SIP registration? |
| 14:23.08 | Katty | WIMPy: ah yes, it certainly can do that. |
| 14:23.28 | plm | [TK]D-Fender: is a good idea to use 4 USBFXO sangoma devices to have 8 analogue lines? Thinking that the embedded board has not pcie slots. |
| 14:25.05 | [TK]D-Fender | plm: I'm thinking you shouldn't use an embedded board for PCI. and I have no idea if there are scaling issues with the USB solution. 4/8 I might consider a SIP gateway. |
| 14:25.06 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 14:25.30 | *** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 14:26.08 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 14:27.29 | *** join/#asterisk serafie (~erin@nat/digium/x-czxdpvijmaboqpof) |
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| 14:31.32 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:42.56 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 14:47.25 | *** join/#asterisk dlynes (~dlynes@d72-38-18-124.commercial1.cgocable.net) |
| 14:50.38 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
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| 14:59.19 | *** join/#asterisk MauriceM_ (~MauriceM_@71-38-129-62.ptld.qwest.net) |
| 15:01.25 | Neozonz|Disc | I get "No Matching Peers found" when using ldap |
| 15:01.34 | Neozonz|Disc | do I need to populate sip.conf with users from ldap? |
| 15:01.44 | Neozonz|Disc | i thought going though ldap was to avoid this |
| 15:01.45 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-wnxfdqohicawzlhf) |
| 15:01.45 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:02.01 | *** join/#asterisk mjordan (~matt@nat/digium/x-quowkngprsxjpznl) |
| 15:02.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:06.50 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:08.21 | *** join/#asterisk kruftindustries (4016eaec@gateway/web/freenode/ip.64.22.234.236) |
| 15:08.29 | kruftindustries | Hello! |
| 15:09.06 | kruftindustries | This is my first time in the asterisk irc channel |
| 15:09.25 | kruftindustries | I'm not sure how active this channel is, so I'll get to it |
| 15:10.08 | Chainsaw | Very active. |
| 15:10.12 | kruftindustries | I need to dial an extension with AGI and connect it to an ongoing channel |
| 15:10.32 | kruftindustries | Listen only, like Fop2 does |
| 15:10.51 | Psi|4ward | Hi there, updated from 1.6 to 1.8 but getting chan_sip.c:20354 handle_response_invite: Failed to authenticate on INVITE |
| 15:10.53 | Psi|4ward | is there anything wrong with Dial(SIP/${DENG}@pbx&SIP/${GIGASET}@pbx,10); |
| 15:13.20 | *** join/#asterisk navaismo (~navaismo@189.146.33.56) |
| 15:14.39 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:14.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:14.41 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 15:16.38 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:21.47 | kruftindustries | Psi|4ward: I'm looking in my dialplan... That looks like it should work |
| 15:22.21 | kruftindustries | Also, I have noticed when updating, it takes some poking around to get the password for your extensions to work |
| 15:22.44 | kruftindustries | maybe amportal reload or something |
| 15:22.53 | kruftindustries | a few times |
| 15:24.14 | kruftindustries | If you know anything about AGI, I could use a tip or two, as the documentation is out there, though an example is somewhat scarce unless base64 encoded and avaliable with a license |
| 15:24.24 | SpeedEvil | Apologies for the silly question. What is the proper term for 'a box I plug into the phone socket which I can then connect to with a SIP phone'. There seem to be eleven billion of the opposite on ebay. Just a single line. |
| 15:25.36 | kruftindustries | external modem perhaps? |
| 15:26.28 | kruftindustries | FXO Gateway? |
| 15:27.26 | navaismo | ATA? |
| 15:27.31 | navaismo | Bid everyone |
| 15:27.38 | kruftindustries | Yeah, that one ^ |
| 15:27.50 | kruftindustries | Oh wait, he wants the opposite |
| 15:27.52 | SpeedEvil | Not a modem - I don't want external VOIP. My internal phone 'system' (crappy DECT phone) has died, and I want to replace it with an internal SIP system - using old phones with SIP clients on them. |
| 15:28.17 | kruftindustries | Aha, yeah ATA |
| 15:28.20 | Chainsaw | endorses navaismo's answer |
| 15:28.48 | SpeedEvil | All ATA thingies I seem to be finding are the opposite |
| 15:29.22 | *** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 15:29.23 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 15:29.23 | navaismo | pap2t |
| 15:29.56 | kruftindustries | If you do go the ebay route, do some searching first, I got a Linksys SPA3102 that had been jtagged, wouldn't work on the local network, external only |
| 15:31.12 | kruftindustries | Or well one of those linksys ATA boxes that said "Unlocked" |
| 15:31.16 | SpeedEvil | http://www.ebay.co.uk/itm/Cisco-SPA3102-Voice-Gateway-with-Router-SPA3102-UK-/131117298172?pt=UK_Computing_Wireless_Routers&hash=item1e873331fc does indeed seem relevant |
| 15:32.22 | SpeedEvil | thanks |
| 15:32.37 | SpeedEvil | probably shouldn't try this today - head is feeling packed full of wool. |
| 15:32.50 | navaismo | Damm it lost the bidding |
| 15:40.55 | pabelanger | Asked this yesterday didn't get much feedback, anybody else using VXML with asterisk? Trying to help sliske_ come up with a solutions to creating some dynamic menus in asterisk |
| 15:43.15 | whizzi | Does anyone has an experience that when a DNS-server is slow, that Asterisk (11) starts disconnecting it's peers and users? |
| 15:43.30 | Chainsaw | whizzi: Yes, it will not cope with that. |
| 15:43.30 | kruftindustries | o.0 how slow is it? |
| 15:43.38 | Chainsaw | whizzi: Host a local BIND instance. |
| 15:44.14 | whizzi | kruftindustries: 100ms reply time |
| 15:44.20 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 15:44.49 | whizzi | Chainsaw: that'd be the best, I agree. |
| 15:45.36 | whizzi | Chainsaw: but even external peers (like @1.2.3.4 for example) get disconnected because the DNS-reply is a little slow |
| 15:45.44 | Chainsaw | whizzi: You're probably experiencing this one: https://issues.asterisk.org/jira/browse/ASTERISK-21378 |
| 15:45.49 | MaliutaLap | I always put a bind instance on my networks |
| 15:46.55 | Chainsaw | whizzi: DNS replies must be instantaneous or Asterisk will have a terrible time. Lack of response on one peer will hold up another. |
| 15:47.21 | plm | [TK]D-Fender: sorry for the long time. But in box with asterisk and four USBFXO I can to do a SIP gateway right? So, that box allows calls to and from analogue lines and via voip. |
| 15:47.58 | [TK]D-Fender | plm: No, I'm recommending a SIP-FXO gateway INSTEAD of the USB one(s) |
| 15:49.21 | Chainsaw | whizzi: Word has it that Asterisk 12 has a different enough design that it won't suffer from this. But I'm not at a point where I'm happy to recommend 12 for general use. So yes, local BIND instance please. And cross a few fingers. |
| 15:49.43 | *** part/#asterisk Psi|4ward (~psi@p3E9C2657.dip0.t-ipconnect.de) |
| 15:51.26 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:53.30 | whizzi | Chainsaw: Ok, clear :). I'll make a local caching DNS then :) |
| 15:53.35 | whizzi | Chainsaw: thanks :) |
| 15:54.24 | kruftindustries | Anyone have any idea about AGI and connecting listen only to an ongoing call? |
| 15:55.27 | WIMPy | kruftindustries: Where's the link? |
| 15:55.40 | file | to be specific - the new SIP channel driver won't suffer from it |
| 15:56.17 | plm | hmm |
| 15:57.31 | kruftindustries | So I used AGI Originate and have a dialplan that it goes to, and eventually dials echotest |
| 15:57.51 | Chainsaw | whizzi: You're welcome :) |
| 15:57.51 | plm | [TK]D-Fender: but what is the difference between use a SIP-FXO gateway and USB one(s)? |
| 15:58.25 | kruftindustries | Let's say while we're in echotest, I need to have another extension ring with only audio of the first call |
| 15:58.28 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 15:59.11 | kruftindustries | [TK]D-Fender: The sip gateway connects to your server over ethernet, the usb one is basically a headset |
| 15:59.19 | kruftindustries | *headset adapter |
| 15:59.33 | kruftindustries | From my experience, at least |
| 16:00.35 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 16:01.31 | [TK]D-Fender | [10:59]plm[TK]D-Fender: but what is the difference between use a SIP-FXO gateway and USB one(s)? <- maybe USB isn't so reliably past a device or two. |
| 16:01.41 | [TK]D-Fender | plm: Or you run out of ports |
| 16:02.13 | [TK]D-Fender | [10:56]kruftindustriesAnyone have any idea about AGI and connecting listen only to an ongoing call? <- you don't need AGI for this. Chanspy is a single dialplan app. |
| 16:02.14 | plm | kruftindustries: but in the sip gateway has many FXO ports and the ethernet right? So if in a box I put asterisk + four USB FXO, so I have a sip gateway.. |
| 16:02.26 | plm | ohh.. reading =D |
| 16:02.42 | [TK]D-Fender | plm: SIP gateway INSTEAD OF the USB apaters\ |
| 16:03.33 | kruftindustries | [TK]D-Fender: I need to do it with php when someone clicks a button |
| 16:03.42 | MaliutaLap | plm: yeah, that reading thing is kinda important :) |
| 16:03.51 | plm | [TK]D-Fender: So, in other words, can I to do my own SIP gateway using asterisk and FXO devices? Or I need to buy a ready appliance SIP gateway? |
| 16:03.59 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:04.25 | kruftindustries | I 2nd the sip gateway instead of adapter thing :D |
| 16:04.41 | [TK]D-Fender | plm.... |
| 16:04.49 | [TK]D-Fender | plm: Try to follow along. |
| 16:05.07 | MaliutaLap | gets the popcorn out |
| 16:05.09 | [TK]D-Fender | plm: You have your embedded compact little Asterisk server. |
| 16:05.10 | plm | ok |
| 16:05.23 | [TK]D-Fender | plm: INSTEAD of getting a pile of USB to FXO dongles.... |
| 16:05.48 | [TK]D-Fender | plm: You get a SIP to FXO gatway and tell Asterisk to send calls to/from THAT INSTEAD |
| 16:06.05 | plm | hmm ok [TK]D-Fender |
| 16:06.22 | [TK]D-Fender | plm Plug lines into gateway box. Plug gateway box to your NETWORK. Tell your server to use it for calls |
| 16:06.55 | plm | [TK]D-Fender: could you recommend me a good SIP gateway? Just for one example.. |
| 16:07.17 | [TK]D-Fender | Media5, AudioCodes |
| 16:07.23 | MaliutaLap | [TK]D-Fender: are you feeling alright? that was not upto your usual cluebat usage ;) |
| 16:07.42 | plm | ok |
| 16:08.20 | MaliutaLap | I should know I have [rightfully] been on the receiving end of a [TK]D-Fender handled cluebat |
| 16:09.43 | plm | [TK]D-Fender: and if I would like to use the analogue devices and the many VOIP accounts, not problem? |
| 16:10.37 | plm | [TK]D-Fender: So people, will just to do call using voip (voip operator), but when receive call will receive via analogue lines (sip gateway) |
| 16:12.36 | [TK]D-Fender | plm: VoIP accounts is irrelevant. |
| 16:13.04 | [TK]D-Fender | plm: using analog phones (FXS), I'd stick with ATA's or larger gateways as needed |
| 16:15.31 | MaliutaLap | plm: http://www.voip-info.org/wiki/view/FXO |
| 16:15.33 | plm | [TK]D-Fender: I still needs the analogue lines becouse Voip Operator don't have a local number, so VOIP accounts is used just to do calls, not to receive. And that is why I'm looking for to use analogue lines, just to receive that call for a local number. See? |
| 16:15.59 | MaliutaLap | plm: read that link, you might learn the difference between FXO and FXS |
| 16:16.13 | plm | MaliutaLap: ok |
| 16:18.08 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:19.49 | plm | MaliutaLap: [TK]D-Fender ohh right. I will use just IP phones. So I will not need the FXS. |
| 16:19.57 | [TK]D-Fender | [11:15]plm[TK]D-Fender: I still needs the analogue lines becouse Voip Operator don't have a local number, so VOIP accounts is used just to do calls, not to receive. And that is why I'm looking for to use analogue lines, just to receive that call for a local number. See? <- I've been suggesting dives for your analog lines for who hours now... |
| 16:20.10 | [TK]D-Fender | two* |
| 16:20.16 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 16:20.22 | plm | [TK]D-Fender: sorry, so sorry |
| 16:20.31 | [TK]D-Fender | devices* |
| 16:21.31 | plm | [TK]D-Fender: I can just to use entire voip, so will be all phones ip phones. So that FXO will need just to have a local number (receive calls) and maybe to do calss when VOIP operator is down. |
| 16:21.49 | plm | *calls |
| 16:22.01 | plm | [TK]D-Fender: I was clear now? =D |
| 16:22.46 | *** part/#asterisk whizzi (~whizzi@82-171-3-8.ip.telfort.nl) |
| 16:23.26 | MaliutaLap | plm: how many incoming PSTN lines are we talking? |
| 16:23.48 | MaliutaLap | plm: and are they standard pairs or ISDN? |
| 16:24.24 | *** join/#asterisk salz212 (~chatzilla@li345-39.members.linode.com) |
| 16:24.40 | plm | MaliutaLap: I was thinking between 2-20 lines incomming PSTN. the FXO =D |
| 16:24.45 | plm | MaliutaLap: depends of client. |
| 16:24.56 | plm | But think just four or six for now |
| 16:25.03 | plm | MaliutaLap: are standard pairs |
| 16:25.34 | *** join/#asterisk aness (~aness@cm-84.215.76.18.getinternet.no) |
| 16:26.32 | MaliutaLap | plm: uh for 20 you'd want to go with ISDN |
| 16:26.56 | WIMPy | 'd say that for >1 |
| 16:27.03 | plm | MaliutaLap: yes. but shtink not more than 8 for now |
| 16:27.06 | MaliutaLap | plm: and for 2-10 standard pairs I'd look at the digium range of internal cards |
| 16:27.30 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:27.38 | MaliutaLap | that's still alot of pairs coming in - unless you can't get ISDN in that location |
| 16:27.52 | wonderworld | plm: i wouldn't buy hardware cards anymore. i'd reccomend a patton media gateway |
| 16:27.56 | MaliutaLap | WIMPy: my house has 3 lots of standard pairs coming in |
| 16:28.16 | plm | MaliutaLap: like this? http://www.digium.com/en/products/telephony-cards/analog/8-port |
| 16:28.26 | MaliutaLap | plm: exactly |
| 16:28.48 | MaliutaLap | plm: if you only need the FXO - the that's how you populate it |
| 16:28.54 | plm | MaliutaLap: that 8 port show is a mix FXO and FXS, can I to use all 8 just FXO (incomming)? |
| 16:29.31 | plm | MaliutaLap: yes, not will be analogue phones, will just ip fones, it easy to deploy |
| 16:29.44 | plm | wonderworld: patton media gateway? |
| 16:30.05 | MaliutaLap | plm: you can choose what modules you populate it with - so you can have all 8 ports as FXO |
| 16:30.26 | MaliutaLap | plm: then it sits in the box and is nicely dealt with by DAHDI |
| 16:30.54 | WIMPy | hasn't found anything nice about DAHDI so far. |
| 16:31.09 | MaliutaLap | plm: I have one of the older models - a TDM400, it's a 4 port card that I have populated with 2FXO and 2FXS |
| 16:31.28 | MaliutaLap | WIMPy: shuddup :P Works just fine |
| 16:31.32 | plm | MaliutaLap: well, for two incomming calls (FXO) I can use just the USBFXO of samgoma.. |
| 16:31.40 | MaliutaLap | WIMPy: maybe your pstn just blows |
| 16:31.49 | WIMPy | MaliutaLap: So you will tell me how to make it work? |
| 16:32.30 | wonderworld | plm: "patton smartnode" they are nice black boxes. connect FXO, get SIP. no drivers, no nothing. |
| 16:32.36 | MaliutaLap | plm: I personally don't trust USB for much ... so I wouldn't recommend it |
| 16:33.16 | MaliutaLap | WIMPy: you can't make DAHDI work? I find that hard to believe |
| 16:33.23 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:34.03 | WIMPy | MaliutaLap: I failed to make it work yesterday. And for the part that works, it doesn't work in a way I like. |
| 16:34.27 | wonderworld | DAHDI + EuroISDN used to be a catastrophy. maybe it improved, but i havent tested for a long time. |
| 16:34.50 | Chainsaw | wonderworld: The Patton gear is very reliable, yes. Hell on earth to configure, especially first time. |
| 16:35.01 | WIMPy | wonderworld: It has improved massively. Still the configuration is rather troublesome. |
| 16:35.17 | plm | wonderworld: just to connect of "patton smartnode" to my embedded board running asterisk and done? |
| 16:35.35 | wonderworld | Chainsaw: but..if it works, it'll work forever. i sold several of them. never had a single support case. |
| 16:35.40 | MaliutaLap | WIMPy: Yeah, I'm in Australia things worked as described |
| 16:35.44 | Chainsaw | wonderworld: Agreed. |
| 16:35.53 | wonderworld | plm: FXO ---> Patton --> LAN --> Asterisk |
| 16:35.54 | MaliutaLap | WIMPy: right down to distinctive rings |
| 16:36.29 | plm | MaliutaLap: the advantage of that USBFXO is just cheaper for just two FXO lines.. |
| 16:37.06 | WIMPy | I have a TDM410 that works, but the handling of held calls is really bad. Just connecting you to another call if the active one goes away without any indication is evil. |
| 16:37.25 | plm | wonderworld: ok. I think to use asterisk in the beagle bone black. As asterisk will encoding/deconding anything, I think will run fine, what you think? |
| 16:37.47 | MaliutaLap | plm: I'm not in charge of the budget ... it's what you can sell the client on. I tend to push for things that are more reliable (people have tendency to play with USB devices), mid range kit is good |
| 16:38.03 | WIMPy | And on the B410 I can't dial for whatever reason. Unfortunatly the Linux drivers don't like it, either. |
| 16:38.26 | MaliutaLap | WIMPy: I don't have that issue |
| 16:39.04 | MaliutaLap | WIMPy: my held calls on the TDM work better than on the Cisco [SIP] handset |
| 16:39.15 | MaliutaLap | 'sif use skinny or mgcp ;) |
| 16:39.26 | WIMPy | I have had DAHDi working with a standard 8 BRI card in my test box some longer time ago. so I hope it's still possible somehow. |
| 16:40.45 | WIMPy | Yesterday I didn't get dialtone, but the call showed up in the *CLI. Interestingly enough it went to simple switch, even though I have immediate=yes. |
| 16:41.05 | MaliutaLap | I did have issues transitioning between zaptel and dahdi ... but that was mostly echo cancelation stuff |
| 16:41.10 | WIMPy | Then after a short delay it goes to s (WaitExten), but at no point does it react to user input. |
| 16:41.34 | WIMPy | Calls to the phone do work, however. |
| 16:41.39 | plm | MaliutaLap: yes. you are right. Wee needs more reliable. I will use the suggestion of wonderworld and use a smartnode with my box =D |
| 16:41.46 | *** join/#asterisk koffel (koffel@173-167-212-107-ip-static.hfc.comcastbusiness.net) |
| 16:41.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:41.55 | koffel | hey all |
| 16:42.11 | MaliutaLap | WIMPy: strange |
| 16:42.22 | MaliutaLap | is trying bed again |
| 16:42.57 | WIMPy | didn't have any DAHDI encounters without strange effecrs so far |
| 16:42.58 | koffel | i have a polycom ip 320 phone and i get it to register 2 lines but one line dont work |
| 16:43.09 | WIMPy | dahdi_genconf didn't do, either. |
| 16:43.16 | koffel | <PROTECTED> |
| 16:43.19 | WIMPy | But that's less important. |
| 16:45.11 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:49.26 | plm | wonderworld: could you recommend me a model of smartnode? |
| 16:55.41 | WIMPy | BTW: There's no shortcut to dial out multiple DAHDI interfaces, is there? |
| 17:00.32 | ChannelZ | like simultaneously? |
| 17:00.44 | ChannelZ | besides DAHDI/1&DAHDI/2&DAHDI/3 ... |
| 17:00.49 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 17:01.08 | WIMPy | yes |
| 17:02.28 | [TK]D-Fender | WIMPy: of course there is... group them |
| 17:02.29 | [TK]D-Fender | Simultaneous = & <- |
| 17:03.49 | [TK]D-Fender | like everything else in Dial() |
| 17:03.51 | WIMPy | Yes, but I need to supply a list of channels and groups, each on a single channel. So it might make sense to have a short cut for that. I just wanted to make sure if such a thing exists. |
| 17:05.15 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 17:05.28 | [TK]D-Fender | nope, simultaneous is a Dial() thing. |
| 17:06.07 | [TK]D-Fender | Can make for some really messy chains... |
| 17:06.07 | [TK]D-Fender | (looking) |
| 17:06.49 | WIMPy | Indeed. That's why I modified chan_lcr to be able to supply a list in a single channel. |
| 17:18.02 | *** join/#asterisk yano (~yano@freenode/staff/yano) |
| 17:18.24 | *** join/#asterisk ZX81 (~ZX81@184.91.194.247) |
| 17:18.40 | ZX81 | hi all, trying to run two asterisk's on one machine - and coming across a strange issue |
| 17:18.47 | ZX81 | asterisk -C /etc/asterisk2/asterisk.conf |
| 17:18.47 | ZX81 | Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. |
| 17:18.59 | WIMPy | Not surprising. |
| 17:19.07 | ZX81 | whereas /etc/asterisk2/asterisk.conf contains: astrundir => /var/run/asterisk2 |
| 17:19.17 | WIMPy | Configure different directories and ports. |
| 17:19.26 | ZX81 | yep thats the problem |
| 17:19.34 | ZX81 | it's ignoring the directory directive :-) |
| 17:19.48 | WIMPy | Is it reading the right asterisk.conf? |
| 17:19.56 | ZX81 | asterisk -C /etc/asterisk2/asterisk.conf |
| 17:20.07 | ZX81 | I mean how would I know if I ask it to? |
| 17:21.27 | ZX81 | yep |
| 17:21.27 | ZX81 | stat("/etc/asterisk2/asterisk.conf", {st_mode=S_IFREG|0644, st_size=3457, ...}) = 0 |
| 17:21.53 | ZX81 | lol |
| 17:21.57 | ZX81 | but then does mkdir("/var/run/asterisk", 0755) = -1 EEXIST (File exists) |
| 17:21.59 | WIMPy | Looks like someone isn't playing nicely. |
| 17:22.05 | ZX81 | agreed :-) |
| 17:22.31 | [TK]D-Fender | Pretty good guess why..... |
| 17:22.36 | ZX81 | yeah? |
| 17:22.40 | [TK]D-Fender | pastbin your asterisk.conf's |
| 17:22.59 | ZX81 | all or head? |
| 17:23.43 | ZX81 | http://pastebin.com/RBimEAqE |
| 17:23.44 | ZX81 | and |
| 17:24.17 | ZX81 | http://pastebin.com/RXTHW7HD |
| 17:24.22 | ZX81 | first is asterisk |
| 17:24.24 | ZX81 | second is asterisk2 |
| 17:24.32 | ZX81 | they'll both be doing identical things |
| 17:25.47 | [TK]D-Fender | (!) <----- FAIL |
| 17:25.56 | ZX81 | ok |
| 17:25.57 | [TK]D-Fender | that IGNORES the settings below |
| 17:26.20 | WIMPy | Where does that come from? |
| 17:26.33 | [TK]D-Fender | comes included that way in the sample config |
| 17:26.47 | WIMPy | evil |
| 17:26.50 | ZX81 | <33>:cat /usr/src/asterisk/configs/asterisk.conf.sample |
| 17:26.51 | ZX81 | [directories](!) |
| 17:32.48 | Neozonz|Disc | THis is so bloody frustrating |
| 17:32.53 | Neozonz|Disc | is ldap broken in asterisk? |
| 17:33.27 | Neozonz|Disc | I keep getting Could not find any entry matching (&(objectClass=AsteriskExtension)(AstExtension=_*)(AstContext=default)(AstPriority=1)) in base dn |
| 17:33.32 | WIMPy | doesn't know about LDAP, but I can confirm the frustrating bit. |
| 17:33.37 | Neozonz|Disc | but looking at ldap its right bloody there! |
| 17:35.45 | Neozonz|Disc | are there any issues with asterisk 1.8? |
| 17:37.22 | Neozonz|Disc | IMPORTANT NOTICE regarding res_ldap.conf: DON'T use quotes (") for basedn=... and user=... configuration entries. Otherwise asterisk will include the quotes in DN which results in "Invalid DN syntax" error messages. I haven't check the rest of the config file, but this is definitely an error in that (using asterisk 1.6.1.9). |
| 17:37.46 | Neozonz|Disc | does this still exist in 1.8? |
| 17:37.58 | [TK]D-Fender | check the tracker. |
| 17:38.10 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:38.20 | Neozonz|Disc | nm |
| 17:38.26 | Neozonz|Disc | doesnt matter i didnt set them either.. |
| 17:38.50 | Neozonz|Disc | can someone plz confirm if ldap works on 1.8... |
| 17:39.03 | Neozonz|Disc | this is driving me nuts and i'm sure theres nothing wrong with config... |
| 17:39.12 | Neozonz|Disc | as it connects to ldap and is able to get users uid and cn |
| 17:39.15 | file | I don't know of anybody who uses that module. |
| 17:39.17 | [TK]D-Fender | sure != checked and vetted |
| 17:39.20 | WIMPy | You'll get used to that feeling. |
| 17:39.51 | Neozonz|Disc | i've checked and vetted |
| 17:39.57 | Neozonz|Disc | for the last 2 days |
| 17:40.01 | [TK]D-Fender | > 50% of the time "sure" people end up being wrong and feeling really bad |
| 17:40.19 | Neozonz|Disc | file, is there an alternative for ldap integration |
| 17:45.36 | file | LDAP is not something I know |
| 17:46.16 | *** join/#asterisk polysics (~Adium@95.236.70.3) |
| 17:46.19 | polysics | hello |
| 17:46.39 | polysics | I have allowguest on but Asterisk is giving 401s to unauthed callers |
| 17:47.00 | polysics | the console say nothing, I just see the 401 |
| 17:47.10 | [TK]D-Fender | I'm sure it says something.... |
| 17:47.23 | polysics | literally nothing, which is weird |
| 17:47.47 | WIMPy | Are you sure you're calling your Asterisk? |
| 17:47.49 | [TK]D-Fender | Doubt it highly... |
| 17:48.01 | [TK]D-Fender | You can't have "literally nothing" and "401" |
| 17:50.46 | polysics | here's all I see |
| 17:50.47 | polysics | https://gist.github.com/polysics/f6d3d21282a7c3da93d2 |
| 17:51.40 | [TK]D-Fender | That is absolutely a "something" .... something you should be reading even a little bit closely.... |
| 17:51.49 | WIMPy | Found peer 'usera' |
| 17:51.52 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 17:51.55 | [TK]D-Fender | NOT ANONYMOUS |
| 17:57.31 | MaliutaLap | [TK]D-Fender: I love you ... when I'm having trouble sleeping you provide me with entertainment. I love the cluebattery :) |
| 17:57.47 | polysics | I got what's wrong, thanks :) |
| 18:02.06 | [TK]D-Fender | MaliutaLap: More like one of those American Gladiators padded pole-arms.... |
| 18:02.12 | *** join/#asterisk danjenkins_ (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 18:02.45 | MaliutaLap | [TK]D-Fender: you have a padded pole? TMI ;P |
| 18:03.03 | [TK]D-Fender | MaliutaLap: I'd taken the bite out of the old ClueBat (tm) I used to wield (including its backup rusty-nail enhanced brother) |
| 18:03.15 | MaliutaLap | I prefer ballistic cluebats - they don't let me carry them :) |
| 18:03.37 | [TK]D-Fender | Ballistic doesn't require carry, just launch :) |
| 18:04.50 | MaliutaLap | [TK]D-Fender: my female DNA (which is native american) would like to have your babies ... the other 2 strands are conflicted |
| 18:06.38 | MaliutaLap | [TK]D-Fender: not when it's a Barret |
| 18:06.53 | *** join/#asterisk Neozonz (~arajakul@unaffiliated/neozonz) |
| 18:07.09 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 18:07.18 | MaliutaLap | I like 22SAS kit :) |
| 18:07.46 | MaliutaLap | [TK]D-Fender: https://en.wikipedia.org/wiki/Barrett_M82 - my cluebat of choice :) |
| 18:11.36 | hrnt | so... can i trust sip show channelstats? specifically the packet loss numbers. |
| 18:12.56 | hrnt | (i am using a linksys SPA942 for testing right now) |
| 18:20.45 | [TK]D-Fender | What tribe? |
| 18:21.14 | MaliutaLap | [TK]D-Fender: they won't give me that kind of info on my donor |
| 18:21.34 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 18:23.16 | [TK]D-Fender | MaliutaLap: a classic.... |
| 18:23.25 | *** join/#asterisk ThatCantBe (~irc@dsl.dyn-206.53.180.155.tbinet.bm) |
| 18:23.49 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 18:23.51 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 18:24.08 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 18:24.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:26.03 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 18:35.58 | *** join/#asterisk salz212 (~chatzilla@li345-39.members.linode.com) |
| 18:40.13 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:40.25 | salz212 | Asterisk-Box 1 and 2 have same extensions.. when asterisk box 1 sends call through trunk to Asterisk box 2 it sends 401 and start authorizing and waiting for reply with credentials, which should not be the case as it is not a registered caller just the same extension but from diff domain. |
| 18:40.43 | salz212 | I remember resolving it before but seems kind of stuck |
| 18:44.09 | hrnt | what kind of user did you register the other asterisk box at? (peer, user, friend etc.) |
| 18:44.56 | salz212 | friend |
| 18:45.06 | salz212 | both sides friend |
| 18:45.48 | hrnt | why not peer? |
| 18:46.18 | salz212 | i'll be doing both receive and send traffic through it. |
| 18:46.31 | hrnt | yes, peer allows that |
| 18:46.51 | salz212 | the issue is why Asterisk takes incoming calls callerid and thinks its the one which is registered |
| 18:48.56 | MaliutaLap | canreinvite? |
| 18:49.48 | hrnt | salz212: do you know what "friend" does? |
| 18:51.23 | MaliutaLap | hrnt: only if it has benefits ;) |
| 18:51.27 | salz212 | matches ip port |
| 18:51.33 | hrnt | that is peer |
| 18:52.31 | hrnt | salz212: friend is peer+user. |
| 18:52.40 | hrnt | salz212: now your asterisk is trying to authenticate the other asterisk box as "user" |
| 18:52.43 | salz212 | yes i know that |
| 18:53.07 | salz212 | but if you have insecure invite or port then it doesnt |
| 18:54.13 | hrnt | uhm, you have type=friend and insecure=invit,port? |
| 18:54.14 | hrnt | invite |
| 18:54.54 | salz212 | yes |
| 18:55.07 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:55.17 | hrnt | for the same entry? |
| 18:56.18 | hrnt | and for the other asterisk box? |
| 18:56.36 | salz212 | same |
| 18:57.37 | salz212 | "When setting up trunks, make sure there's no risk that any From: username (caller ID) will match any of your device names, because then Asterisk might match the wrong device" -- SIP.conf default :S |
| 19:01.58 | hrnt | anyways, regarding type=friend. somebody here should confirm, but as far as i know that is a pretty dangerous thing to do together with insecure=invite,port. |
| 19:02.01 | Neozonz | Someone please tell me they have LDAP working with asterisk |
| 19:02.45 | MaliutaLap | Neozonz: there are people that do have it working - not an integration I have tried yet |
| 19:03.04 | Neozonz | is there a bug with a certain version of asterisk? |
| 19:03.20 | Neozonz | I'm having issues with grabbing the extentions live |
| 19:03.24 | Neozonz | but everything else works.. |
| 19:03.28 | Neozonz | so frustrating |
| 19:03.55 | *** join/#asterisk u0m3 (~u0m3@92.80.86.213) |
| 19:04.12 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 19:05.07 | MaliutaLap | Neozonz: we call them "features" ;) |
| 19:07.22 | Neozonz | ? |
| 19:07.48 | MaliutaLap | Neozonz: not "bugs" ... "features" ;) |
| 19:07.57 | salz212 | hrnt: its for testing but the main thing is .. that if the callerid is same as a sip registered user... everything gets messed up |
| 19:12.02 | hrnt | salz212: you could try adding deny=ip-address-of-your-trunk to your extension configuration, but i am not sure if it works |
| 19:12.35 | hrnt | best case: deny means that it only ignores this particular channel and continues looking for other matching channels. worst case: it immediately rejects the invite when it seems the deny. |
| 19:15.37 | *** join/#asterisk scouture (~scouture@unaffiliated/scouture) |
| 19:17.51 | salz212 | actually the problem is "Incoming call callerid from outside ..matching with SIP registered user on the Asterisk box receiving the call" .. |
| 19:20.03 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
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| 19:39.56 | salz212 | been searching on internet for this issue ..anyone any idea? |
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| 19:47.07 | salz212 | just for info ..it was type issue for user not trunk. thanks |
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| 19:51.26 | Neozonz | how do I add ldap to asterisk (via the install menu) |
| 19:51.45 | [TK]D-Fender | What "install menu"? |
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| 20:41.36 | lanning | you know... the "install" menu... (make install) :) |
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| 20:45.09 | [TK]D-Fender | there is no "option" there.... whatever was actually compiled in is what will install |
| 20:50.05 | Neozonz | how do i get asterisk to show debug on the console? |
| 20:50.20 | Neozonz | i set it to debug 15/verbose 15 |
| 20:50.23 | Neozonz | still nothing |
| 20:52.57 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:56.11 | navaismo | Neozonz, edit logger.conf add debug to the console line |
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| 20:58.13 | navaismo | then run logger reload in the cli |
| 21:00.49 | Neozonz | <navaismo> then run logger reload in the cli |
| 21:01.06 | navaismo | ? |
| 21:01.10 | Neozonz | sorry |
| 21:01.11 | Neozonz | wrong paste |
| 21:01.31 | Neozonz | http://prntscr.com/2vttf8 |
| 21:01.40 | Neozonz | set it, but nothing on cli |
| 21:02.29 | navaismo | core set verbose 10 then core set debug 10 |
| 21:02.50 | Neozonz | yeah set them already |
| 21:02.53 | Neozonz | still nothing :( |
| 21:03.31 | navaismo | if you exit the cli and you didnt have those values in the asterisk.conf then you need to run those commands every time you enter in the cli |
| 21:03.48 | Neozonz | i set it on the asterisk.conf already |
| 21:03.55 | Neozonz | soo... they're automatically set on run |
| 21:04.15 | navaismo | thats the reason of my 'IF' |
| 21:05.06 | navaismo | the main console ctrl+alt+F9 still dont show anything? |
| 21:05.49 | Neozonz | i'm ssh'd in |
| 21:05.54 | Neozonz | whats ctrl+alt+f9 do? |
| 21:06.42 | navaismo | show the tty running asterisk |
| 21:06.52 | navaismo | if is enabled in the asterisk.conf |
| 21:07.09 | Neozonz | ? |
| 21:13.33 | Neozonz | when i do a reload |
| 21:13.40 | Neozonz | i see all the debug stuff |
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| 22:11.48 | Neozonz | what does this mean |
| 22:11.50 | Neozonz | <PROTECTED> |
| 22:18.29 | dwayne | maybe you have a priority for that extension without specifying an application ? |
| 22:19.12 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:26.09 | Neozonz | weird.. |
| 22:26.15 | Neozonz | now when i call someone it just crashes |
| 22:26.21 | Neozonz | -_- |
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| 23:18.34 | mavrix | hi |
| 23:18.55 | mavrix | can someone guide me how to deal with NAT? |
| 23:19.29 | mavrix | my remote asterisk operates on non standard port |
| 23:19.46 | mavrix | when i call in i get no audio |
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| 23:50.57 | SpeedEvil | Is there a generally accepted solution for plugging a phone line into an asterisk box over USB so it can accept/make calls over USB? |
| 23:52.01 | WIMPy | yes |
| 23:52.09 | WIMPy | What kind of phone line? |
| 23:52.24 | SpeedEvil | Specifically a UK phone line. |
| 23:52.48 | WIMPy | What kind of phone line? |
| 23:53.00 | SpeedEvil | POTS |
| 23:53.18 | WIMPy | Sangoma has an USB FXO thing. |
| 23:55.37 | SpeedEvil | That is unfortunately expensive. |
| 23:55.46 | SpeedEvil | is trying to work out a really cheap solution. |
| 23:55.50 | SpeedEvil | Being on a budget sucks. |
| 23:55.53 | WIMPy | Yes. POTS is expensive. |
| 23:56.50 | SpeedEvil | idly wonders about USB modems and if any do linux. |
| 23:58.56 | WIMPy | You can get USB ISDN for a fiver if you're lucky. But they are becoming rare. |
| 23:59.20 | SpeedEvil | Alas(?) no ISDN here. |
| 23:59.48 | WIMPy | is also looking for PCI cards. Sourcing them has become some issue as well. |
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